PRINCIPLES OF COMMUNICATION SYSTEMS Second Edition Herbert Taub Donald L. Schilling Professors of Electrical Engineering The City College of New York McGraw-Hill Publishing Company New York St Lous San Francisco Auckland Boqots Caracas Hamburg Liston London Madrid Mexico Milan ‘Montreal New Delhi Okioma City) Paris San Juan ‘Sho Paulo ‘Singapore Srdney Tokyo TorontoCHAPTER ONE SPECTRAL ANALYSIS INTRODUCTION Suppose that two people, separated by a considerable distance, wish 10 commu ricate with one another. I there i a pair of conducting wires extending from one Tocation to another, and if each place is equipped with # microphone and eat piece, the communication problem may be solved. The microphone, at one end of the wire communications channel, smpresses an electric signal voltage on the line, ‘which voltage is then received at the other end. The received signal. however, will have associated with it an erratic, random, unpredictable voltage wavelorm whieh is deseribed by the term nose. The origin of this noise will be discussed more flly in Chaps. 7 and 14, Here we need but to note that at the atomic level the universe is in a constant state of agitation, and that this agitation is the source of a very great deal of this noige, Hecause of the length of the wire link. the received message signal voltage will be greatly attenvated in comparison with its level at the transmitting end of the link. As a result the message signal voltage may not be very large in comparison with the noise voltage. and the message will be perceived with difficulty or possibly not at all. An amplifier at the receiving end will not solve the problem, since the amplifier will amplify signal and noise slike. As a matter of fat, as we shall see, the amplifier itself may well be @ source ‘of additional noise. 'A principal concern of communication theory and a matter which we discuss extensively in this book is precisely the study of methods to suppress, as far as possible, the effect of noise. We shall see that, for this purpose, may be better fot to transmit directly the original signal (the microphone output in our2 puNcIPLS OF COMMUNICATION SYSTEMS example). Instead, the original signal is used to generate a different signal wave form. which new signal waveform is then impressed on the line. This processing of the original signal to generate the transmitted signal is called encoding ot nodulation. At the receiving end an inverse process called decoding or demodu- Tation is requited to recover the original signe! ‘may well be that thee i a considerable expense in providing the wire com: munication link. We ate therefore, naturally led to inquire whether we may use the link more effectively by arranging forthe simultaneous transmission over the Tink of more than just a single waveform. It wins out that such multiple tran mission is indeed possible and may be accomplished in a number of ways. Such ‘Simultaneous multiple transmission is called multiplexing and is again @ principal trea of concern of communication theory and of this book. It isto be noted that ‘when wire communications links are employed, then, at least in principle feparate links may be used for individual messapes. When, however, the comme: fications medium is free space, as in radio communication from antenna to antenna, multiplexing i essential in summary then, communication theory addresses itself to the following questions: Given a communication channel, how do we arrange to transmit as many simultaneous signals as possible, and how do we devise to suppress the tllect of noise to the maximum extent possible? In this book, after a few mathe- tatical preliminaries, We shall address ourselves precisely to these questions, frst, to the matter of multiplexing, and thereafter tothe discussion of noise in commu 'A branch of mathematics which is of inestimable value in the study of com- munications systems is spectral analysis. Spectra) analysis concerns itself with the description of wavelorms in the jrequeney domain and with the correspondence between the Irequency-domain description and the time-domain description. It is assumed that the reader has some familiarity with spectral analysis, The presen tation in this chapter is intended as a review, and will serve further to allow 9 Compilation of results which we shall have occasion to use throughout the remainder ofthis text 1.1 FOURIER SERIES! [A periodic function of time oft) having a fundamental period T, ean be represent fd as an infinite sum of sinusoidal waveforms. This summation called a Fourie ‘series. may be written in several forms. One such form is the following. ot) = Ag+ z A, cos Te + ze sin aay ned [08 0 SPECTRAL ANALYSIS 3 while the coeficients 4, and B, are given by ‘Am alternative form for the Fourier series is wip Cu+ 5 Cscos 224 o,) as) whete Co, Cyvand @, are related t0 Ag, A,vand B by the equations Com Ae (1.166) C= JAE (11-66) wd é.2tn (1-60 “The Fourier series of @ periodic function is thus seen to consist of a summ tion of harmonies of a fundamental frequency fy = 1/Tq. The coefficients C, are called spectral amplitudes; that i, C, isthe amplitude of the spectral component 08 (2anfot @,) at frequency nfp- A typical amplitude spectrum of a periodic wavelorm is shown in Fig, I.t-1a, Here, at each harmonic frequency, a vertical Tine has been drawn having a length equal to the spectral amplitude associated with each harmonic frequency. Of course, such an amplitude spectrum, lackins the phase information, does not specify the waveform 1). tlrttiedy 4 3h a o Figure 11-1 (a) A onesided plot of spectral ampbiude of erode waveform, (6) Te eorespondinz ‘wosidd plo4 rnixciris oF conmunacaTion sysrins 1.2 EXPONENTIAL FORM OF THE FOURIER SERIES ‘The exponential form of the Fourier series finds extensive application in commu: nication theory. This form is given by a= 5 erm a2) where His given by vine a (12.2) “The coefficients ¥, have the property that ¥, and V-, are complex conjugates of ‘one another, that is, V, = V2,. These coefiients are related to the C's in Ea (11-5) by © (1239) is om (12.36) ‘The V's are the spectral amplitudes of the spectral components V,e%¥*, The amplitude spectrum ofthe Vs shown in Fig. 1.1-1b corresponds to the amplitude spectrum of the C,'s shown in Fig. -1-la, Observe that while Vo = Co, otherwise each spectral line in I.l-1a at frequency f is replaced by the 2 spectral lines in Li-1b, each of half amplitude, one at frequency f and one at frequency ~f. The amplitude spectrum in 1.1-1a is called a single-sided spectrum, while the spectrum in L.1-1b is called a two-sided spectrum, We shall find it more convenient t0 use the two-sided mplitude spectrum and shall consistently do so {rom this point on. 1.3 EXAMPLES OF FOURIER SERIES ‘A waveform in which we shall have occasion t0 have some speci interest is shown in Fig. 13-1a, The wavelorm consists of @ periodic sequence of impulses of Strength J. As a siier of convenience we have selected the ume feale £0 that an impulse occurs at t= 0. The impulse at ¢= 0 is wntten as I 6). Here d(r) is the delta function which has the property that Jt) = 0 except when 1 = O and further Pos i ‘The strength of an impulse is equal to the area under the impulse, Thus the strength of ) is 1, and the strength of! (0) ‘The periodic impulse train is writen w=1 Atk) (132) SPECTRAL ANALYSIS 5 reuse) HausTe) — [18 umTy) UBT) =e i om ah oe A a Figure 13.4 Examples of perio frtions. (a) A perio aia of implies. (6) pedi ain of pubes of erator» We then find, using Eqs. (1-2) and (1.13). mek fe wwaed 038 asa Vo= Ve te (13-7)6 pnmicins oF COMMUNICATION SYSTEMS Hence oft) may be writen in the forms 1 wal Yok = 7 38) [As a second example, let us find the Fourier series for the periodic train of pulses of amplitude A and duration tas shown in Fig. 13-1. We find (13.9) _2At sin (na Te) el, (1310) and 0 0 3a) Thus At, 2At & sin ne/Te) ... 2am 1) = ES OT, (13-120) sin net/TS) rear, (13.128) Suppose that in the waveform of Fig. 1.3-1b we reduce + while adjusting A so that Arisa constant, say At = I, We would expect that inthe limit, as t=» Othe Fourier series forthe pulse train in Eq, (13-12) should reduce to the series for the impulse train in Eq. (1.3). It is readily verified that such is indeed the case since ast0 sin nns/To)_ fe 03413) 14 THE SAMPLING FUNCTION ‘A function frequently encountered in spectral analysis is the sampling function ‘Sox) defined by [A closely related function is sine x defined by sinc x tion Sala) is plotted in Fig. 14-1. 1 is symmetrical about x = 0, and at x = 0 has SPECTRAL ANALYSIS 7 Figure 144 (0) The fncion Sb) Te spectl amplitudes Wf the 1wosided Fourie represen ation ofthe pulse tino Fig. Lebar = and Ty = the valve Sa(0) = 1, It oscillates with an amplitude that decreases with increasing x The function passes through zero at equally spaced intervals at values of x= tnx, where mis an integer other than zero. Aside from the peak at x = 0. the maxima and minima occur approximately midway between the 2er0s, ie, at x = “(n+ 3)n, where [sin x| = 1, This approximation is poorest for the minima Ria) (1183) ‘This result is rather intuitively obvious since we would surely expect that simi larity between o(t) and (+ x) be a maximum when t = 0, The student is guided through a more formal proof in Prob. 118-1 RG) = Rl) (1.184) Ths the suacorelton oncton san even ution of To prove Es, (1.184, sume atthe aust Ok moved inte ete ection fy aunt The the megrand in E181) mold become th and RC) mols Rose Hi) Si: were meton cen eed em" ys Shit inte at cin bane mo ete on te tle fhe. Ipa Ther Rit) = R(—1) ae Th tres characteris given in as, (1.1210 (1.14 re ears a cnt of Re) dene by E81) bt to fr Hh ened y ae 1163) tnd (163) or the pei case and he ower ese eee. athe Iter fcoue RO) = Ethene aire he ower 1.19 AUTOCORRELATION OF A PERIODIC WAVEFORM When the function is periodic, we may write E neon (119-1) iom ntgrl in the form of Ea (1.162), we have pny moe 2[™ (FE memnll § erm) gus OTs Jara 4 ‘and, using the corte ‘The order of integration and summation may be interchanged in Eq, (1.19-2) If we do so, we shall be left with a double summation over m and n of terms Jy, given by i = oie. a eee PO he “ (11930) items U8 + I) xim=m) (119.38) Since m and m are integers, we see from Eq, (1.19-38) that Iq, = 0 except when m= ~nor m+ n=O. To evaluate Jp in tis latter ease, we return to Eq, (1-19- 3a) and find WV, een (119-4)[32 PRINCIPLES OF COMMUNICATION SYSTEMS Finally, then. Ripe E vyvegetmete= SF pvyetre (4.1950) F cos 2am = 19-58) VP 608 2am > (119-56) = lh +2 We note from Eq, (1.19-Sb) that Rie) = R(—1) as anticipated, and we note as well that for this case ofa periodic wavelorm the correlation R(t) is also periodic with the same fundamental period Ty ‘We shall now relate the correlation function R(x) of a periodic waveform to its power spectral density. For this purpose we compute the Fourier transform of (), We find, using R(e) as in Eq (1.19-Sa), that SCE ate eH a (19.9 FIR) Interchanging the order of integration and summation vielé= sinen= Fine [emerson a as-n Using Ea, (111-22, we may writ Eg (119-7) 2» sima= 5 Kel s- 2 (119-8) ‘Comparing Eq, (1.19-8) with Eq, (18-6), we have the interesting result that for 3 periodic waveiorm Gif) = F TREN (119.9) and, of course. conversely Ri) = FOUN (119-10) Expressed in words, we have the following: The power spectral density and the correlation function f aperiodic waveform are a hourter transform pair. 1.20 AUTOCORRELATION OF NONPERIODIC WAVEFORM OF FINITE ENERGY For pulse-type waveforms of finite energy there isa relationship between the cor- relation function of Eq, (1.181) and the energy spectral density which corre: sponds to the relationship piven in Eg. (1.19.9) for the periodic waveform. This relationship is that the cortelation function R(t) and the energy spectral density are a Fourier translorm pai. This result is established as follows. SPECTRAL ANALYSIS 33, We use the convolution theorem. We combine Eqs. (1.12-1) and (112-3) for the case where the waveforms rj(2) and 0,() are the same waveforms, that is, 14 () = nt) = off, and get Foon aa)tt — 9) és (1201) Since V(—f) = V*f) = F[x(—], Eq, (120-1) may be writen FUN = FICK = | eens — ny de (1.202) ‘The integral in Eq, (1.20-2) isa function of t, and hence this equation expresses F TVW) as « function of tI we want to express F~*[V(/W*(f)} as fonction of r without changing the form of the function, we need but to intet- change rand ¢, We then have * worn f Aojat ~ 2) a (120.3) ‘The integral in Eg. (1.20-3 is precisely R(e), and thus FIR] = VIAN) = LY (120-4) which verifies that R(s) and the energy spectral density | V(/)|? are Fourier trans- form pairs 1.21 AUTOCORRELATION OF OTHER WAVEFORMS In the preceding sections we discussed the relationship between the autocorrela- tion function and power or energy specttal density of deterministic waveforms, We use the term “deterministic” to indicate that at least, in principe, it is pos sible to write a function which specifies the value of the tunction at all times. For such deterministic waveforms, the availability of an autocorrelation function is of ‘no particularly preat value. The avtocorrelation lunetion does not include within itself complete information about the function. Thus we note that the autocorrel ation function is related only to the amplitudes and not to the phases of the spe: tral components of the wavelorm. The wavelorm cannot be reconstructed from a knowledge of the autocorrelation functions. Any charactenstie of a deterministic waveform which may be calculated with the aid of the autocorrelation function may be calculated by direct means a least as conveniently ‘On the other hand, in the study of communication systems we encounter waveforms which are not deterministic but are instead random and unpredictable in nature. Such waveforms are discussed in Chap. 2. There we shall find that for such random waveiorms no explicit function of tume can be writen, The wave- forms must be described in statistical and probabilistic terms. It isin connection with such waveiorms that the concepts of correlation and autocorrelation find their true usefulness. Specifically, it turns out that even for such random wave-[4 PaINcIPLES OF CONOLUNICATION SYSTEMS forms, the autocorrelation funetion and power spectral density are 8 Fourier transform pair. The proof? that such is the case is formidable and will not be undertaken here 1.22 EXPANSIONS IN ORTHOGONAL FUNCTIONS Let us consider a set of functions 9 (x) g0%h -»» 040) °°» defined over the inter~ val x, lo Hi: Conse (nae topes 112 07 Expand and iterate erm by term, Show thst = 2180) RUN} 2 1.182 Determine am expeson forthe coneation function of square wae vig the wales | or (and a period 7.4 princiPits OF COMMUNICATION SYSTEMS 194 () Find the power petal density of squasewave vege by Fourie transforming he cr resto fenton, Use the eats of Prob 1182 18) Compare the answer fo (a) wh he spectral density obcained rom the Fourier sis (Prob 9.1 1982.1 = sin gs (a) Find Ris) (0) MG{/)~ (RL), fd Gs) diety and compare 1204, A wavelorm consis single pub of amplitude A extening fom 1 = (a) Fin the auoeorelation uncon Ref this waseor (8) Calla ne ney spel desy ofthis ule by evabaing Gf) = FRC) (o)Caleaat Gl) rectly by Paseva’s theorem and compare 1.44, Inerchang he nb) a5 0) 08 te waveforms of Fig. 1.-1e and wit his new labe- Ing apply the Grat- Sebi procedure fo nd expentone of the wavelorms i ems of orthonormal 200 1.242, Use the Gram-Schid procedure to expr the anctons in Fig PL2#-2 in verms of ortho vm components 0) sa so] 1242 Ase of seal lk = 12,34 en by 40 cs(erstd) osest ‘Ue the Gram- Schmidt procedure to fad an orthonormal vet functions wich the fonction 5) 1264, Verify Eq (1.251 262 (a) Show that he potsporesn theorem apis o sen functions. Thai show that i) nd 0) ate orthogonal then the sat othe length of) etned a in Ea (25-104) pls the ate ofthe length of xual the guar of he ng of he sum) + 0) (t) Les) ad 3) be the spe shown on Fig, P1262. Draw the ial 0) +540 Show ‘hat g)and tare otbopoea Show ha a6)+ 5? = INP + Ll) igwe P1282 1288. Very Eq (1.2513) SPECTRAL ANALYSIS 58 1284.) Refer to Fig 128.2 and (125-19) Ves tha the parameter in Bg, (1.25.3) she tangent ofthe ange between the anes the au, coordinate tem ad thew connate syste 1b) Expand the lonetion 0) ands) of Fig 121 i terms of orihonrma functions wt) and (whic ate the aes of coordinate stm which rotated 60 eunterlokone rom the cord ste system whose aes are (and wll he same re 1261. Apply the Gram-Schmid procedure to the eight signals of Ea. (1266) Vey that tno thonormal components ufc o alo Tepesenation of any ofthe signals Show thal aa coo. ‘ite sytem ofthese two orthonormal sgn, the geometric representation ofthe cet spas 3° shown in Fig. 126 1262 A set of sah is 5K)» VIP, in wt, f= ~V3P, sm eg A second set of sina 16s = VRP in tus = DP, gt +2) the two se ial ac have the same de Nepasabiy, what the abo PP. 126 Ges 00 seas swe) OStSTs ayo=—M) OsteT Stow that independent fxm ff) he distinguish of Bese is ea by /2E whee Elsie normaiues ener ofthe wart 1264, () Use the Gram Sehr prcrdre t expe the functions in Fig. L264 in ee of boreal components In apy the proce, invoie the fncvons nthe ore Sh ane 30) Pt the lanctons as poi a vont system n whch the sori ee ‘eshed in nit fhe orthonnmal (0) Repeat pat) ener tha the tion ae to ead th de 0 oh fh and 01 Pte acon (0) Show tht te pres of pars) a () the same datas bees ncn reins ou 500 sa sa 5| a 5) 4| 2 2| 2 2| 1 tts 1 ompapat ype T tT? ype 2} -2 -3 -3 -3 igre P64GiarreR TWO RANDOM VARIABLES AND PROCESSES ‘A waveform which can be expressed, at least in principle, as an explicit function of time xf is called a deterministic waveform. Such a waveform is determined for all times in that, il we select an arbitrary time ¢ = 1, there need be no uncertainty bout the value of xt) at that time. The wavelorms encountered in communica~ tion systems, on the other hand, are in many instances unpredictable. Consider, say, the very waveform itself whichis transmitted for the purpose of communica- tion, This waveform, which is called the signal, must, at least in part, be unpri dictable. If such were not the case, ie, if the signal were predictable, then sts transmission would be unnecessary, and the enlire communications system would serve no purpose, This point concerning the unpredictability of the signal is explored further in Chap. 13. Further, as noted briefly in Chap. 1, transmitted signals are invariably accompanied by noise which results from the ever-present apitation of the universe atthe atomic level These noise waveforms are also not predictable. Unpredictable wavelorms such as a signal voltage () or noise voltage ait) are examples of random processes. [Note that in writing symbols lke sft) and n(t) we do not imply that we can write explicit functions for these time functions.) While random processes are not predictable, neither are they completely unpredictable. Its generally possible to predict the foture perlormance of a random process with a certain probability of being correct. Accordingly, in this chapter we shall present some elemental ideas of probability theory and apply them to the description of random processes. We shall rather generally limit our discussion to the development of only those aspects of the subject which we shall have occasion to employ inthis text. [RANDOM VARIABLES AND PROCESSES 57 2.1 PROBABILITY! ‘The concept of probability occurs naturally when we contemplate the possible ‘outcomes of an experiment whose outcome is not always the same. Suppose that ‘one of the possible outcomes is called A and that when the experiment is repeat ‘ed N times the outcome A occurs N, times. The relative frequeney of occurrence of A is NIN, and this ratio N,/N is not predictable unless N is very large. For ‘example, et the experiment consist of the tossing of a die and let the outcome A ‘correspond to the appearance of, say, the number 3 on the die. Then in 6 tosses the nomber 3 may not appear at all, or it may appear 6 times, or any number of times in between. Thus with N = 6, N,/N may be 0 or 1/6, etc, up to NIN = 1 On the other hand, we know ftom experience that when an experiment, whose outcomes are determined by chance is repeated very many times, the relative fre- quency of a particular outcome approsches a fixed limit. Thus, if we were 10 toss 4 die very many times we would expect that Ny/N would turn out (0 be very close to 1/6, This limiting value of the relative frequency of occurrence is called the probability of outcome A, written P(A), s0 that im Ba en) In many cases the experiments needed to determine the probability of an ‘event are done more in thought than in practice. Suppose that we have 10 balls in a container, the balls being identical in every respect except that § are white and 2 are black, Let us ask about the probability that, in a single draw, we shall select a black ball. If we draw blindly, so that the color has no influence on the ‘outcome, we would surely judge that the probability of drawing the black ball is 2/10, We arrive at this conclusion on the basis that we have postulated that there is absolutely nothing which favors one ball over another. There are 10 possible ‘outcomes ofthe experiment, that is, any of the 10 alls may be drawn: ofthese 10 outcomes, 2 are favorable to our interest. The only reasonable outcome we can imagine is that, in very many drawings, 2 out of 10 will be black. Any other ‘outcome would immediately suggest that either the black or the white balls had been favored, These considerations lead to an alternative definition of the prob lity of occurrence of an event A, that s ‘number of possible favorable outcomes total number of possible equally likely ovicomes Its apparent from either definition, Eq. (21-1) oF (21-2) that the probability of occurrence of an event P isa positive number and that O'< P < 1. Ifan event 4s not possible, then P =O, while il an event is certain, P = | PA) A) 212) 22 MUTUALLY EXCLUSIVE EVENTS ‘Two possible outcomes of an experiment are defined as being mutually exclusiee if the occurrence of one outcome precludes the occurrence of the other. In this case, if the events are A, and A; with probabilities P(4,) and P(4,), then the‘94 pRrsciPLs OF COMMUNICATION SYSTEMS the shaded area in Fig, 222-2 measures the probability that m, was transmitted and read as my In Fig. 2.22-3 we represent the situation in which one of four messages m,, ‘mz,ms, and mg are sent yielding receiver responses 5, 5;, 55, and s4. The prob- ability density function fyi) ofthe noise (multiplied by 4) is shown in Fig. 222- 3a. Also shown are the functions Pim,fr— 5) for k= 1,2, 3, and 4, We have taken all the probabilities Pim) to be equal at Pim, = 4. The boundaries of the received response R at which decisions change are also shown and are marked 131 P2384 Pau "The sum of the two shaded areas shown, each of area A, equals the probabil ity P(E, m;) that m, is transmitted and that the message is erroneously read. Thus tne « Lies) Aide) Ars) Aye) Ly OM Figure 222.3 Probailitycemty when fur signals are tarsi (Probably density off) then oie alone i reeved. 8) Probability density hen one of fur sioals i ransmited and ‘esd in nove RANDOM VARIABLES AND PROCESSES. 96 PEE, m,) = 2A. The remaining area under 4fr ~ 53) i the probability that m, is transmitted and is correctly read. This probability is P(C, m,) = 4 — 2A. We find of course that PLE, ms) = P(E, m,). But it is most interesting to note that when wwe evaluate P(E, m,) or PIE, m,) there is only a single area A involved. Hence PLE, m,) = PLE, ma) = 4P(E, my) = 4PLE, ms). Thus, we arrive at the most inte esting result that given four equally likely messages, when we make boundary decisions which assure minimum average likelihood of error, the probability of ‘making an error is not the same for all messages. Of course, ths result applies lor any number of equally likely messages greater than two. “The probability that a message i read correctly is PIC) = PIC, my) + PLC, ma) + PIC, ms) + PIC, ma) A) +d -20)+G-28) 40-4) 68 222-4) Thus the probability ofan error PIE) =1— AC) = 64, 2225) 223 RANDOM PROCESSES ‘To determine the probabilities of the various possible outcomes of an experiment, it is necessary to repeat the experiment many times. Suppose then that we are interested in establishing the statistics associated with the tossing of a die. We ‘might proceed in either of wo ways, On one hand, we might use a single die and toss it repeatedly. Alternatively, we might toss simultaneously a very large number of dice. Intuitively, we Would expect that both methods would give the ‘same results, Thus, we would expect that a single die would yield a particular ‘outcome, on the average, of | time out of 6, Similarly, with many dice we would expect that 1/6 ofthe die tossed would vield a particular outcome ‘Analogously, let us consider s random process such as & noise waveform mie) ‘mentioned at the beginning of this chapter. To determine the statistics of the noise, we might make repeated measurements of the noise voltage output of a single noise source. or we might at least conceptually, make simultaneous mea surements of the output of a very large collection of statistically identical noise sources. Such a collection of sources is ealled an ensemble, and the individyal noise wavelorms are called sample functions. A statistical average may be detet- ‘mined from measurements made at some fixed time 1 = t, on all the sample func. tuons of the ensemble. Thus to determine, say, (, we Would, at =f, measure the voltages n(,) of each noise source, square and add the voltages, and divide by the (large) number of sources in the ensemble, The average so determined is the ensemble average of r(y) [Now nft,) is a random variable and will have associated with ita probability density function. The ensemble averages will be identical with the statistical aver ‘ages computed earlier in Secs. 2.11 and 2.12 and may be represented by the same96 PRINCIPLES OF COMMUNICATION SYSTEMS symbols, Thus the statistical or ensemble average of n’(t,) may be written Elr'(G))] = 1G). The averages determined by measurements on a single sample function at successive times will yield @ time average, which we represent as «ry In general, ensemble averages and time averages are not the same. Suppose. for example, that the statistical characteristics of the sample functions in the ensemble were changing with time, Such a variation could not be reflected in -measorements made ata fixed time, and the ensemble averages would te different at different times. When the statistical characteristics of the sample functions do rnot change with time, the random process is described as being stationary However, even the property of being stationary does not ensure that ensemble ‘nd time averages are the same. For it may happen that while each sample func- tion is stationary the individual sample functions may differ statistically from one another. In this case, the time average will depend on the particular sample func tion which is used co form the average. When the nature of a random process is such that ensemble and time averages are identical, the process is referred to as ‘ergodic. An erpodie process is stationary, but, of course, a stationary Process is not necessarily ergodic ‘Throughout this text we shall assume that the random processes with which we shall have occasion to deal are ergodic. Hence the ensemble average E{nl)} is the same as the time average = ELVIO! since V(t is ergodic. Since E{V(0)] = 0, the de meter reads zero. (6) The true rms meter reads VO) = VEVO} since V() is ergodic. Since V(Q) has a 2er0 mean, the true ms meter reads a= Dolls {) The square and average meter (a fll-wave rectifier meter) yielis & deflec tion proportional te Oy = EO} 224 AUTOCORRELATION A random process nf, being neither periodic nor of finite energy has an autocor~ relation function defined by Eq. (118-1). Thus pre Re tin + | tt +0) dt (2241) In connection with deterministic waveforms we were able to give a physical significance to the concept of a power spectral density G(f) and to show that GU) and Ris) constitute a Fourier transform pair. As an extension of that result wwe shall define the power spectral density of a random process in the same way. ‘Thus for a random process we take G(f) to be eipr= #1R.0)~ [" Row 6 224It is of interest to inquire whether G(f) defined in Eq. (2.24-2) for a random ‘process has a physical significance which corresponds to the physica significance ‘of Gf) for deterministic wavetorms For this purpose consider a deterministic waveform 1!) which extends trom a to ao, Let us select a section of this waveform which extends from ~T/210 T/2. This waveform vt) = et) in this range, and otherwise vt) = 0. The wave form oft) has a Fourier transform Va(S). We recall that |Vo(f)I? is the energy spectral density; that i, [V3(/)F? d's the normalized energy in the spectral range @, Hence, over the interval T the normalized power density is {Vq(/)/?/T. As ‘T+ ce et} ft and we then have the result thatthe physical significance of the power spectral density G(), at leas fora deterministic waveform, i that Gf) = tim : VAS! (224-3) Correspondingly, we state, without proof, that when G() i defined for a random process, asin Eq, (224-2) as the transform of Rie). then G(f) has the significance that aun» tim ef inn} 224-4) where E{_ represents the ensemble average or expectation and Nl/) rep tesems the Fourier uansfonm ofa truncated section of sample function ofthe random process) “The autocorrelation function Rls) i a8 indicated in Eq. (224-1, a time average of the prott ni) and e+ 2), Since we have assumed an erpodic proces, we area ibety to perform the averaging over any sample fencuon of the ensemble, sige every sample function wil veld the same result. Howeres din because the nose process ergodie. we may replace the ime average bY ab msermble average and write, instead of Eq (2.28 Res) = Ejmcnte +9) (2245) “The averaging indicated in Eq, (224-5) has the following significance: At some fixed time 1, ns @ random variable, the possible values for which are the values nt) assumed at time £ by the individual sample functions of the ensemble. Simi: larly, at the fixed time f+ t, n(t+ 2) is also a random variable. It then appears that R(t) as expressed in Eq, (224-5) is the covanance between these two random variables, ‘Suppose then that we should find that for some «, R(r) = 0. Then the random variables nt) and nit +) ae uncorrelated, and for the gaussian process of intr: est t0 us, nc) and nif +4) are independent, Hence, if we should select some sample function, 2 knowledge of the value of ft) at time ¢ would be of no assist nce in improving our ability 1o predict the valve attained by that same sample function at time +t. “The physical fact about the noise, which is of principal concern in connection with communications systems, is that such noise has a power spectral density [RANDOM VARIABLES AND PROCESSES 97 G(J) which is uniform over all frequencies. Such noise is referred to as * white” noise in analogy with the consideration that white light is a combination of all colors, that is, colors of all frequencies. Actually as is pointed ovt in Sec. 14, there is an upper-irequency limit beyond which the spectral density falls off sharply. However, this upper-frequency limit is so high that we may ignore it for ‘our purposes. Now, since the autocorrelation R(t) and the power spectral density G(f) are 1 Fourier translorm pair, they have the properties of such pairs. Thus when G(/) extends over a wide Irequency range, R(s) is restricted to a narrow range of t. In the limit, if GL) = (a constant) forall frequencies from — 0 0 O80 ee110 PRINCIPLES OF COMMUNICATION SYSTEMS 2124. Calelate he variance ofthe andom variables having desis (0) The gvsian density fn) = = (0) The Rayleigh densi fa) = 2e"%,x 20 (0) The unto density fa) = Ya, a2 x a2 fl 2122 Conse the Cauchy density function Sore -wsxee (a) Find K so hat fe) is a density anton (0 Find E00 (c) Find the variance of X. Comment on he inane of his ess 2212. The random variable X has a variance ¢ and a mean m. The random vriabe Yi elated 10 A by Y= aX + where aan Bate constants Find the mea ad vane of ZLIB. A probably density uncon fs unlorm ove the ange rom — L10 + (o)Calulat and plt the probabity PE Tx! se) a2 fneion of 10) alelate and pot he Quay cad very that Tebebyche's ales alii thi cre 24H, Reker tothe gaussian dens piven in Eg (214, (o) Show that B= m=") = (b) Show that BLY — m9 = 1-3-5 (8 De 2142. Givens gausian probly fonction fix) of mean vue eo an variance (a) Asa faneion of, plot the probaly Px > (0) Catelate and plot te genni of and verily that Tebebyehels ral valid in he cas 2143. A random variable V = b+ X, where X isa gsusian stibued random variable wih mean (ang vananee o,and bi constant, Show that V i gausian ditbated random vaabe wh 22144 The ont densiy fnston of wo dependent variables X and Ys pane gemewnn (0) Show tha, when X and Y ate each considered witout tletence 10 the other, ach © ussan variable ie, lh) ad ae passan density Tetons (b) Find of ae 2-1, bin vale for ad plot ef ere 2182. On the same vt fates wed in Prob, 215.1 plot «and ele w versus Compare your 2183. The probabity Pyug= Ambo SX 5 m+ he (o) Change varsles by lxing w= x iV. (by Show that Py, = ei 4, Show tha «random variable with aRayeigh density asin E.(2.16-1) has mean valve ~ Viti mean squire value R= 20, and a ananeeg? = 2~ 23 2462. (a)A voltage Vi anevon of te ad is ven by Vi) = X coors ¥ sn ‘in which oa constant angular frequency and X and Yate independent gaussian variables each ‘ih zero mea an variance Show tat V4) maybe writen YU) eos tat +8) ‘in which Ria random vrable wih a Rayleigh probability density and @ is random variable wih ‘walorm des. (0) Me? = what isthe probably that R > 1 2174, Derive Eg (217-6 iret rom the debton of = (2 —m RIDE Z= Xy 4Xp 404 Kup EXD (e) Find 2 Wmaxy= ip fate) [0 oinerwse find) £2 08 2)03 2.184, The independent random vrables X and Yar aed to form ZI Hear Oexs@ and Siete” Iyer fia fa 2.162 The independent random varisble and Y have he probably deni Sayer? Ozxe% SWn%” Osyen Find and plot the probability density ofthe vane Z = X + 2188 The random variable X has a probity deny uiform inthe range @-s x1 and ze ‘seuhee. The dependent variable "has a density unorm inthe range O's» 2 nd 20 ee thee Find and plot the density of Z = X-+ 2.14 The independent puss random variables... Xy are added form Z ibe mes of 18 Tad ts vance nd 2.1941 Two gaussian random variables X and Y, each with mean zero and variance 6, beter hich hee a corelationeoeficien , hve a joint probably density given ten eat aaa (a) Very tha the symbol pin the expression for fx i indet the corelation cutie ‘That is cvluste E(X Vo" and show that the rests p a requite by Eg, 195, (8) Show that he ate p= Ocoresponds to the ereumstanee where X an ¥ are independert 2.192 The random varible X and Y ate related tothe sandomn vale © by X = in © ane Y = cos©.The vale © has» uni‘orm probability density tthe ange om 102, Show that 3 sand Y are ot independent bt that ponte, BONY) = Oso that they ae uncreles LAVA The candom variables Xj, Xs, Nay Me dependent bat uncorrelated, Z = Xy+ a+ “boos Show that ef =f ee 403 = 22041. The random varshlesX, 25. X, ate independent end each hs aur probability des inthe ange Os 1-Find and pot he probabty desi of Xy + Xsshdo X, +; 3 2244. Very Eq 221-1 2212 In communication system ved to transmit a sequence of messages is known that the ‘ros probably x 107. Asam survey of Veesnges made, eeu tht, with prof aby not 1oexcnd 005, the ero ae in te sampl isnt tobe les than $= 10"? How man ‘espe mst te inelade inthe sr 2221, A communication chanel ransmits in random over, two messages mand m The mesage 'm,oscrs hres times more feqoey than m,. Mensagem, eneratesurcever response r= —1V ‘nites generates ry = +1V- The chanel coraped by nose m wih mlm probably density Sthich eens fom n= —1SV ton= 4150 (o) Find the probabity that my i mistaken for my and tbe probably tht mis mistaken form, {ty Wha tbe probability that the exver wl determine mesg corey?112 pmscaPLss oF COMMUNICATION SYSTEMS 2222. A commonication chanel iansmits. in random ower. wo messages m, and m, with equ lead. Message m, generates esonter, = —1V athe eer and message my Feet r= “FIV. The channel is corrupted with passa aoe with varaeeo? = 1 voll: Find te probability that he rece il etermine a message corel. 2223. A commoniaton chanel ansmits,in random otder, two mesagesm, and m,. The message fm ocut thee ines more ees thas m. Menage m, generate Tevet reponse, = IY ‘thle, generates r= —1¥. The chanel is corrupted by nose mwhowe probably deity hs the Uwangl form shown in Fig. 22220 wit) = Oat ns) = =2V anda fn) = +20 a) What are the ranges ofr for which the decison oe ade that m, was rans a for which tbe decision st be made tat, was trnsmitee (0) Whatie the probability tha the meseage wl be read core 223.1. The farction of tne Zt) = X, cos a1 — X, sin apt isa andom process WX, and X, are Independent gaussian random variables each mth er mean and vanance ene (a) EZ) EC) .an0 ® 2282. 211 = Mio cos ot + ©) Mis random process wih EM) = Oand ELM) = Mf; (a) MO = 0 fed E24 1 249 stoner (6) 18 isan independent random variable such that f6) = BZ) = EP Eeos (oy! + @)) = M2 Is Zi pow sana 2244, Hele 10 Prod 2251. Find 21242. A random proces that x power spectral density Gf) = 2 for — 0 =f w-The random process pase though a lm-put filter whith basa tases fncion HU) =2 for mf fy, and consequently the spectral range of this louble-requency signal and the baseband signal are widely separated. Therefore hhe double-frequency signal is easily removed by a low-pass filter. ‘This method of signal recovery, fo all ts simplicity, is beset by an important ‘convenience when applied in a physical communication system. Suppose that he auxiliary signal used for recovery differs in phase from the auxiliary signal ‘sed in the initial translation. If this phase angle is 8, then, as may be verified Prob. 33-1) the recovered baseband waveform will be proportional to "40) cos 6. Therefore, unless itis possible to maintain 6 = 0, the signal strength at ‘covery will suffer. I it should happen that @ = n/2, the signal will be lost atirely. Or consider, for example, that 0 drifts back and forth with time. Then in 08 20, 3.15) AweurTupe-moouLATION systems 119 this case the signal strength will wax and wane, in addition, possibly, to disap- pearing entirely from time to time. Alternatively, suppose that the recovery auxiliary signal is not precisely at ‘requency f, but is instead at f, + Af In this case we may verify (Prob. 33-2) that the recovered baseband signal will be proportional to mt) cos 2x Aft, resulting in 8 signal which will wax and wane or even be entiely unacceptable if A/ is comps rable to, or larger than, the frequencies present in the baseband signal. This latter contingency is a distinct possibility in many an instance, since usually f. > fy 30 ‘that a small percentage change in f, will eause a Af which may be comparable or larger than fy. In telephone or radio systems, an offset Af < 30 Hz is deemed acceptable, We note, therefore, that signal recovery using a second multiplication requires that there be available at the recovery point a signal which is precisely synchronous with the corresponding auxiliary signal atthe point of the first mult®- plication. In such a synchronous or coherent system a fixed initial phase discrep. ancy is of no consequence since a simple phase shifier will correct the matter. Similarly itis not essential that the recovery auxiliary signal be sinusoidal (see Prob. 3.3-3). What is essential is that, in any time interval, the number of cycles executed by the two auxiliary.signal sources be the same. Of course, in a physical system, where some signal distortion is tolerable, some lack of synchronism may be allowed, ‘When the use of a common auxiliary signal is not feasible, itis necessary to resort to rather complicated means to provide a synchronous auxiliary signal at the location of the receiver. One commonly employed scheme is indicated in Fig. 33-1. To illustrate the operation of the synchronizer, we assume that the baseband signal is a sinusoidal cos wt. The received signal is s(t) = A005 (oq €08 «21, with A a constant amplitude, This signal s(t) does not have 4 spectral component at the angular frequency a. The output of the squaring circuit is Si(0 = A? 608? gt cos? 0,1 (0320) Pb + $ 608 20, 0Kb + 4 cos 20,8) 63:26) [1 boos Yo, + eat | 408 2, — org + £08 2eagt + 608 20, f] 8329 ‘The filter selects the spectral component (42/4) cos 24, which is then applied to 4 circuit which divides the frequency by a factor of 2. (See Prob. 3.3-4) This fr Ses = Syetonaing aod ea egtcor gto} Seeing |) canteed Lal oy" fre " a root th 2 Figae 3341 A simple suarng ynchronie.‘quency division may be accomplished by using, for example, @ bistable mu vibrator. The output ofthe divider is used to demodulate (sultiply) the incoming signal and thereby recover the bascband signal co «yt We turn our attention now to a modification of the method of frequency translation, which has the great merit of allowing recovery of the baseband signal by an extremely simple means. This technique is called amplitude modulation, 34 AMPLITUDE MODULATION AA frequency-translated signal from which the baseband signal is easily recover: able is generated by adding, to the product of baseband and carrier, the carrier signal itself. Such a signal is shown in Fig. 341. Figure 3.4-1a shows the carrier signal with amplitude ,, in Fig. 34-16 we see the baseband signal. The trans- lated signal (Fig. 34-1) is given by ALI + mi] 608 0 ¢ G41) We observe, from Eq. (24-1) as well as from Fig. 34-Lc, thatthe resultant wave> form is one in which the carrier A, cos wt is modulated in amplitude, The process 4f generating such a waveform is called amplitude modulation, and a communica tion system which employs such a method of frequency translation is called an ‘amplitude-modulation system, or AM for short. The designation “carrier” for the auxiliary signal 4, cos «1 seems especially appropriate in the present connec tion since this signal now “ carries” the baseband signal as its envelope. The term “carrier probably originated, however, in the early days of radio when this rela- tively high-frequency signal was viewed as the messenger which actually “carried” the baseband signal from one antenna to another, ‘The very great merit of the amplitude-modulated carrier signal is the ease with which the baseband signal can be recovered. The recovery of the baseband, signal, a process whichis referred to as demodulation or detection, is accomplished with the simple circuit of Fig. 34-2a, which consists of a diode D and the resistor- capacitor RC combination. We now discuss the operation of this circuit briefly and qualitatively. For simplicity, we assume that the amplitude-modulated carrier which is applied at the input terminals is supplied by a voltage source of zero internal impedance. We assume further that the diode is ideal, ie, of eto or infinite resistance, depending on whether the diode current is positive or the diode voltage negative Lot us initially assume that the input is of fixed amplitude and that the resistor R is not present. In this case, the capacitor charges to the peak positive voltage of the carrier. The capacitor holds this peak voltage, and the diode would ‘ot again conduct. Suppose now that the input-carrier amplitude is increased, ‘The diode again conducts, and the capacitor charges to the new higher carrier Peak. In order to allow the capacitor voltage to follow the carrier peaks when the carrier amplitude is decreasing, it is necessary to include the resistor R, so that the capacitor may discharge. In this case the capacitor voltage v, has the form AMPLITUDE-MODULATION systiNs 121 Accor mw o Aclbe mascot Figure 341 (0) A sinusoidal cre. (8) A modulating waveform, () The sinusoidal carrie in (a) ‘modulated by the wavelorm in) shown in Fig. 14-26. The capacitor charges to the peak of each cartier cycle and decays slightly between cycles. The time constant RC is selected so that the ‘change in v, between cycles is at least equal to the decrease in carrier amplitude between cycles. This constraint on the time constant RC is explored in Probs. 34-1 and 34-2. It is seen that the voltage », follows the carrier envelope except that v_ also hhas superimposed on it a sawtooth waveform of the carrier frequency. In Fig. 34-2b the discrepancy between o, and the envelope is greatly exaggerated. In122. rnavctPLss OF CONDAUNICATION SYSTEMS Figure 4.42 (e)A demodolator for an AM signa. (8) Input waveform and output voltages, acro practice, the normal situation is one in which the time interval between cartier cycles is extremely small in comparison with the time required for the envelope to make a sizeable change. Hence 0, follows the envelope much more closely than is suggested in the figure. Further, again because the carrier frequency is ordinarily ‘much higher than the highest frequency of the modulating signal, the sawtooth distortion of the envelope waveform is very easily removed by a filter, 3.5 MAXIMUM ALLOWABLE MODULATION I we are to avail ourselves of the convenience of demodulation by the use of the simple diode circuit of Fig. 34-2a, we must limit the extent of the modulation of the carrier. That such is the case may be seen from Fig. 35-1. In Fig. 35-1a is shown a carrier modulated by a sinusoidal signal. It is apparent that the envelope of the carrier has the waveshape of the modulating signal, The modulating signal is sinusoidal; hence mt) = m cos gt, where m iS a constant. Equation (341) becomes 1) = A,(L +m 608 aq8) 608 2. G54) ASPLITUDE-MODULATION svsTeNS. 123 In Fig. 35-16 we have shown the situation which results when, in Eq, (3.5-1), we adjust m > 1. Observe now that the diode demodulator which yields as an output the positive envelope (a negative envelope ifthe diode is reversed) will not repro- duce the sinusoidal modulating waveform. In this later case, where m > 1, we may recover the modulating waveform but not with the diode modulator. Recovery would require the use of a coherent demodulation scheme such as was employed in connection with the signal furnished by a multiplier is therefore necessary to restrict the excursion of the modulating signal in the direction of decreasing carrier amplitude to the point where the carrier ampli- tude is just reduced to zero. No such similar restriction applies when the modula- tion is increasing the carrier amplitude. With sinusoidal modulation, as in Eq (5-1), we require that |m| <1, More generally in Eq, (3.41) we require that the ‘maximum negative excursion of m() be —1. ‘The extent to which a cartier has been amplitude-modulated is expressed in terms of a percentage modulation. Let A,, (max), and A,(min) respectively, be “a | “ih o Figure 3541 (0) A sinusoidally modulated caver (m <1. (8) A cartier overmadulstd fm > 1) by a sinesoidl modal124, pnayciptss oF COMMUNICATION SYSTEMS the unmodulated carsier amplitude and the maximum and minimum carcier levels. Then if the modulation is symmetrical, the percentage modulation is defined as P, given by P__ Afmax)~ A, _ A, = Admin) _ A,fmax) ~ A,(rnin) 10% A, Ae TA, a) In the ease of sinusoidal modulation, given by Ea. (251) and shown in Fig. 35-10, P= mx 100 percent. Having observed that the signal m{t) may be recovered from the waveform A. + m)} 08 0, ¢ by the simple circuit of Fig. 34-24, itis of interest to note that a similar easy recovery of mie) is not possible from the waveform it) cos ct. That such is the case is to be seen from Fig. 35-2. Figure 35-20 shows the carrier signal. The modulation or baseband signal mis shown in Fig. Fire 35.2 (0) A cacer on (A see ad signal (0) The product mf 08 and its envelope. AMPLITUDE-MODULATION SYSTEMS 125 35-2h, and the product m(t) cos «1 is shown in Fig. 35-2c. We note that the envelope in Fig. 35-2c has the waveform not of m(t) but rather of |), the absolute value of mit). Observe the reversal of phase of the carrier in Fig. 35-2e whenever mi) passes through zero. 36 THE SQUARE-LAW DEMODULATOR ‘An alternative method of recovering the bascband signal which has been super- ‘imposed as an amplitude modulation on a carrier is to pass the AM signal ‘through a nonlinear device. Such demodulation is illustrated in Fig. 36-1. We assume here for simplicity that the device has a square-law relationship between input signal x (current or voltage) and output signal y (current or voltage). Thus y= kx, with ka constant. Because of the nonlinearity of the transfer character- istic of the device, the output response is different for positive and for negative excursions of the carrier away from the quiescent operating point 0 of the device. , « q « ge ay Time ume |e igae 36-1 Thstrating the operation of « aquarelaw demodulator, The output the value of y averaged over many carr cyl,126 PRINCIPLES OF COMMUNICATION SYSTEMS ‘Asa result, and as is shown in Fig. 36-1c, the output, when averaged over atime ‘which encompasses many carrier cycles but only a very small part of the modula- tion cycle, has the waveshape of the envelope. ‘The applied signal is t+ ALI + md] cos et G61) ‘Thus the output ofthe squaring circuit is Y= HA, + ALL + ml] cos 0,1}? 062) Squaring, and dropping de terms as well as terms whose spectral components are located near «, and 20,, we find that the output signal s,(), that is, the signal ‘oulput of a low-pass filter located after the squating cceuit is cAZLin(t) + bm*(0}) 6-3) Observe that the modulation mt is indeed recovered but that m2() appears as Wel. Thus the total recovered signal is a distorted version ofthe original modula- tion. The distortion is small, however, if $0) < IQ) or if [me)| <2. There are two points of interest to be noted in connection with the type of ‘demodulation described here; the first is that the demodulation does not depend ‘on the nonlinearity being square-law. Any type of nonlinearity which does not have odd-function symmetry with respect to the initial operating point will simi- larly accomplish demodulation. The second point is that even when demodu- lation is not intended, such demodulation may appear incidentally when the ‘modulated signal is passed through a system, say, an amplifier, which exhibits some nonlinearity sf0 3.7 SPECTRUM OF AN AMPLITUDE-MODULATED SIGNAL ‘The spectrum of an amplitude-modulated signal is similar to the spectrum of a signal which results from multiplication except, of course that in the former case 4 carrer of frequency fis present. If in Eq. (34-1) mie) is the superposition of, three sinusoidal components me) = m, cos cf + ms COS 3 + my 608 at then the (one-sided) spectrum of this baseband signal appears as at the left in Fig. 3.7-1a. The spectrum of the modulated carrier is shown at the right. The spectral lines at the sum frequencies fc + fy, + fs, and f, + fy constitute the upper- sideband frequencies. The spectral lines atthe difference frequencies constitute the lower sideband, The spectrum of the baseband signal and modulated carrier are shown in Fig. 3.7-1b for the case of a bandlimited nonperiodic signal of finite energy. In this figure the ordinate is the spectral density, ie, the magnitude of the Fourier transform rather than the spectral amplitude, and consequently the carrier is rep resented by an impulse ‘AMPLITUDE-MODULATION sysTeNs 127 pote ot seat compcnnt carer mike mite Specrat sen wer siaband Upper sieband toh 4 tthe o Figure 37-1) Atle the one side spectrum of tA, whereas thee spectral components. ‘At ght the spectrum of4,[1+ mi} cos 21. (Same as (a) except mis 8 nonpeode signal ‘and the Yer uss petal density rather han spetal armpltde- 38 MODULATORS AND BALANCED MODULATORS We have described a “multiplier” as a device that yields as an output a signal which is the product of two input signals. Actually no simple physical device now exists which yields the product alone. On the contrary, all such devices yield, at a ‘minimum, not only the product but the input signals themselves. Suppose, then, that such a device has as inputs a carrier cos «,¢ and a modulating baseband signal mi), The device output will then contain the product mf.) cos w, t and also the signals m(t) and cos w,t. Ordinarily, the baseband signal will be bandlimited to a frequency range very much smaller than , = w/2x. Suppose, for example, that the baseband signal extends from zero frequency to 1000 Hz, while f, = 1 MHz. In this case, the carrier and its sidebands extend from 999,000 to 1,001,000 Ha, and the baseband signal is easily removed by a filter,128 pruncietts oF coummurtcaTIon svsreus me) Aobaadt a amity Accent a Aca | Moise |__F maton J (meet =m) Figwe 34:1 Showing how the output of two amplitude modulators ae combined fo produce # oublesidehand suppresied-arerovipet ‘The overall result is that the devices available for multiplication yield an output carrier as well as the lower- and upper-sideband signals. The output is therefore an amplitude-modulated signal. If we require the product signal alone, we must ake steps to cancel or suppress the carrier. Such a suppression may be achieved by adding, to the amplitude-modulated signal, a signal of cartier fre- quency equal in amplitude but opposite in phase to the earrier of the amplitude- ‘modulated signal. Under these circumstances only the sideband signals will remain. For this reason, a product signal is very commonly referted to as a ‘dduble-sideband suppressed-carrer signal, abbreviated DSB-SC. An alternative arrangement for carsier suppression is shown in Fig, 38-1 Here two physical multipliers are used which are labeled in the diagram as ampli- tude modulators. The carrier inputs to the two modulators are of reverse polarity, as are the modulating signals. The modulator outputs are added with consequent suppression of the carrier. We observe a cancellation not only of the earrier but of the baseband signal m() as well. Ths last feature is not of great import, since, as noted previously, the baseband signal is easily eliminated by a filter. We note that the product terms of the two modulators reinforce. The arrangement of Fig. 3.81 is called a balanced modulator, 39 SINGLE-SIDEBAND MODULATION We have seen that the baseband signal may be recovered from a double-sideband suppressed-carrier signal by multiplying a second time, with the same carrier. It can also be shown that the baseband signal can be recovered in a similar manner even if only one sideband is available. For suppose a spectral component of the baseband signal is multiplied by a carrier cos «wt, giving rise to an upper side- band at ©, + w and a lower sideband at «, — o, Now let us assume that we have filtered out one sideband and are left with, say, only the upper sideband at + 0. If now this sideband signal is again multiplied by cos «,t, we shall generate 4 signal at 20, + o and the original baseband spectral component at If we hhad used the lower sideband at «, ~, the second multiplication would have yielded a signal at 200, — « and again restored the baseband spectral component, Since itis possible to recover the baseband signal from a single sideband, there is aan obvious advantage in doing so, since spectral space is used more cco: romically. In principle, two single-sideband (abbreviated SSB) communications systems can now occupy the spectral range previously occupied by a single amplitude-modulation system or a double-sideband suppressed-carrier system. ‘The baseband signal may not be recovered from a single-sideband signal by the use of a diode modulator. That such is the case is easily seen by considering. for example, that the modulating signal is a sinusoid of frequency f. In this case the single-sideband signal will consist also of a single sinusoid of frequency, say, J.+f. and there is no amplitude variation at all at the baseband frequency to Which the diode modulator can respond, Baseband recovery is achieved at the receiving end of the single-sideband communications channel by heterodyning the received signal with a local carrier signal which is synchronous (coherent) with the earrier used at the transmitting tnd to generate the sideband. As in the double-sideband case itis necessary, in Principle, that the synchronism be exact and, in practice, that synchronism be ‘maintained to a high order of precision. The effect ofa lack of synchronism i dif- ferent in a double-sideband system and in a single sideband system. Suppose that the carrier received is cos «1 and that the local cacvie is cos («1 +8). Then with DSB-SC, as noted in Sec. 33, the spectral component cos xt will, upon demodulation, reappear as cos wt cos @. In SSB, on the other hand, the spectral component, cos ct will reappear (Prob. 39-2) in the form cos (wt — 8). Thus, in fone case a phase offset in carriers affects the amplitude of the recovered signal and, for @ = x/2, may result in a total loss of the signal. In the other case the offset produces a phase change but not an amplitude change. Alternatively, let the local oscillator carrier have an angular frequency offset ‘Aw and so be of the form cos (a, + A). Then as already noted, in DSB-SC, the recovered signal has the form cos wt cos Aa. In SSB, however, the recovered. signal will have the form cos (w + Ao). Thus, in one case the recovered spectral component cos wt reappears with a “ warble,” that is, an amplitude fluctuation at the rate Ao. In the other case the amplitude remains fixed, but the frequency of the recovered signal is in error by amount Aco ‘A phase offset between the received carrier and the local oscillator will cause distortion in the recovered baseband signal In such a case each spectral com- ponent in the bascband signal will, upon recovery, have undergone the same phase shift, Fortunately, when SSB is used to transmit voice or music, such phase distortion does not appear to be of major consequence, because the human ear seems to be insensitive to the phase distortion A frequency offset between carters in amount of Af will use each recovered spectral component of the baseband signal to be in error by the same amount Af. Now, ifit had turned out that the frequency error were proportional to the fre- quency of the spectral component itself, then the recovered signal would sound like the original signal except that it would be at a higher or lower pitch. Such, hhowever, isnot the case, since the requency error is fixed, Thus frequencies in the original signal which were harmonically related will no longer be so related after150. PaInciPtes oF coMuNteATION sysTa ‘eovery. The oeal reli tht equency oer between cases adversely ates the ineligibility of spoken communication and not wel ene onnecton with mic. Asa matter of experienc, turns out tat tn anos fot tess than 30 Hz ieaceeptable othe en The need to keep the tequeneyoffet Af between cases small nomally imposes severe retction onthe fequeney stabilities Of the cir seh sors at both ens ofthe communication sytem For supe: that we renee keep 4/010 Hz or less and that out system uses a carer hequensy of IG Mi Then the sm ofthe fequncy dit in the to ctr Beers oy na oh 1 part in 108 The requted equality in eater frequency may Te moored theough the se of quartz crystal oailitors using etal cot ie sna quency at transmiter an reaver The rcer mua ue su ney ase oe equivalent signals derived from ryt) as thet are channels ne coo tons system ICs lo possible o tone an SSB resver manually and theteby reduce the trequoncy oft. To do thy the operator manual adja te henge ore reeiver case generator unl the recived Sigal sounds “nova Erno operators are able fo tune caret to within 10 or 20 Hz Howeren Cee at oclator dit such tuning must be readjusted pesadialy When the erie equency is very high even quate eystalosilators may she hard presied to maintain adequate sabi. Tn such case i nose ‘tansmit the carer isl along withthe sideband signal At the eee ee carr may be spuatd by leing and ied fo synctonse Weal cate pe erator, When wed for such synchronization the cari tered War ee cas" anday be rns at a sbstanaly ded pow ee I nteesting io note that the aquaring ceived to eves the equ and phase information ofthe DSB-SC system cannot be ued hase In ge ee, is lear that principal compton inthe may of more ideeneed toe Single sideband i the need for supplying an accurate carer shones tthe receiver, 340 METHODS OF GENERATING AN SSB SIGNAL Filter Method A straightforward method of generating an SSB signal is illustrated in Fig. 310-1 Here the bascband signal and a carrier are applied to a balanced modulator, The output of the balanced modulator bears both the upper- and lower-sideband signals. One or the other of these signals is then selected by a filter. The filter is a bandpass filter whose passband encompasses the frequency range of the sideband selected. The filter must have a cutoff sharp enough to separate the selected side- ‘band from the other sideband. The frequency separation of the sidebands is twice the frequency of the lowest frequency spectral components of the bascband signal. Human speech contains spectral components as low as about 70 Hz. AMPLUTUDE-MODULATION sesTEMS 131 roogime 10a aHe eqn SL? (2 J oa] et LL a, IH a fm Figure 340-1 Bick diagram of he iter method of generating a singlesideband signal However, to alleviate the sideband filter selectivity requirements in an SSB system, it is common to limit the lower spectral limit of speech to about 300 Hi. tis found that such restriction does not materially affect the intelligibility of speech. Similarly, itis found that no serious distortion results if the upper limit of the speech spectrum is cut off at about 3000 Hz, Such restriction is advantageous for the purpose of conserving bandwidth. Altogether, then, a typical sideband filter has @ passband which, measured from f,, extends from about 300 to 3000 Hz and in which range its response is quite flat. Outside this passband the response falls off sharply, being down about 40 dB at 4000 Hz and rejecting the unwanted sideband also be at least 40 dB. The filter may also serve, further, to suppress the carter itself. Of course, in principle, no carrier should appear at the output of a balanced modulator. In practice, however, the modulator may not balance exactly, and the precision ofits balance may be subject to some variation with time. Therefore, even if a pilot cartier is to be transmitted, itis well to suppress it at the output of the modulator and to add it to the signal at a later point in a controllable manner. Now consider that we desire to generate an SSB signal with a carrier of, say, 10 MHz. Then we require a passband filter with a selectivity that provides 40 4B of attenuation within 600 Hz at a frequency of 10 MH, a percentage frequency change of 0.006 percent, Filters with such sharp selectivity are very elaborate and difficult to construct. For this reason, it is customary to perform the translation of the baseband signal to the final carrer frequency in several stages. Two such stages of translation are shown in Fig. 3.10-1. Here we have selected the first carrier to be of frequency 100 kHz. The upper sideband, say, of the output of the balanced modulator ranges from 100.3 to 103 kHz. The filter following the bal- ‘anced modulator which selects this upper sideband need now exhibit a selectivity of only a hundredth of the selectivity (40 dB in 0.6 percent frequency change) required in the case of a 10-MHz carrier. Now let the filter output be applied to a second balanced modulator, supplied this time with a 10-MH2z carrier. Let us again select the upper sideband. Then the second filter must provide 40 4B of attenuation in a frequency range of 200.6 kHz, which is nominally 2 percent of the cartier frequency.132 PRINeIPLES OF COMMUNICATION sysTEMS We have already noted that the simplest physical frequency-translating. device is a multiplier or mixer, while a balanced modulator is a balanced arrange- ‘ment of two mixers, A mixer, however, has the disadvantage that it presents at its output not only sum and difference frequencies but the input frequencies as well Still, when its feasible to discriminate against these input signals there is a merit of simplicity in using a mixer rather than a balanced modulator. In the present case, if the second frequency-translating device in Fig, 310-1 were a mixer rather than multiplier, then in addition to the upper and lower sidebands, the output would contain a component encompassing the range 100.3 to 103 kHz as well as the 10-MHz carrier. The range 100.3 to 103 kHz is well out of the range of the second filter intended to pass the range 10,100,300 to 10,103,000 Hz. And it is realistic to design a filter which will suppress the 10-MHz carrer, since the carrier frequency is separated from the lower edge of the upper sideband (10,100,300) by nominally a t-percent frequency change. Altogether, then, we note in summary that when a single-sideband signal is to be generated which has a carrier in the megahertz or tens-of-megahertz range, the frequency translation is to be done in more than one stage—frequently two but not uncommonly three. Ifthe baseband signal has spectral components in the range of hundreds of hertz or lower (as in an audio signal), the fist stage invari- ably employs a balanced modulator, while succeeding stages may use mixers Phasing Method ‘An alternative scheme for generating a single-sideband signal is shown in Fig, 310-2, Here two balanced modulators are employed. The carrier signals of angular frequency w, which are applied to the modulators differ in phase by 90°. woes Aunsinays | esa 0" pase Sheet Figure 310-2 A method of generating single-sideband signal ang balanced modulators and phase shifter, AMPLITUDE-MODULATION SYSTEMS 133 Similarly the baseband signal, before application to the modulators. is passed through a 90° phase-shifting network so that there is a 90° phase shift between any spectral component of the baseband signal applied to one modulator and the like-frequency component applied to the other modulator, To see most simply how the arrangement of Fig. 3.10-2 operates, let us assume that the baseband signal is sinusoidal and appears at the input to one modulator a8 605 oq and hence as sin aq at the other. Also, let the carrier be 0s «7, at one modulator and sin a, ¢ at the other. Then the outputs of the bal- anced modulators (multipliers) are £08 04! 608 a £ = H{e08 (2, — a4) +008 (0, + oq)] (310-1) sin eg sin 0, t = [cos (0, — @g}t ~ 608 (@, + OW) (310-2) If these waveforms are added, the lower sideband results; if subtracted, the upper sideband appears atthe output. In general, if the modulation m(t) is given by ame) = FA, cos (ot + 6) 6.103) then, using Fig, 3.10-2, we se that the output ofthe SSB modulator isin general 1m) 608, & Ht in ot 010-4) where fal) = FA, sin (wt + 0) 6.105) ‘The single-sdeband generating system of Fig, 310-2 generally enjoys less popularity than does the filter method. The reason for this lack of favor is that the present phasing method requires, for satisfactory operation, that a number of constraints be rather precisely met if the carrier and one sideband are adequately to be suppressed. It is required that each modulator be rather carefully balanced to suppress the carrer. It requires also that the baseband signal phase-shifting network provide to the modulators signals in which equal frequency spectral components are of exactly equal amplitude and differ in phase by precisely 90°, Such a network is difficult to construct for a baseband signal which extends over ‘any octaves, It is also required that each modulator display equal sensitivity t0 the baseband signal. Finally, the carrier phase-shift network must provide exactly 50° of phase shift. If any of these constraints is not satisfied, the suppression of the rejected sideband and of the carrier will suffer. The effect on carrier and side- band suppression due to a failure precisely to meet these constraints is explored. in Probs. 3.10-3 and 3.10-4. OF course, in any physical system a certain level of, carrier and rejected sideband is tolerable. Stl, there scems to be a general incli- nation to achieve a single sideband by the use of passive fiers rather than by a ‘method which requires many exactly maintained balances in passive and active circuits, There is an alternative single-sideband generating scheme? which avoids the need for a wideband phase-shifting network but which uses four balanced ‘modulators.138 paivctPLss oF commuNIcATION SYSTEMS Comer f, cans, mic—of Mea Tee efor] ee Le men f % 4 4 a0], TF fem] Pear Le min Fon 4 A 4 — 4 mato a a Te foes] aa 1. mn Figo 134 Muitipexing many baseband sigal ver single commonictions channel ‘multiplexing. As a matter of fact, to facilitate this separation of the individual signals, the carrier frequencies are selected to leave a comfortable margin (guard band) between the limit of one frequency range and the beginning of the next. ‘The combined output ofall the modulators, ie, the composite signal, is applied to @ common communications channel. In the case of radio transmission, the channel is free space, and coupling to the channel is made by means of an antenna. In other cases wires are used, At the receiving end the composite signal is applied to each of a group of bandpass filters whose passbands are in the neighborhood fj, fz,....f,. The fier ‘Ac isa bandpass filter which passes only the spectral range of the output of modu. lator 1 and similarly for the other bandpass filters. The signals have thus been Separated. They are then applied to individual demodulators which extract the baseband signals from the carrier. The cartier inputs to the demodulators are required only for synchronous demodulation and are not used otherwise. The final operation indicated in Fig. 313-1 consists in passing the demodu- lator output through a bascband filter. The bascbund filer is a low-pass fiter with cutoff at the frequency fy, to which the baseband signal is limited. This base- band filter will pass, without modification, the baseband signal output of the ‘modulator and in this sense serves no function in the system as described up to the present point. We shall, however, see in Chaps, 8 and 9 that such baseband Alters are essential to suppress the noise which invariably accompanies the signal REFERENCES 1. el Telephone Laboratori: “Transmission Systems for Communications,” Western Ete Company, Tech, Pub, Winston-Salem, NC 1964 AMPLITUDE MODULATION SYSTEMS 139 2. Norpard,D. EA Third Method of Generation and Detection of Singlesiehand Signals Pro. IRE, December 1956, 5. Voeiker, H.: Demodalation of Single sideband Signals Vin Envelope Detection, IEEE Troms on Communication Technology, pp 22-0, February, 1966 PROBLEMS 24, A signal ois banlimited to the foqueey range Ot fy. Its equncy-ransatd by mut lying bythe signal) = cost Fed 0 thatthe bandwidth ofthe talated signal 1 eoent ofthe Frequency f 22. The Fourier anslorm of mt) is [nt] = MIS) Show Freon 2 = EMU) MUP = 101 2) = 2 cn sgt + 08 ant and (9 = 008 ct -+2 608 Jost 231 The un ola nt gency ea) =m 29 nd by iahpaga by te eves {The pda! wanton annie hough a lowpass a he dub _" (b) What is the maximum allowable value for the phase & if the recovered signal is to be 90 wih pres ny Sg pa mec) 8 SE Te band gn vn heute ial a) = eve y intone tty tneerm ce Bt f2 Wh, protest econ eed tae Sep ir whe bet Fhe oa pa fhe er 22 (Teta al tn he enone gl = ml of a se ee ined esos eden vars iSite el prc fam ta ee, he — E24 The el min DSBS Spl =m ee toy sav bry ara ommend mpl 1 mse fad rey oat free find the expected vale ofthe amplitude ofthe component fe) a 2 MLL The envelope setector shown in Fig. 34-20 i used to recover the signal mf) fom the AM. Signal) ~ (1+) os there mg) a aquare wave aking onthe vals and =O vol and having a period 7» i, Sketch the acovered signal RC ~ 7/20 47.140 princi oF COMMUNICATION SYSTEMS a 442, (o) The waveform a) = (1m 608 gt CO 4 with ma constant (m Sh is appli to the ‘iode demodulator of Fig. 34-22 Show tha ithe demodulator output i 10 flow the envelope of (0) Using the stu of put), show tht i he demodulator isto follow the envelope a times then mst be ss has ore the vale of my. determined rom the equation am (© Draw. qutatively the form of the demectlator output when the condition spied in part, (isnot, SA. The signal i) = (1+ mcor og cor ai detested ing & diode envelope detector. Sketch the detector output when m= 2. 61. The sgn) = (1 +02 608 (y/ 3H) 0050, 8 demodulated wing a square demode intr having the characteristic = [0+ 3. The output 1 then ered by an weal low-pass er having a coo Frequency at fy He Sketch the amplie-requancy chtdcteisics ofthe ouput ‘veform inthe fequeney range O f= fy 162 Repeat Prob 361 ithe squarelaw demodulto is centered at the origin so that 9, =e 263. The signal a ~ [1+ mf] e080, guacelaw detected by detector having the character: inc,» 0 TT the Fourier uaesorm of mea constant M extending fom ~fy tot, sketch he Founértanslorm of inthe frequency ange —ju << fy. Mint: Convolution inthe fequeney domain i needed to find the Fourie easton of mith Se rob. L122 364, The signal) = (14 01 cos 2 +01 €o8 202) cos 0, detected by squire law detector, 1, = 2° Plt the amplitude fequeney characeriticof eh 394. a) Show thatthe signal n= [08,08 out +0) ~ in 0, i (ot + 0) ‘san SSBC signa (o,» 0) I tthe upper of lower sideband? (b) Write anexpesion forthe missing sideband (0) Obtain an expression or he otal DSB-SC sgn 92. The SSO signal in Prob, 39-1 is mulled by cos, and then lowpass ered to reaver the (2) Show that the modulation is completely recovered if the cut frequency ofthe low-pass ese fo = (by ibe lpg signal were oe (t+ 0 nd the recovered signa (6) the mplying onal mer eu f Soye i the recovered signal Ase tat 0 € 49.3 Show thatthe squaring ccult show in Fig. 33-1 will ot pet the generation ofa local ‘slstorvigal capable of demedslatng am SSB-SC signa ‘A404, A baseband signal bandimited to the equeny range 00 to 3000 Hx to be superimposed ‘ona carrier of equate of 40 MHz a8 singe sideband modulation ung the fier metho. Assume ‘hat bandpass lers ae available which will povide 40 JB of attenuation in a fequency interval hich aboot I perent of the ler enter frequency Draw a block diagram ofa stable stem, At ck pint inthe System daw pots inciting the epectal cage Oospied by te signal preset there 5102. The system shown in Fig. 310-2 i wel to generate sngleideband spn Hower an ideal 50° phase shitingnework wich i independet of legueney unattainable: The 30” phase shits approximated by 2 atice network having the rane fonction Hy etoemr3 darrueTupe-MooULATION sysTIMs HL “The input tothe network i sven by Ea. (3.103) IY = 300 He and fy = 3000 show that Hye 22 of f the input signals; however, the frequencies of the spectral components do not. ln the second place, all the operations performed on the signal (addition, subtrac- jon, and multiplication) are linear operations so that superposition applies. Thus, if a baseband signal m(0) introduces one spectrum of components into the ‘nodulated signal and a second signal m() introduces a second spectrum, the pplication of the sum m,(t) + m3(} will introduce a spectrum which is the sum of the spectra separately introduced. All these systems are referred to under the Jesignation “amplitude or linear modulation.” This terminology must be taken a | with some reservation, for we have noted that, at least in the special case of single sideband using modulation with a single sinusoid, there is no amplitude variation, at all, And even more generally, when the amplitude of the modulated signal does vary, the carrier envelope need not have the waveform of the baseband signal We now turn our attention to a new type of modulation which is not charac terized by the features referred to above, The spectral components in the modu- lated waveform depend on the amplitude as well as the frequency of the spectral components in the baseband signal. Furthermore, the modulation system is not linear and superposition does not apply. Such a system results when, in connec: tion with a carrier of constant amplitude, the phase angle is made to respond in some way to a baseband signal. Such a signal has the form At) = A c0s fo. t+ HO] (ata) in which 4 and e, are constant but in which the phase angle () isa function of the baseband signal. Modulation of this type is called angle modulation for obvious reasons. It is also referred to as phase modulation since 4) is the phase angle of the argument of the cosine function. Still another designation is fre- ‘quency modulation for reasons to be discussed in the next section 42 PHASI E AND FREQUENCY MODULATION To review some elementary ideas in connection with sinusoi ‘recall that the function A cos 7,1 can be written as A c08 w= real part (Ae) aay ‘The function eis represented in the complex plane by a phasor of length A and an angle # measured counterclockwise from the real axis. IF = «then the phasor rotates in the counterclockwise direction with an angular velocity « ‘With respect to a coordinate system which also rotates in the counterclockwise direction with angular velocity a, the phasor will be stationary. If in Bq. (41-1) 4 is actually not time-dependent but is a constant, then 1) is to be represented precisely in the manner just described. But suppose = ot) does change with time and makes positive and negative excursions. Then 1() would be represented by a phasor of amplitude 4 which runs ahead of and falls behind the phasor rep- resenting 4 cos ot. We may, therefore, consider that the angle 0.1 + (0), of 10), undergoes a modulation around the angle 0 = 0,1. The waveform of oft) is, therefore, a representation ofa signal which is modulated in phase. If the phasor of angle @ + 40) = c+ (0) alternately runs ahead of and falls behind the phasor = ot, then the first phasor must alternately be rotating ‘more, or less, rapidly than the second phasor. Therefore we may consider that the angular velocity of the phasor of u(t) undergoes a modulation around the nominal angular velocity «,. The signal v(t) is, therefore, an angular-velocity- modulated waveform. The angular velocity associated with the argument of a sinusoidal function is equal to the time rate of change of the argument fie, the I waveforms, let usangle) of the function. Thus we have that the instantaneous radial frequency 2» = dO + p)ldt, and the corresponding frequency f= «/2n is Ld iw Sn zg leet + Od = Fee wo. 422) The waveform o(t)is, therefore, modulated in frequency. In initial discussions of the sinusoidal waveform it is customary to consider such a waveform as having a fixed frequency and phase. In the present discussion we have generalized these concepts somewhat. To acknowledge this gener alization, it is not uncommon to refer to the frequency f in Eq, (42-2) as the instantaneous frequency and (0) as the instantaneous phase. If the frequency variation about the nominal frequency «, is small, that is, if dd(eyde < a, then the resultant waveform will have an appearance which is readily recognizable as ‘a“sine wave,” albeit with a period which changes somewhat from eycle to cycle Such a waveform is represented in Fig. 42-1. In this figure the modulating signal is a square wave. The frequency-modulated signal changes frequency whenever the modulation changes level ‘Among the possibilities which suggest themselves for the design of @ modula tor are the following. We might arrange that the phase #¢) in Eq. (4.1-1) be directly proportional to the modulating signal, or we might arrange a direct pro- portionality between the modulating signal and the derivative, dg(o/dt. From Eq, (42.2, with f, = w,/28 $0 -as—19 29 min} te ay Figure 42:1 An anglemodulaed waveorm. (2) Modulating signal (2) Frequonly-moduted sna where fis the instantaneous frequency. Hence in this latter case the proportion- ality is between modulating signal and the departure of the instantaneous fre- quency from the carrier frequency. Using standard terminology, we refer to the ‘modulation ofthe frst type as phase modulation, and the term frequency modula tion refers only to the second type. On the basis of these definitions itis, of course, not possible to determine which type of modulation is involved simply from a visual examination of the waveform or from an analytical expression for the waveform, We would also have to be given the waveform of the modulating signal. This information is, however, provided in any practical communication system, 43 RELATIONSHIP BETWEEN PHASE AND FREQUENCY MODULATION ‘The relationship between phase and frequency modulation may be visualized further by a consideration of the diagrams of Fig. 4.31. In Fig. 43-1a the phase- modulator block represents device which furnishes an output 1) which is a carrier, phase-modulated by the input signal m{(). Thus H) = A 0s (oo, + Km) 30) K being a constant. Let the waveform m(t) be derived as the integral of the ‘modulating signal m() so that mit te [moa (432) in which k” is also a constant. Then with k = k'K" we have sss fa a Moauiing mie) | totewrter | mt | itty | ete) = litters, (a ‘varal ey m0) | modulator 10" moduited sige) Meanie 5] Favca | feavensy [oe = Apts tno ef mcr | Frguency Toit) = ata Sanat (0) Figee 4341 Itai the reationship between phase and fequency modiThe instantaneous angular frequency is ong [ose [moa] The deviation of the instantaneous frequency from the carrier frequency «,/2nis, ret nk) 43.4) f- kan a Since the deviation of the instantaneous frequency is directly proportional to the ‘nodulating signal, the combination of integrator and phase modulator of Fig. 43-1a constitutes a device for producing a frequency-modulated output. Similarly, the combination in Fig, 43-1b of the differentiator and frequency modulator gen- srates a phase-modulated output, ie, signal whose phase departure from the -atrier is proportional to the modulating signal, In summary, we have referred generally to the waveform given by Eq, (4.1-1) 1s an angle-modulated waveform, an appropriate designation when we have no nterest in, or information about, the modulating signal. When ¢(0 is proportion- a1 to the modulating signal m(), we use the designation phase modulation or PM. When the time derivative of 6(t) is proportional to m(2), we use the term frequency rodulation or FM. In an FM waveform, the form of Eq. (43:3) is of special terest, since here the instantaneous frequency deviation is direetly proportional © the signal m(e) which appears explicitly in the expression. In general usage, sowever, we find that such precision of language is not common, Very frequently he terms angle modulation, phase modulation, and frequency modulation are ased rather interchangeably and without reference to, or even interest in, the ‘nodulating signal, 44 PHASE AND FREQUENCY DEVIATION In the waveform of Eg. (4-1) the maximum value attained by (t, that is, the ‘aximum phase deviation of the total angle from the carrier angle of, i called the phase deviation. Similarly the maximum departure of the instantaneous fre- uency from the carrier frequency i called the frequency deviation. ‘When the angular (and consequently the frequency) variation is sinusoidal with frequency fy, We have, With (4 = 2q 0) = A c08 (wt + sin 40) (44) here fis the peak amplitude of 4(0). In this case which is the maximum phase Jeviation, is usually referred to as the modulation index. The instantaneous fre- jency is (44-24) Let By 008 orgt (44.26) ‘The maximum frequency deviation is defined as Af and is given by = ln (443) Equation (44-1) can, therefore, be written wr-tenfarefias) a While the instantaneous frequency flies in the range f, + Af, it should not be 45 SPECTRUM OF AN FM SIGNAI SINUSOIDAL MODULATION In this section we shall look into the frequency spectrum of the signal Eo which is the signal of Eq. (44-1) with the amplitude arbitrarily set at unity as a matter of convenience. We have = 008 (0,6 + Pinot) (4s) 605 (2,1 + f Sin qt) = €08 0, £08 (f sin Gg) sin o,f sin (P sin og) (45-2) Consider now the expression cos (f sin mt) which appears as @ factor on the right-hand side of Eq. (4-5-2) tis an even, periodic function having an angular frequency «%. Therefore it is possible to expand this expression in a Fourier seties in which «,/2x is the fundamental frequency. We shall not undertake the evaluation of the Coefficients in the Fourier expansion of cos (8 sin «¢) but shall instead simply write out the results. The coefficients are, of course, functions of , and, since the function is even, the coefficients of the odd harmonics are zero. The result is £08 (f sin yt eB) + 25) C08 2948 + 21418) 608 dog $5 + Wa lP) 608 Png + (453) while for sin (8 sin «0, which is an odd function, we find the expansion contains ‘only odd harmonics and is given by sin (sin oq) = 24,(8) sin gt + 24,(8) sin 3ayt $+ Dag ff) sin On — Doge to (45-8) ‘The functions J,(A) occur often in the solution of engineering problems. They are known as Bessel functions of the first kind and of order n. The numerical values of J) are tabulated in texts of mathematical tables.?Putting the results given in Eqs. (4.5-3) and (4.5-4) back into Eq, (45-2) and using the identities, 605 A cos B= 4 cos (A — B) + 4.c0s (4 + B) assy $005 (A — B) — 4 c0s (A + B) (45-6) wwe find that 1(9) in Eq. (4.5-1) becomes sin A sin HE) = Jel) £08 2, ~ Jy(BILC08 (0, ~ ght — 608 (0. + OQ) 4 J, (PIfe08 (2, = 2eag)t + 608 (2, + 20h] ~ J,(f){208 (w, ~ 309} — C08 (, + 30,9)1) : asy Observe that the spectrum is composed of a carrier with an amplitude Joff) and ‘set of sidebands spaced symmetrically on either side of the cartier at frequency separations Of cq, 20m, 30, ef. In this respected the result is unlike that which prevails in the amplitude-modulation systems discussed earlier, since in AM a sinusoidal modulating signal gives rise to only one sideband or one pair of side- ‘bands. A second difference, which is left for verification by the student (Prob. 45-1), is that the present modulation system is nonlinear, as anticipated from the discussion of Sec. 4. 46 SOME FEATURES OF THE BESSEL COEFFICIENTS Several of the Bessel functions which determine the amplitudes of the spectral ‘components in the Fourier expansion are plotted in Fig. 46-1. We note that, at B=, Ja(0) = 1, while all other J,’s are zero. Thus, as expected when there is no ‘modulation, only the carrier, of normalized amplitude unity, is present, while all sidebands have zero amplitude. When f departs slightly from zero, J(P) acquires ‘ magnitude which is significant in comparison with unity, while all higher-order ‘Fs are negligible in comparison. That such is the case may be seen either from Fig. 46-1 or from the approximations? which apply when f <1, that is, JelB) at (ff . JO) = 8 ro ‘Accordingly, for f very small, the FM signal is composed of a cartier and. pair of sidebands with frequencies 1, + @,. An FM signal which is so const- tuted, that is a signal where fi is small enough so that only a single sideband pair is of significant magnitude, is called a narrowband FM signal. We see further, in Fig. 46-1, as 8 becomes somewhat larger, that the amplitude J, ofthe first side- ‘band pair increases and that also the amplitude J» of the second sideband pair (46-1) dain 0.1.2) : Es 7 \ AAI AY or 2, mocsitin index Figee 461 The Bessel functions 4) pote a fonction off for m= 01,2005becomes significant. Further, as continues to increase, Jy, J4, et. begin to soquire significant magnitude, giving rise to sideband pais at frequencies «, £2, (3 2 ‘Another respect in which FM is unlike the Tinar-modulation schemes 1 we have B= 2fq. Therefore the ‘maximum value we may allow for f is determined by the maximum allowable bandwidth and the modulation frequency. In comparing AM with FM, we maythen note, in review, that in AM the recovered modulating signal may be made progressively larger subject to the onset of distortion in a manner which keeps the occupied bandwidth constant. In FM there is no similar limit on the modul tion, but increasing the magnitude of the recovered signal is achieved at the expense of bandwidth. A more complete comparison is deferred to Chaps. 8 and. 9, where we shall take account of the presence of noise and also of the relative ‘power required for transmission. 49 SPECTRUM OF “CONSTANT BANDWIDTH” FM. Let us consider that we are dealing with a modulating signal voltage ¢,oS 2nfqt with vq the peak voltage. In a phase-modulating system the phase angle di!) ‘would be proportional to this modulating signal so that (f(t) = Kg 008 2h, with Ka constant, The phase deviation is = K',, and, for constant 0, the bandwidth occupied increases linearly with modulating frequency since’ > Bla = 2k'%m fr. We may avoid this variability of bandwidth with modulating fre- {quency by arranging that oft) = (k/2afa)eq sin 2xfyt (Ka constant) For, in this Hon, 2 and the bandwidth is B= Q2k/2x)e,, independently of f,. In this latter case, however, the instantaneous frequency is @= ©, + kg COS 2rfyt- Since the instantaneous frequency is proportional to the modulating signal, the initially ‘angle-modulated signal has become a frequency-modulated signal. Thus a signal intended to occupy a nominally constant bandwidth is a frequently-modulated rather than an angle-modulated signal In Fig. 49-1 we have drawn the spectrum for three values of f for the condi- tion that fj is kept constant, The nominal bandwidth B = 2 Af = 2, is conse- {quently constant, The amplitude of the unmodulated carrier at fis shown by a dashed line. Note that the extent to which the actual bandwidth extends beyond the nominal bandwidtl is greatest for small f and large fy and is least for large and small f, In commercial FM broadcasting, the Federal Communications Commission allows a frequency deviation Af 75 kHz. If we assume that the highest audio frequency to be transmitted is 15 kHz, then at this frequency f= Af!fq = 75/15 = 5, For all other modulation frequencies f is larger than 5. When there are f +1 =6 significant sideband pairs so that at f, = 15 kHz the band- width required is B= 2 x 6 x 15 = 180 kHz, which is to be compared with 2 Af= 150 kHz When f= 20, there are 21 significant sideband pairs, and B= 2x 21 x 15/4 = 157.5 kHz, In the limiting case of very large f and corre- spondingly very small fq, the actual bandwidth becomes equal to the nominal bandwidth 2 Af. B= ao) ott t \ -/-_—_ a1 Matias haba ——— Figure 49-1 Spectra of sinusoidally modulated FM signal. The nomial bandwidth 8 = 28), = 2 4f ‘sept fae, 4.10 PHASOR DIAGRAM FOR FM SIGNALS. With the aid of a phasor diagram we shall be able to arrive at a rather physically intuitive understanding of how so odd an assortment of sidebands as in Eq (45-7) yields an FM signal of constant amplitude. The diagram will also make clear the difference between AM and narrowband FM (NBFM), In both of these ceases there is only a single pair of sideband components Let us consider fist the case of narrowband FM. From Eqs. (44-1) (46-1) and (46-2) we have for p< | that 1) = 608 (0,1 + B sin yt) (410-14) wow arf lo,-ot+ Besta, +o (4101) hero Fg 10a Asin cordate wih eae ont octet napa vcicty sepa erty ‘nate system, the phasor for the term (8/2) cos (w, + @_)¢ rotates in a counter- Shue icon n'a taper ey Sie as oe08 tat £ mite B an otannt PY LX =F ona, o Figure 4.1041 (a) Phasor cin fora narrowband FM signal) Phasordingram for sn AM sina (8/2) cos (a, ~ ca} rotates in a clockwise direction, also at the angular veloc- ity «@,. At the time r= 0, both phasors, which represent the sideband com- ponents, have maximum projections in the horizontal direction. At this time one is parallel to, and one is antiparallel to, the phasor representing the cartier, so that the two cancel. The situation depicted in Fig. 4.10-1a corresponds to @ time shortly after = 0. At this time, the rotation of the sideband phasors which are in opposite directions, as indicated by the curved arrows, have given rise to a sum phasor A,. In the coordinate system in which the carrier phasor is stationary, the Phasor 4, always stands perpendicularly to the carrier phasor and has the mag- nitude 4 ‘The cartier, now slightly reduced in amplitude, and A, combine to give rise to a resultant R. The angular departure of R from the carrier phasor is 4. Itis readily seen from Fig, 4.10-1a that since f <1, the maximum value of @ ~ tan $ = f, a8 is to be expected. The small variation in the amplitude of the resultant which appears in Fig. 4.10-1a is only the result of the fact that we have neglected higher- order sidebands. ‘Now let us consider the phasor diagram for AM. The AM signal is (4 main expen, Banta, +e —B nto, 608 at + a) (410-3) Bsin ont (4.102) © and the individual terms are represented as phasors in Fig. 4.10-1b. Comparing Eqs. (4.10-1) and (410-3), we see that there is a 90° phase shift in the phases of the sidebands between the FM and AM cases. In Fig. 4.10-1b the sum A of the side- ‘band phasors is given by a sin ot (410-4) ‘The important diflerence between the FM and AM cases is that in the former the sum A, is always perpendicular to the eatrier phasor, while in the latter the sum Ais always parallel to the cartier phasor. Hence in the AM case, the resultant R does not rotate with respect to the carrier phasor but instead varies in amplitude between 1 4 mand 1 =m. ‘Another way of looking at the difference between AM and NBFM is to note that in NBFM where f <1 0) © 60s @£— sin gt sin ot (410-5) while in AM 4) 08 c.f +m Sin eq C05 et (410-6) Note that in NBFM the frst term is cos @, t, while the second term involves sin ot, a quadrature relationship. In AM both first and second terms involve £08 «wf, an in-phase relationship. To return now to the FM case and to Fig. 410-1a, the following point is worth noting. When the angle completes a full eycle, that is, A, varies from +8. to —f. and back again to +8, the magnitude of the resultant R will have exe- cuted two full cycles. For R isa maximum at Ay =, a minimum at Ay =0, a maximum again when A, =~, and so on. On this basis, it may well be ‘expected that ifan additional sideband pair is to be added to the first to make R ‘more nearly constant, this new pair must give rise to a resultant A, which varies at the frequency 24, Thus, we are not surprised to find that as the phase devi ation f increases, a sideband pair comes into existence at the frequencies 2, + 2a, ‘AS long as we depend on the first-order sideband pair only, we see from Fig. 4.10-1a that cannot exceed 90°. A deviation of such magnitude is hardly ade- «quate. For consider, as above, that Af = 75 kHz and that fy 75,000/50 = 1500 rad, and the resultant R must, in this'case, spin completely about 1500/2 or about 240 times. Such wild whirling is made possible through the effect of the higher-order sidebands. As noted, the first-order sideband pair ives rise to a phasor 4 = J,(f) sin «qt, which phasor is perpendicular to the cartier phasor. It may also be established by inspection of Eq. (45-7) that the second-order sideband pair gives rise to a phasor A; = J2(6) c08 2i»4¢ and that this phasor is parallel to the carrier phasor. Continuing, we easily establish that all odd-numbered sideband pairs give rise to phasors 4, = JB) sin mgt nodd (410-7)Which are perpendicular to the carrier phasor, while all even-numbered sideband pairs give rise to phasors /(B) COS noygt — meven (410-8) which are parallel to the carrier phasor. Thus, phasors Aj, As, Ay, ete, alter= nately perpendicular and parallel to the carrier phasor, are added to carry the end point of the resultant phasor R completely around as many times as may be required, while maintaining R at constant magnitude. Iti left as an exercise for the student to show by typical examples how the superposition of a carrier and sidebands may swing a constant-amplitude resultant around through an arbi- trary angle (Prob. 410-2) 4.11 SPECTRUM OF NARROWBAND ANGLE, MODULATION: ARBITRARY MODULATION Previously we considered the spectrum, in NBFM, which is produced by sinus- ‘idal modulation. We found that, just as in AM, such modulation gives rise to two sidebands at frequencies «, + oy and @, — aq. We extend the result now to an arbitrary modulating waveform. We may readily verify (Prob. 4.11-1) that superposition applies in narrow= band angle modulation just as it does to AM. That is, ff, sin at + 8 sin wat is substituted in Eq. (410-12) in place off sin om, the sidebands which result are the sum of the sidebands that would be yielded by either modulation alone. Hence even if a modulating signal of waveform nf), with a continuous distribu- tion of spectral components, is used in either AM of narrowband angle modul: tion the forms of the sideband spectra will be the same in the two cases, ‘More formally, we have in AM, when the modulating waveform is m() the nal is Pall Let us assume, for simplicity, that m() i a finite energy waveform with a Fourier transform Mljoo). We use the theorem that if the Fourier transform ‘Ml, then FLmfe) cos «, 1] is as given in Eq. (342-4), We then find ALL + mi] 605 wt = A 008 0, + Amt) cos ot (4.111) $80 + 0.) + Heo a) +4 [MG + jo.) + MU — jn) ‘The narrowband angle-modulation and with phase modulation m(), may be written, for |m()| <1, na) A 00s @,t — Amit) sin 0,1 13) so that Flenddl =4 [Ho + 04) + Ho ~ 0) +4 eM EMUo + fo) — MU — jo.) (14) ‘Comparing Eq, (411-2) with Bg, (411-8), we observe that (Flea? = 1FLepal (ats) Thus, if we were to make plots of the energy spectral densities of rau(t) and of nadt, We Would find them identical, Similarly, if m() were a signal of finite power, ‘We would find that plots of power spectral density would be the same. 4.12 SPECTRUM OF WIDEBAND FM (WBFM): ARBITRARY MODULATION* In this section we engage in a heuristic discussion of the spectrum of a wideband FM signal, We shall not be able to deduce the spectrum with the precision that is possible in the NBFM case described in the previous section. As a matter of fa We shall be able to do no more than to deduce a means of expressing approx: imately the power spectral density of @ WBFM signal, But this result is impor- tant and useful Previously, to characterize an FM signal as being narrowband or wideband, we had used the parameter f = Afif., where Afis the frequency deviation and fa the frequency ofthe sinusoidal modulating signal. The signal was then NBFM oF WBFM depending on whether B < 1 or f > 1. Alternatively we distinguished one from the other on the basis of whether one or very many sidebands were produc ed by cach spectral component of the modulating signal, and on the basis of ‘whether or not superposition applies. We consider now still another alternative. Let the symbol v= f—J, represent the frequency difference between the instantaneous frequency and the carrier frequency f; that is, 2) = (k/2z)mie) [see Eq, (43-5)]. The period corresponding to v is T = 1/v. As f varies, $0 also will y and T. The frequency v ie the frequency with which the resultant phasor R in Fig. 4.10-1 rotates inthe coordinate system in which the carrier phasor i fixed In WBFM this resultant phasor rotates through many complete revolutions, and its speed of rotation does not change radically from revolution to revolution, Since, the resultant R is constant, then if we were to examine the plot as a func- tion of time of the projection of R in, say, the horizontal direction, we would rec- ‘ognize it as a sinusoidal waveform because its frequency would be changing very slowly. No appreciable change in frequency would take place during the course of a cycle. Even a long succession of cycles would give the appearance of being of rather constant frequency. In NBFM, on the other hand, the phasor R simply oscillates about the position of the cartier phasor. Even though, in this case, we may still formally calculate a frequency v, there is no corresponding time intervalmoe i-sseemme Py -z0me er maave ar =t6wme far =u a wae wen Mampree | fo} igs | £ of Mager sem sae of 7704 tani Fer 4181 Bock singe fan Armstrong of gnerting an FM sg sing mips to increase the frequency deviation. = ™ before multiplication, to the extent of $= 05. Thus, at f= 50 Hz, we have ‘Af = 25 Hz. Note that at higher modulating frequencies, gis les than 0.5 rad ‘The carrier frequency before multiplication has been selected at 200 kHz, a fre- quency at which very stable crystal oscillators and balanced modulators are readily constructed, ‘As already noted, if we require that Af 75 kHz, then a multiplication by a factor of 3000 is required. In Fig 419-1 the multiplication is actually 3072{= 64 x 48). The values were selected so that the multiplication may be done by factors of 2 and 3, that is, 64 = 2°, 48 = 3 24, Direct multiplication would yield a signal of carrier frequeney 200 kHe x 307 6144 MHz This signal might then be heterodyned with a signal of frequency. say, 6144 ~ 960 = 518.4 MHz The diference signal output of sucha miter would be 2 signal of cari frequency 6 MHz Note particularly that a mise, sine i Yields sum and derence frequencies, will translate the frequency spectrum of an FM signal but wll have no elect on its fequency deviation. In the system of Fig. 4419-1, inorder to avoid the inconvenience of heterodyring ata frequency in the tang of undo nega the eaten ation ae sampled ata point in the chain of multiplies where the frequency is only inthe neighbor hood of approximately 10 MHz. as 2 a ‘A featue, not indicated in Fig. 4194 but which may be incorporated, is to derive the 108 MH mixing signal not fom a separate oscilator but rather through mulpirs from the 02. MHz crystal oscillator. The mltplication required it 108/02 = S42 3, Such a derivation of the 108-MUzsigal will Suppeess the effect of any dit in the frequency of this signal (see Prob. 4191). 4.20 FM DEMODULATORS With a view toward describing how we can recover the modulating signal from a frequency-modulated carrier we consider the situation represented in Fig, 420-1 Here a waveform of frequency fo and input amplitude 4, is applied to a frequency selective network which then yields an output of amplitude A,. The ratio of amplitudes A,/A, is the absolute value of the transfer fonction of the network, that is, | HGea)|, This output waveform is then applied to a diode AM demod) lator Gee Fig. 24-2). The diode demodulator generates an output which is equal to the peak value of the sinusoidal input so that the diode demodulator output is equal to A,, Suppose now that the input waveform, instead of being of fixed fre- quency fo, is actually a frequency modulated waveform. Then even for a fixed input amplitude A,, the output amplitude A, will not remain fixed but instead be modulated because of the frequency selectivity of the transmission network. Cor- respondingly the diode demodulator output will follow the variation of Ay. In short, a fixed amplitude, frequency-modulated input will generate, at the output of the frequency selective network, a waveform which is not only fre- quency modulated but also amplitude modulated. The diode demodulator will ignore the frequency modulation but will respond to the amplitude modulation, (dn general, the frequency-selective network will not only give rise to an ampli= tude change but will also generate a frequency-dependent phase change, as noted in Fig. 420-1. But such a phase change is simply additional angle modulation, which the diode demodulator will ignore.) ‘What we require of an FM demodulator is that the instantaneous output signal A, be proportional to the instantaneous frequency deviation of the received signal from the cartier frequency. If the carrer frequency is fy then we require a linear relationship between A, and (J fo) where fis the instantaneous frequency, Such a linear relationship is indicated in the plot of Fig. 420-16. We Asan elt +8) Aicot ast Frequency Diode a seine Bemodulator -—> @ rection, slopes 2 = Ll" jp Figee 420-1 (a) FM demodulation. (0) Frequency wetve network, ypicaly an LC deirequire actually thatthe linearity extend only as far as is necessary to aocommo- date the maximum frequency deviation t which the carrie is subject. ‘As indicated in Fig. 420-16 let us consider that the frequency selective network has a linear transfor characteristic of slope a over an adequate range in the neighborhood of the carrier frequency f, and that at f= fo, A,/Ai| = R (HU = fe. Then we shall have, a8 required RoAi + Alf fo) (420-1) the term Ry A, in-Eq, (420-1) is a term of fixed value which displays no response to frequency deviation and the second term a4(f- fo) provides the required response to the instantaneous frequency deviation of the input frequency- modulated signal We observe however from Eq. (420-1) that if the amplitude 4, of the input signal is not fixed then the demodulator output will respond to the input ampli- tude variations as well as to frequency deviations. Ordinarily in a frequency- modulated communication system the amplitude of the transmitted signal will not be modulated deliberately so that any such modulation which does appear ‘will be due to noise. Hence itis of advantage, for the purpose of suppressing the noise, to reduce the dependence of 4, on 4, in Eq. (420-1). This is accomplished by passing the incoming signal through a hard limiter as shown in Fig. 4.20-2. ‘The purpose of the hard limiter (or comparator) is to insure that variations of A(t) are removed. Figure 4.20-2 shows the original FM waveform with amplitude variations at the output of the hard limiter. Since the waveform has been reduced to a frequency modulated “square wave" a bandpass filter is inserted to extract to Birst harmonic frequency fg. The resulting FM waveform is now applied to the FM demodulator. Herd Banda Torn {3} Ad songs + H—9] Tee Fite Semeriltor ‘urput of had mie’ ‘igor 420-2 (a) Hard limiter (or comparator) inpt to FM demodulator. (0) FM waveform at ipet sd output of ard ime, Vaal a acne Dios ahs aM ~f)+ BAU f0P two L— ognaiter |“ Mot sone co Derocited 4 iterence |e Frequency ee ei ose eee | eee KoA ~ 2ALf =f) + BALS—fa® Figure 4203 A blanced FM demodulator. Unfortunately, one cannot construct a network having the precisely linear transfer characteristic shown in Eq. (420-1). Indeed, in practical networks the ‘output amplitude appears as y= Ro Ay + AALS ~ fo) + BALS— fa)? + (420-2) ‘The halanced FM demodulator shown in Fig. 420-3 can be used to remove the constant term Ry A, and all even harmonies, thereby reducing the distortion pro- duced by the nonlinearity of the bandpass filters. Here two demodulators are ‘employed, diflering only in that in one case the frequency-selective network has a slope 2 and in the other the slope is —a. The output provided by this balanced ‘modulator is the difference between the two individual demodulators. These indi- vidual outputs ate assuming a second order nonlinearity) Ae + @ALS — fo) + BALL — fa)? (4203) 0 Ay BALL — fa) + BALL So}? (420-4)‘The difference output is then y= 22A(f fo) (420-5) land we see that the linearity of the output of the FM demodulator has been improved, In practice an LC tuned circuit is used as the frequency selective network The relationship between the center frequency of each of the two tuned circuits and their bandwidths on the linearity of the FM demodulator is explored in Prob, 420-4 It has also been shown in the literature that a hard limiter is not required when using a balanced discriminator and that any overdriven amplifier (i, an amplifier which is driven betweeen cutoff and saturation) can be used. 421 APPROXIMATELY COMPATIBLE SSB SYSTEMS SSB-AM We noted earlier in Sec. 3.12 the advantages that would accrue from the avail- ability of compatible singlesideband systems. While precisely compatible systems Aare presently impractical, approximately compatible systems are feasible, such systems are in use, and commercial equipment for such systems is available. In a strictly compatible AM system a waveform would be generated whose envelope «exactly reproduces the baseband signal. Additionally ifthe highest baseband fre~ quency component is fy, the frequency range encompassed by the compatible signal would extend from f,, the carrier frequency, (0, say, f, + fy. An approx- imately compatible system is one in which there is some relaxation of these spe- cifications concerning envelope shape or spectral range, On the basis of our discussion of FM-type waveforms we are now able briefly and qualitatively 10 discuss the principle of operation of one type of approximately compatible AM system, We saw in Sec. 3.11 that if we suppressed one sideband component of an amplitude-modulated waveform, the envelope of the resultant waveform would sill have the form of the modulating signal, provided the percentage modulation, was kept small. Consider now the phasor diagram of the AM signal shown in Fig. 410-16, From this phasor diagram itis apparent that when one of the side- bands is suppressed, the resultant is a waveform which is modulated in both amplitude and phase. The amplitude will vary between 1 + m/2 and 1 ~ m2. The resultant R will no longer always be parallel to the earrier phasor. Instead it will rotate clockwise by an angle g such that tan @=m/2 and rotate counter- clockwise by a similar angle, ‘Thus we find that a carrier of fixed frequency (or phase), when amplitude- ‘modulated, gives rise to two sidebands. But an angle-modulated carrier, when amplitude-modulated, may give rise to a single sideband. Suppose then that we arrange to amplitude-modulate a carrier in such manner that its envelope faith- fully reproduces the modulating signa. Is it then possible to also angle-modulate that carrier so that a single sideband results? It turns out that, to a good approx imation, the answer is yest In the system described by Kahn’ the modulating signal modulates not only the amplitude but the carrier phase as well. The relationship between phase and modulating signal is nonlinear and has been determined, at least in part experi- mentally, on the basis of system performance. An analysis of the system is very. involved because of the wo nonlinearities involved: the inherent nonlinearity of FM and the additional nonlinearity between phase and modulating signal. The system is not strictly single sideband in the sense that 2 modulating tone gives rise not to a single side tone but toa spectrum of side tones. However, all the side tones are on the same side of the cartier. The predominant side tone is separated from the carrier by the tone frequency, and the others are separated by multiples of the modulating-tone frequency. ‘The system is able to operate, in effec, asa single-sideband system because of the characteristics of speech or music for which it is intended, and because the side tones, other than the predominant one, fall off sharply in power content with increasing harmonic number. Most of the power in sound is in the lower- frequency ranges, High-power low-frequency spectral components in sound may sive rise to numerous harmonic side tones, but because of the low frequency of the fundamental, the harmonics will stil fal in the audio spectrum, On the other hand, high-frequency tones, which may give rise to harmonic side tones that may. {all outside the allowed spectral range, are of very small energy. The overall result is that there may well be spectral components that fall outside the spectral range allowable in a spectrum-conserving single-sideband system. However, it has been shown experimentally that such components are small enough to cause no inter- ference with the signal in an adjacent single-sideband channel. SSB-FM ‘A compatible SSB-FM signal has frequency components extending either above or below the carrier frequency. In addition it can be demodulated using a stan- ‘dard limiter-discriminator. The compatible FM signal is constructed by adding amplitude modulation to the frequency-modulated signal. Since the limiter femoves any and all amplitude modulation, the addition of the AM does not affect the recovery of the modulation by the discriminator. By properly adjusting the amplitude modulation, either the upper or lower sideband can be removed. 422 STEREOPHONIC FM BROADCASTING In monophonic broadcasting of sound, a single audio baseband signal is transtai ted from broadcasting studio to a home receiver. At the receiver, the audio signal is applied to a loudspeaker which then reproduces the original sound. The orig-‘Turning now to the composite signal M(Q) in Eq. (422-1), we note that 0s 2nf¢ oscillates rapidly between +1 and —1, and, ignoring temporarily the pilot cartier, we have that M(e) oscillates rapidly between M(Q) = 2L(t) and M(@) = 2R(), The maximum attained by MQ) is then My, = 2Lq Ot My = 2Ry. From Eq, (422-2) My = Voq. Hence, in summary, we find that the addition of the difference signal ¥ t0 the Sum signal ¥, does not increase the peak signal excur Effect of the Pilot Carrier Unlike the DSB-SC signal, the pilot carrier, when added to the other components of the composite modulating signal, does produce an increase in peak excucsion, Hence the addition of the pilot carrier calls for a reduction in the sound signal modulation level. A low-level pilot carrier allows greater sound signal moduli tion, while a high-level pilot carrier eases the burden of extracting. the pilot carrier at the recziver. As an engineering compromise, the FCC standards call for 4 pilot carrier of such level that the peak sound modulation amplitude has to be reduced to about 90 percent of what would be allowed in the absence of a cartier, ‘This 10 percent reduction corresponds toa loss in signal level of les than | 4B. REFERENCES, 1 Bell Telephone Laborstrit: “Transmission Systems for Communication,” Western Eleire Company, Tee Pub, Winston Sater. NC, 1964 2 Jahnke, and F. Emde: "Tele of Functions” Dover Publications tne, New York, 1945, 4. Pipes. As "Applied Mathematics for Eaginers and Physics,” McCraw Hill Book Company, ew York, 1988 4 Blachmn, N: Catelation of the Spec ofan FM Signal Using Woodwatds Theorem, IEEE Trans Communication Tebroogy, August 196 5. Kahn, LR: Compatible Single Sseband, Proc. IRE, vo. 9p. 1503-1527, October, 186. PROBLEMS 4241, Consider the signa cos f+ #0} where 4) agure wave taking om the valves 9/3 vey 2, se (a) Sketch cos fu, ¢+ 40) (0) Pot the phate aa function of ine, (6) Pot the requney ae uation of time 422 tr the wavetor os fot sino) phase-modulated cri, sketch the waveform of the ‘modulating signal Sketch the waveform of the modulating signal the care i fequeney ‘modal. 4341, What are the dimensions of he constant K *, and that appear in Eg. 42-1, (42) and (aan 441, An EM sina in gven by + Enotes (6) 10, ~0.and K = 1, 2.84 the maximum frequency deviations. (hIretch an independent random varabl wifey distibuted between —x and find ‘the ems frequency devition {c) Under the condon of) calle the rms phase deviation. 442. tra [eee fra) where mas proba dey Tin Son) = nla the mt frequency deviation 443. carer which attains a peak voltage ofS volts has a lequency of 100 MHz. This ease is ‘requeney-modulated by a sinsoual waelorm of frequency? KH? torch extent tha theese (@) Sketch |) as anction of (6) Calelate the 3.8 hsadwidth ofthe fier tr of B. (6) Find 8 0 that 98 percent ofthe signal power af the WBEM signal is pase 44132. The two independent modulating signs m() nd m0) ate both gaussian and both of zo mean and variance I voIP. The modulating sigaal mt) is connected to 2 source which can be Trequency-tnedulatd ia such manne that, when m(}= volt eonstas) he soar frequen. i Lay TMH, increas by 3 Kis The modulating signal mis connected in sch manner that when mm4® =| vol (eonstant the source fequeny decteses by 44H. The carer ampitde i volte The imo modulating spnals are applied smoltanenualy. Write an expresion forthe power spel esi of the output ofthe eqeeney- modulated sures. |AUEL, Consider the FM signal wzomfors Eacotor ay] Let fay = Horcach (a) Find BiB = 2009), + (Af) + + (4Pg) (0) Find BiB = 2)... + gg + AD 4142.1 Gi) is gnusin and giver by By. (613-1) ind the rms bandwith B. Compare your ‘esl withthe vale B= 46 Aven in Eq (125). 4181. In Fig. 41541 he vltagesvariale capacitor if a revered-ied pn junction diode whose apacance elated to the revere-iasingvolnge« by Cy (1OQ)/T 45) pk The capactonce {Cy= 200 pF and Li ated for resonance at S Mils when aSned revere vliage 9 = voll ‘pple to the capacitor C. The modulating voltage s ma) = 4-4 M5 sin 2e« 10 the ose Infor ample js 1 volt write am expreion for the aglemedulted outpst waveform which spneare across the tank ct, 441741 2) Inthe multiple ere of Fig 417-1 assume thatthe tantra as cure soaree and iso bined and 50 ven that the eolecoe current conse of alternate haley ofa sinks. ‘ial waveform with peak value of 8 mA The input fequeney of the diving signal st MEta and the maltipiation bya factor of 3s tobe acsomplihed M C= 200 pean the ncior Q = 3nd the inductance ofthe Inductor snd calculate the amped ofthe third Parone voltage acon the tank (t) If enatipiation by 10 is to be accomplished caluate the amplitude ofthe tank voltage. Assume that the font impedance ofthe tank remain the sme a in pat a} “4181 () Consider the narowband waveform 1) = cos (1 + ia oath wih ft and with ‘a, Show that) which has requney deviation f= f.may be writen appronimately as 4 and that this approximation ir coositent with the general expansion for an angle modulated wave formas piven by Ea (45-7). Use the approuimations of Ege (46) and (462). (0) Le) be applied asthe input toa device whose output) (the device is ponies sand isto be used for fequeney multiplication bya Tato of 2) Square the approximate expeeston fo 1) a given in par (oh Compare the spectrom of 0) 90 slelted withthe tact spectam fot a !ngle modulated waveform with frequency deviation 28, “4181, Assume thatthe 108-MHz signal in Fig. 419-1 is derived fom the 200KHE oxisior by rltiplying by Sand tha the 200-4 oxilstor dif by 1 He (a) Find the din het in he L08-MHs ig (0) Find the di inthe cece the resling FM signal 44192. In an Armtzong modsltor, as shown in Fig, 419+, the cata osiatorfequene is 20 ‘re Tis desired in onde to avoid diatoctin, to lint the maximom angular devision 1 #02 ‘The system isto secommedate modulation frequencies down o 40 Hz. Al the oupat ofthe moda tor the carrier fequeney i tobe 108 MHz and the equency deviation 80 RHE Seles mule and ne oselatrequncies to ecomplish thi end 442041. The rarrowband phase modulator of Fig. 4.161 i converted tm frequency modulator by receding the balanced modulator with an intepatr. The input signal i a anusld of ange fe suey ont = fI2 co (0, — a) + A 28 (a, +(6) Show that, unless the frequency deviation is kept small the modulator outpt, when emodulate, wl yell not oly the np sgl bul al its odd harmonics 10) Ifthe modulation freqoency is 50 find the allowable frequen deviation i the normale tend power associated with the third harmon i to be more than percent ofthe fundamental {£202 (a) Consider the FM emodulator of Fig. 420-1 Let the frequency selective network be an RC integrating network. The ba equency ofthe network f= /2eRC) Il the carer fequency ofthe FM waveform 8. how should f be selected so that the demodulator has the grates en tivity, greatest change in output per change in input fequeny)? (0) ith selected for maxim rensitvty and th J, 1 MHz fd the change in dered tor output for a -Ha change inp Hequency 44203, A “zero-crosing™ FM discriminator operates inthe following manner. The modulated wave: ‘oom wo Aco[ aire ‘a ts applied to an electronic cirelt which generates a narrow pute on each acasion when of) pases ‘trough zero The pases are of fied pole, ampitade, and dation. This pas ain apied 10 1 low paste, ap an RC lowpass network of a8 frequency, Asume thatthe bandwith of the baseband waveform ri) 5 y- Discuss the operation ofthis disiminator. Show that Wf. ff the output ofthe low-pass network is indeed proportional 1 the nstantanousFaqueney of i. 4204 (0) A requeny setve network i shown in Fig, P&.20-4,Calelate the ratio IVA/Y¥LN) {the rao ofthe amplitude ofthe ouput to the amplitude ofthe inpt, «fection of requency. ID) fe, 88 rye a flaps Ff 2a/ TEE isthe resonant equeney and Qe RY2ae Liste eneray storage actor of i {> a wn es ‘un L Lene raze (bind the fequencles a which || is hgher by 3B than its value atthe resonant te- ‘avensy, Show tat fo arp Quy thee fequencies are f= fl + VO} Calelate the spe of 11 tthe frequency fg fl + 1193) A frequency modulated waveform with carer fequeny fyi ‘poled as Vf) Whats the aio af the change n amplitude AV) othe change, fof the aantae (©) In the dncimnator of Fig, 420 let the wo foqueny aectve network be the network shown in Fig: PA204 The remnant frequencies of the two networks a to be fy =f + 1/0) nd f, “/Al'= Oak fe being the arin frequency of am input frequency mote sraveorm, Mate a plot ofthe disiminator output as a function ofthe instantaneous frequency ofthe np Use 0, = 9, FIVE ANALOG-TO-DIGITAL CONVERSION PULSE-MODULATION SYSTEMS In Chaps. 3 and 4 we described systems with which we can transmit many signals simultaneously over a single communications channel, We found that at the transmitting end the individual signals were translated in frequency so that each ‘occupied a separate and distinct frequency band. It was then possible at the re- ceiving end to separate the individual signals by the use of filters, In the present chapter we shall discuss a second method of multiplexing. This second method depends on the fact that a bandlimited signal, even if it isa con- tinuously varying function of time, may be specified exactly by samples taken suliciently frequently. Multiplexing of several signals is then achieved by inter- leaving the samples of the individual signals. This process is called time-division iplexing. Since the sample is a pulse, the systems to be discussed are called pulse-amplitude modulation systems, Pulse-amplitude modulation (PAM) systems are analog systems and share a common problem with the AM and FM modulation systems studied earlier That is, each of these analog modulation systems are extremely sensitive to the noise present in the receiver. In this chapter we consider how to convert the analog signal into a digital signal prior to modulation. Such a conversion is called source encoding. We shall see in Chap. 12 that when a digital signal is ‘modulated and transmitted, the received signal is far less sensitive to receiver noise than is analog modulation, Noisy Communications Channels We consider a basic problem associated with the transmission of a signal over a noisy communication channel. For the sake of being specific, suppose we require that a telephone conversation be transmitted from New York to Los Angeles. If(6) Stow that, unless the frequency dition is Kept small, the modsltor output, when emodslte, wl yell aot only the mpl signal but alo its 04d harmonics 10) If he modulation regency is 50 Ha, find the allowable frequen deviation if the normale teod power associated with the third harmonic to be o more than percent ofthe fundamental {£202 (a) Consider the FM demodulator of Fig. 420-1 Let the frequency selective network be an [RC integrating network. The MB frequency ofthe network f= 12eRC) I the ear egoeney ofthe FM waveform 8/,. how should f be sete so thatthe demodulator has the grntst en tity he, greatest change in output pr change in input fequeny)? (0) tn sels for maxim rensivty and th J, 1 Ma od the change in demo lator output for a He change in input Hequency. 4203, A “zerocrosing™ FM discriminator operates inthe following manner. The modulated wave ‘orm wo Acn[ aire na] ts applied o an electronic creuit which generates a narrow ple on each acasion when of) pases ‘trough aero The pases are of fied poleiy amplitude, and dation. This pase train apied 10 ‘low ps fey, ay an RC lowpass network of 8 frequency Assume thatthe bandwith of the baseband waveform ri) y- Diss the operation ofthis disiminator. Show that Wf. J fu the output ofthe low-pass network is indeed proportional 1 the nstantancousFaqueney of i. 44204 (0) A fequeny setve network i shown in Fig, P&.20-4, Calculate the ratio IVASY¥LN) Te the rao ofthe amplitude ofthe ouput to the amplitude ofthe int, «fection of requency. YOY fe 5 88 ge alga Ff a/ TEE iste resonant quency and Qe BYR ste eneray trap actor of ® {> K-9 Pa ¢ un L Ere pases (bind the trequencles a which || is higher by 3B than its value atthe resonant te- ‘avensy, Show tat for arp Que hee equencies ate = fl -+ VO} Calelate the spe of 11 the equency fy =f + 1/Q,) A frequency modulated waveform with eater Fequeny fy 6 ‘poled as Af) What the aio af the change In amplitude AV) othe change, fof the santae (©) In the dncimnatr of Fig, 420-1 lt the wo foqueny sectve network be the network shown in Fig, PA204 The remnant freguences of the two networks a to be fy =f +1105) and f, “/Al'— Oak f being the arin frequency of am input frequency mote sraveorm, Mate a plot ofthe discriminator output as a function ofthe instantaneous frequency of th np Use Q, = 50 FIVE ANALOG-TO-DIGITAL CONVERSION PULSE-MODULATION SYSTEMS In Chaps. 3 and 4 we described systems with which we can transmit many signals simultaneously over a single communications channel, We found that at the ‘transmitting end the individual signals were translated in frequency so that each ‘occupied a separate and distinct frequency band. It was then possible at the re- ceiving end to separate the individual signals by the use of filters, In the present chapter we shall discuss a second method of multiplexing. This second method depends on the fact that a bandlimited signal, even if it is a con- tinuously varying funetion of time, may be specified exactly by samples taken suliciently frequently. Multiplexing of several signals is then achieved by inter- leaving the samples of the individual signals. This process is called time-division iplexing. Since the sample is a pulse, the systems to be discussed are called pulse-amplitude modulation systems, Pulse-amplitude modulation (PAM) systems are analog systems and share a common problem with the AM and FM modulation systems studied earlier That is, each of these analog modulation systems are extremely sensitive to the noise present in the receiver. In this chapter we consider how to convert the analog signal into a digital signal prior to modulation. Such a conversion is called source encoding. We shall see in Chap. 12 that when a digital signal is ‘modulated and transmitted, the received signal is far less sensitive to receiver noise than is analog modulation, Noisy Communications Channels We consider a basic problem associated with the transmission of a signal over a noisy communication channel. For the sake of being specific, suppose we require that a telephone conversation be transmitted from New York to Los Angeles. Ifthe signal is transmitted by radio, then, when the signal arrives as its destination, it will be greatly attenuated and also combined with noise due to thermal noise present in all receivers (Chap. 14), and to all manner of random electrical dis- turbances which are added to the radio signal during its propagation across ‘country. (We neglect as irrelevant, for the present discussion, whether such direct radio communication is reliable over such fong channel distances) As a result, the received signal may not be distinguishable against its background of noise. ‘The situation is not fundamentally different ifthe signal is transmitted over wires ‘Any physical wire transmission path will both attenuate and distort a signal by ‘an amount which increases with path length, Unless the wire path is completely and perfectly shielded, asin the case of a perfect coaxial cable, eletrical noise and crosstalk disturbances from neighboring wire paths will also be picked up in amounts increasing with the path length, In this connection it is of interest to note that even coaxial cable does not provide complete freedom from crosstalk. External low-frequency magnetic fields will penetrate the outer conductor of the coaxial cable and thereby induce signals on the cable. In telephone cable, where coaxial cables are combined with parallel wire signal paths, itis common practice to wrap the coax in Permalloy for the sake of magnetic shielding, Even the use of fiber optic cables which are relatively immune to such interference, does not sig- nificantly alter the problem since receiver noise is often the noise source of largest power, One attempt to resolve this problem is simply to raise the signal level at the transmitting end to 50 high a level tha, in spite of the attenuation, the received signal substantially overrides the noise. (Signal distortion may be corrected separately by equalization, Such a solution is hardly feasible on the grounds that the signal power and consequent voltage levels at the transmitter would be simply astronomical and beyond the range of amplifiers to generate, and cables to handle. For example, at 1 kHz, a telephone cable may be expected to produce fan attenuation of the order of 1 dB per mile. For a 3000-mile run, even if we were satisfied with a received signal of ! mV, the voltage at the transmitting end would have to be 10" volts ‘An amplifier at the receiver will not help the above situation, since at this Point both signal and noise levels will be increased together. But suppose that ‘epeater (repeater is the term used for an amplifier in a communications chasse) is located at the midpoint of the long communications path. This repeater will raise the signal level; in addition, it will raise the level of only the noise intro- duced in the first half of the communications path. Hence, such a midway repea- ter, as contrasted with an amplifier at the receiver, has the advantage of improving the received signal-to-noise ratio. This midway repeater will relieve the bburden imposed on transmitter and cable due to higher power requirements when the repeater is not usd. The next step is, of course, to use additional repeaters, say initially at the one-quarter and three-quarter points, and thereafier at points in between, Each added repeater serves to lower the maximum power level encountered on the ‘communications link, and each repeater improves the signal-to-noise ratio over what would result ifthe corresponding gain were introduced at the receiver. In the limit we might, conceptually at least, use an infinite number of repea- ters. We could even adjust the gain of each repeater to be infinitesimally greater than unity by just the amount {0 overcome the attenuation in the infinitesimal section between repeaters. In the end we would thereby have constructed a channel which had no attenuation. The signal at the receiving terminal of the channel would then be the unaitenuated transmitted signal. We would then, in addition, have at the receiving end all the noise introduced at all points of the channel. This noise is also received without attenuation, no matter how far away fom the receiving end the noise was introduced. If now, with this finite array of Fepeaters, the signal-to-noise ratio is not adequate, there is nothing to be done but to raise the signal level or to make the channel quieter, The situation is actually somewhat mote dismal than has just been intimated, since each repeater (transistor amplifier) introduces some noise on its own accord Hence, as more repeaters are cascaded, each repeater must be designed to more exacting standards with respect to noise figure (see Sec. 14.10) The limitation of the system we have been describing for communicating over long channels is that ‘once noise has been introduced any place along the channel, we are “stuck” with it If we now were to transmit a digital signal over the same channel we would find that significantly less signal power would be needed in order to obtain the same performance at the receiver. The reason for this is that the signifieant par ameter is now not the signal-to-noise ratio but the probability of mistaking a digital signal for a different digital signal. In practice we find that signal-to-noise ratios of 40-60 dB are requited for analog signals while 10-12 dB are required for digital signals. This reason and others, to be discussed subsequently, have resulted in a large commercial and military switch to digital communications. ‘5.1 THE SAMPLING THEOREM. LOW-PASS SIGNALS We consider at the outset the fundamental principle of digital communications; the sampling theorem: Let m(t) be a signal which is bandlimited such that its highest frequency spec- tral component is fy. Let the values of m() be determined at regular intervals, Separated by times T, < 1/2fy, that is, the signal is periodically sampled every T, seconds. Then these samples n(n), where n is an integer, uniquely determine the signal, and the signal may be reconstructed from these samples with no distor- tion. The time 7, is called the sampling time. Note that the theorem requires that the sampling rate be rapid enough so that at least two samples are taken during the course of the period corresponding to the highest-frequency spectral com-mio) aol Jscaxmcey Figure S11 (2) A signal nt whic eo be sample () The sampling function St) const fa rin ‘of very marow unit amplitude poles (} The mpling operation prone inn muir 2) The ‘ample ofthe signal nh ponent. We shall now prove the theorem by showing how the signal may be Feconstructed from its samples. ‘The baseband signal mt) which is to be sampled is shown in Fig, S.l-la, A periodic train of pulses S(0) of unit amplitude and of period T, is shown in Fig. S.1-1b. The pulses are arbitrarily narrow, having a width dt. The two signals ms) and S(t) are applied to a multiplier as shown in Fig. 5.1-1c, which then yields as, ‘an output the product Sic). This product is seen in Fig. $1-1d to be the signal ‘m(¢) sampled at the occurrence of each pulse, That is, when a pulse occurs, the multiplier output has the same value as does mt), and at all other times the ‘multiplier output is zero. ‘The signal Sc is periodic, with period 7, and has the Fourier expansion [see ee tsdea ee (en rontxets:) enn For the case T, = 1/2/y, the product S(tm() is sent) = my + & 2m) cos 2h + 2m) cos nla +=]. (5.2) We now observe thatthe fist term in the sees i, aside from a constant factor, the signal m) sell Again, aide from a multiplying factor, the second term ithe provct of mit) and a sinusoid of frequency fy. This product then, as discussed in Se. 32, gives rie to a doublesidcband suppresed-carcier signal wih easier frequency 2fy. Similarly, sueceedng terms yield DSB-SC signals with carrie lr avencies Sr Save pe o iw csemnn Figere 5:12 (o} The magnitude plot ofthe specel density of signal bandimited ty. (8) Plot of amplitude of pectrom of spl sina. Let the signal m() have @ spectral density M(j«) = F{nit)} which is as shown in Fig. 51-22. Then m() is bandlimited to the frequency range below fy The spectrum of the first term in Eq, (51-2) extends from 0 to fy. The spectrum of the second term is symmetrical about the frequency 2fy and extends from 2jy fu = fu 10 Ya + fu = 3y- Altogether the spectrum of the sampled signal has the appearance shown in Fig. 5.1-2b. Suppose then that the sampled signal passed through an ideal low-pass filter with cutoff frequency at fy. If the filter transmission were constant in the passband and if the cutoff were infinitely sharp at fg, the filter would pass the signal mt) and nothing else “The spectral pattern corresponding to Fig. 5.1-26 is shown in Fig, 51-34 for the case in which the sampling rate f,= 1/T, is larger than fy. In this ease there ir cacomeon 42% At te 4 ‘ Nhe i estomio Figure S13 () A guard band appears when f,> 2. (0) Over- lapping of specs when, < 2his a gap between the upper limit fy of the spectrum of the baseband signal and the lower limit of the DSB-SC spectrum centered around the carrie frequency J, > By. For this reason the low-pass filter used to select the signal mi) need not have an infinitely sharp cutoff. Instead, the fiter attenuation may begin at fy but need not attain a high value until the frequency f, — fy. This range from fy to Jy fue is called a guard band and is always required in practice, since a filter with infinitely sharp cutoffs, of course, not realizable. Typically, when sampling is used in connection with voice messages on telephone lines, the voice signal is limited to fy = 33 kHz, while f, is selected at 8.0 kHz. The guard band is then 80-2 33 = 14 eH The situation depicted in Fig. 5.1-3b corresponds to the case where f, < fy Here we find an overlap between the spectrum of mit) itself and the spectrum of the DSB-SC signal centered around f,. Accordingly, no filtering operation will allow an exact recovery of m(). We have just proved the sampling theorem since we have shown that, in prin- ciple, the sampled signal can be recovered exactly when T, < 1/2fy. It has also been shown why the minimum allowable sampling rate is 2jy. This minimum sampling rate is known as the Nyquist rate. An increase in sampling rate above the Nyquist rate increases the width of the guard band, thereby easing the problem of filtering. On the other hand, we shall see that an increase in rate extends the bandwidth required for transmitting the sampled signal. Accordingly ‘an engineering compromise is called for. ‘An interesting special case is the sampling of a sinusoidal signal having the frequency fy. Here, all the signal power is concentrated precisely atthe cutof fre- quency of the low-pass filter, and there is consequently some ambiguity about whether the signal frequency is inside or outside the filter passhand. To remove this ambiguity, we require that f,> Ypy rather than that f, > jy. To see that this condition is necessary, assume that f, = %fy but that an inital sample is taken at the moment the sinusoid passes through zero, Then all successive samples will also be zero, This situation is avoided by requiring f, > 2fy Bandpass Signals For a signal mit) whose highest-frequency spectral component isfy, the sampling frequency f, must be no less than f, = 2fy only if the lowest-frequency spectral component of mis f, = 0. In the more general case, where f, #0, it may be that the sampling frequency need be no larger than f, = 2(Jy — fi). For example, if the spectral range of a signal extends from 10.0 to 10.1 MHz, the signal may be reco- vered from samples taken ata frequency f, = (10.1 — 10.0) = 02 MHz To establish the sampling theorem for such bandpass signals, let us select a sampling frequency f, = 2Uf — fz) and let us initially assume that it happens that the frequency f, turns out to be an integral multiple off, that is, f=, with fn integer. Such a situation is represented in Fig. 51-4. In part a is shown the tworsided spectral pattern ofa signal m() with Fourier transform M( joo). Here it hhas been arranged that m = 2; that is f, coincides with the second harmonic of ah a ° ‘ 2 3 Figure 51-4 (0) The spectrum ofa bandpass signal. (6) The spectrum ofthe sampled bandpass signal the sampling frequency, while the sampling fequency is exactly f, = fy fi. In part bis shown the spectral pattern of the sampled signal Smt) The product of tm) ad the de tr of St) CE, (5.1-1)] duplicates in part b the form ofthe spec- tral pattern in part a and leaves i inthe same frequeney range from, t0fy. The product of m() and the spectral component in St) of frequency f{ = IT) gives Fie in part b to. spectral pattem derived from part by shifting the pattern in part ato the right and also tothe let by amount, Similarly, the higher harme- ies of fin S@) give rise to corresponding shits right and tet, of the spectral pattern in part a We now note that ifthe sampled signal S(@m() is pased through a bandpass filter with arbitrarily sharp cutofs and with passband from Si '0 fue» the signal m(t) will be recovered exactly. Tn Fig 51-4 the spectrum of m() extends over the ist half ofthe fequency interval between harmonics ofthe sampling frequency, that i fom 20, to 2.5 Asa reslt there is no spectrum overlap, and signal recovery is possible. Te may ako be seen from the figure that if the spectral range of mf) extended over the Second half ofthe interval fom 2,0 3.0, there would similary be no overlap. Suppose, however that the spectrum of mt} were confined neither tothe frst halt not tothe second half ofthe interval between samping-requency harmonic. In Such a case, there would be overlap between the spectrum patterns, and signal recovery would not be possible. Hence the minimum sampling frequency allow- abl is, = 2Uy ~ J) provided that either fy or fis a harmonic off Tenither fy nr f, is harmonic off, more general analysis required. In Fig. S1-5a we have reproduced the spectral pattern of Fig. 5-4, The positive frequency part and the negative-trequency part of the spectrom are called PS and INS respectively. Let us, for simplicity, consider separately PS and NS and theselection of @ sampling frequency f, = 2B =2 kHz brings us to a point in an overlap region. As f, is increased there is a small range of f,, corresponding to N= 3, where there is no overlap. Further increase in f, again takes us to an overlap region, while still further increase in f, provides @ nonoverlap range, cor- responding to N= 2 (from f, = 3.5B to f, = 5B). Increasing f, further we again enter an overlap region while at f, =7B we enter the nonoverlap region for N = |, When f, = 7B we do not again enter an overlap region. (This isthe region where f, 2 2fyi that is, we assume we have a lowpass rather than @ bandpass signal) ‘The Diserete Fourier Transform ‘There are occasions when the only information we have available about a signal is a set of sample values, NV in number, taken at regularly spaced intervals T, over 4 period of time T,. From this sampled data we should often like to be able to arrive at some reasonable approximation of the spectral content of the signal, If the sample period and the number of samples is adequate to give us some con- fidence that what has been observed of the signal is representative of the signal generally, we may indeed estimate its spectral content. We pretend that the signal is periodic with period Tp and we pretend, as well, that the sampling rate is adequate to satisfy the Nyquist etiterion. A typical set of sample values is shown in Fig. 5.1-7. Here, for simplicity we have assumed an ‘even number of samples so that we can place them symmetrically in the period. interval To and symmetrically about the origin of the coordinate system. The sample values are located at + 7,/2, 37,2, ete If there are N samples then the samples most distant from the origin are at -£((N — 1217, Ifthe waveform to be sampled is m(), then after sampling, the waveform we have available is m)S(), where S(o) is the sampling function, ie, S() ~ 1 during the sampling duration dr, as shown in Fig. 51-7, and S(t) = 0 elsewhere. We note from Figs. $.1-2 and 5.1-3a that the spectrum of mit) and the part ofthe spectrum, of mi)S() up to fy are identical in form. Hence, to find the spectrum of mit) we may evaluate instead the spectrum of mie)S(). The spectral amplitudes of m(}S() mile Pome a 619) (En cee Using these values for inthe integrand, we find that Eq, (51-9) becomes: tO dk TjePaeTere (S.1-10) Tone -ir- aya {i _ 3 a Ay F fi =e : 8 mt a TA Figne 81-7 A possible st of sample values of» waveform mf taken every T, over atime itera ‘As a purely mathematical exercise we could use Eq. (.1-10) (0 caleulate spectral components #, for any value of n, But consistently with our assumptions, the largest value of m allowable is determined by the Nyquist criterion. The highest frequency component should have a period which is 27, s0 that: Gat) The fundamental period is T, so that the fundamental frequeney is therefore fy MTq. Since To = NT, we have: 7 1 ph (1.13 fArmax) Hence, the highest value of n for which Eq. (5.1-10) should be used is m = N/2. 52 PULSE-AMPLITUDE MODULATION ‘A technique by which we may take advantage of the sampling principle for the purpose of time-division multiplexing is illustrated in the idealized representation of Fig, 52-1. At the transmitting end on the left, a number of bandlimited signalsCammutsor ecommatr min =— —)] 2 Sieh hee emia Figure $21 Iustating how the sampling principe maybe sed to tans number of han clsignals over a singlecommuricaion anna are connected (the contact point of a rotary switch. We assume that the signals are similarly bandlimited. For exariple, they may all be voice signals, limited to 3.3 KHz, As the rotary arm of the switch swings around, it samples cach signal sequentially. The rotary switch at the receiving end is in synchronism with the switch at the sending end. The two switches make contact simultaneously at simi- larly numbered contacts. With each revolution of the switch, one sample is taken of each input signal and presented to the correspondingly numbered contact of the recciving-end switch. The train of samples at, say, terminal 1 in the receiver, pass through low-pass filter 1, and, at the filter output, the original signal m,(0) appears revonstructed. Of course, if fy is the highest-frequency spectral com- Ponent present in any of the input signals, the switches must make at least 2y revolutions per second. When the signals to be multiplexed vary slowly with time, so that the sam- pling rate is correspondingly slow, mechanical switches, indicated in Fig. 52-1, ay be employed, When the switching speed required is outside the range of samping Figure 82-2 The itevacng of wo Baseband signal, S| ee ERR ‘mechanical switches, electronic switching systems may be employed. In either ‘event, the switching mechanism, corresponding to the switch at the left in Fig. 52-1, which samples the signals, is called the commutator. The switching ‘mechanism which performs the function of the switeh at the right in Fig. 52-1 is called the decommutator. The commutator samples and combines samples, while the decommutator separates samples belonging to individual signals so that these signals may be reconstructed, The interlacing of the samples that allows multiplexing is shown in Fig. 52-2. Here, for simplicity, we have considered the case of the multiplexing of just (wo signals m,() and m,(t). The signal m,() is sampled regularly at intervals of 7, and atthe times indicated in the figure. The sampling of m( is similarly regular, but the samples are taken at a time different from the sampling time of m,(). The input waveform to the filter numbered 1 in Fig. 52-1 is the train of samples of im,(t), and the input to the filter numbered 2 is the train of samples of m,(0). The timing in Fig, 52-2 has been deliberately drawn to suggest that there is room to ‘multiplex more than two signals. We shall see shortly, in principle, how many signals may be multiplexed. We observe that the train of pulses corresponding to the samples of each signal are modulated in amplitude in accordance with the signal itself Accordingly, the scheme of sampling is called pulse-amplitude modulation and abbreviated PAM. Multiplexing of several PAM signals is possible because the various signals are kept distinct and are separately recoverable by virtue of the fact that they are sampled at different times. Hence this system is an example of a timecdivision ‘multiplex (TDM) system. Such systems are the counterparts in the time domain of the systems of Chap. 3. There, the signals were kept separable by virtue of their translation to different portions of the frequency domain, and those systems are called frequency-division multiplex (FDM) systems. I the multiplexed signals are to be transmitted directly, say, over a pair of wires, no further signal processing need be undertaken. Suppose, however, we requite to transmit the TDM-PAM signal from one antenna to another. It would then be necessary to amplitude-modulate or frequency-modulate a high- frequency carrier with the TDM-PAM signal; in such a case the overall system ‘would be referred to, respectively, as PAM-AM or PAM-FM. Note that the same terminology is used whether a single signal or many signals (TDM) are transmitted. 53 CHANNEL BANDWIDTH FOR A PAM SIGNAL Suppose that we have N independent baseband signals m,(0), ma(t ete each of, which is bandlimited to fy. What must be the bandwidth of the communications channel which will allow all N signals to be transmitted simultaneously using PAM time-division multiplexing? We shall now show that, in principle at least, the channel need not have a bandwidth larger than Nf.‘The baseband signal, say m(t), must be sampled at intervals not longer than. /2fy. Between successive samples of m,(t) will appear samples of the other N—1 signals, Therefore the interval of separation between successive samples of different baseband signals is 1/2f, N. The composite signal, then, which is pre= sented to the transmitting end of the communications channel, consists of & sequence of samples, that is, a sequence of impulses. If the bandwidth of the channel were arbitrarily great, the waveform at the receiving end would be the same as at the sending end and demultiplexing could be achieved in a straightfor- ward manner. If, however, the bandwidth of the channel is restricted, the channel response to an instantaneous sample will be a waveform which may well persist with sig- nificant amplitude long after the time of selection of the sample. In such a case, the signal at the receiving end at any particular sampling time may well have sig nificant contributions resulting from previous samples of other signals. Conse- quently the signal which appears at any of the output terminals in Fig. 52-1 will not be a single baseband signal but will be instead a combination of many or ‘even all the baseband signals. Such combining of baseband signals at a communi cation system output is called crosstalk and isto be avoided as far as possible. Let us assume that our channel has the characteristics of an ideal low-pass filter with angular cutoff frequency «@, = 2xf,, unity gain, and no delay. Let a sample be taken, say, of m(0}, at ¢=0. Then at 1 = O there is presented at the transmitting end of the channel an impulse of strength I, =m,(0) dt. The response atthe receiving end is sp (0) given by (see Prob, 53-1) Lye sin ot . Salt) AST (534) The normalized response nsg(i/o, is shown in Fig. 53-1 by the solid plot. At { = O the reponse attains a peak value proportional tothe strength ofthe impulse bot ia Figure 53:1 The respons of a ideal low-pass iter to an instantaneous sample at = 0 (slid plot. “The respons toa sample at t= 1% (athe pot. 1 = m,(0) dt, which isin turn proportional to the value of the sample m,(0). This response persists indefinitely. Observe, however, that the response passes through zero at intervals which are multiples of x/o, = 1/2. Suppose, then, that a sample of mi) is taken and transmitted at ¢ = 1/2f..1f Ty = ms(t = 1/2) dt, Lo, sin oxlt ~ Uh) x oft 12h) This response is shown by the dashed plot. Suppose, finally, that the demultiplex- ing is done also by instantaneous sampling at the receiving end of the channel, for my() at ¢ = 0 and for ma(t) at ¢ = 1/2. Then, inspite ofthe persistence of the cchannel response, there will be no crosstalk, and the signals my(t) and m,(t) may bbe completely separated and individually recovered, Similarly, additional signals ‘may be sampled and multiplexed, provided that each new sample is taken syn- chronously, every 1/2f, s. The sequence must, of course, be continually repeated every 1/2f 5,80 that each signal is properly sampled, ‘We have then the result that with a channel of bandwidth f, we need to separate samples by intervals 1/2J,, The sampling theorem requires that the samples of an individual baseband signal be separated by intervals not longer than 1/2fy. Hence the total number of signals which may be multiplexed is N= Sells 0 f. = Ny 38 indicated earlier, Tn principle then, multiplexing a number of signals by PAM time division requires no more bandwidth than would be required to multiplex these signals by frequency-division multiplexing using single-sideband transmission, salt) (53.2) 54 NATURAL SAMPLING It was convenient, for the purpose of introducing some basic ideas, to begin our discussion of time multiplexing by assuming instantaneous commutation and fy» ot, correspondingly, if t € iffy. the aperture distortion will be small ‘The distortion becomes progressively smaller with decreasing . And, of course, as ++ 0 (instantaneous sampling) the distortion similarly approaches zero.Equalization? As inthe case of natural sampling, so also inthe present ese of fat-tp sam ling its advantageous to make # a lange a practeable forthe ake of rene ing the amplitude ofthe output signal ina particular case should harpea that the consequent distortion isnot acceptable, it may be corre by ince an equalizer cascade withthe output low pas flee An equlaes the present instance, sa pasive network whose transfer funciona een pendence ofthe form w/sin x, that a frm inverse othe form of Ho) git in Eq, (55-2) The equalizer in combination with the aperue eet ail ty vied Mat overall transfer characteristic between the original Sachand sal and the oviput atthe receiving end ofthe system, The equalizer Wen cannot be exactly synthesized, but ean be approximated. TEN signals are muller, = 1/3 8 and hence fr large N «Uf and xs eT th ese the eat not nd se nelle oon 56 SIGNAL RECOVERY THROUGH HOLDING We have already noted that the maximum ratio 1/7, of the sample duration to the sampling interval, is 1/7, = 1/N, N’ being the number of signals to be mult plexed. As N increases, «/T, becomes progressively smaller, and, a is to be seen, from Eq. (54-8), so correspondingly does the output signal. We discuss now an alternative method of recovery of the baseband signal which raises the level of the Signal (without the use of amplifiers which may introduce noise). ‘The ‘method has the additional advantage that rather rudimentary filtering is often ‘uite adequate, but has the disadvantage that some distortion must be accepted. ‘The method is illustrated in Fig 5.61, where the baseband signal mi) and its flat-topped samples are shown. At the receiving end, and after demultiplexing, the sample pulses are extended; that is, the sample value of each individual baseband signal is held until the occurrence of the next sample of that same baseband signal. This operation is shown in Fig 5.61 as the dashed extension of the sample pulses. The output waveform consists then, as shown, of an up and down stairease waveform with no blank intervals. Sams of mie) me Fgare S61 Iustaing the operation 7 ofhol rm, | ie L °C coma Figere 62 ustrating «method of performing he operation of holding ‘A method, in principle, by which this holding operation may be performed is ‘shown in Fig, 16-2. The switch S operates in synchronism with the occurrence of input samples. This switch, ordinarily open, closes somewhat after the occurrence ‘of the leading edge of a sample pulse and opens somewhat before the oocurrence of the trailing edge. The amplifier, whose gain, if any, is incidental to the present discussion, has a low-output impedance, Hence, at the closing of the switch, the capacitor C charges abruptly to a voltage proportional to the sample value, and the capacitor holés this voltage until the operation is repeated for the next sample. In Fig. 56-1 we have idealized the situation somewhat by showing the ‘output waveform maintaining a perfectly constant level throughout the sample pulse interval and its following holding interval. We have also indicated abrupt transitions in voltage level from one sample to the next. In practice, these Voltage transitions will be somewhat rounded as the capacitor charges and discharges, exponentially. Further, if the received sample pulses are natural samples rather than flat-topped samples, there will be some departure from a constant voltage level during the sample interval itself. As a matter of practice however, the sample interval is very small in comparison with the interval between samples, and the voltage variation of the baseband signal during the sampling interval is small ‘enough to be neglected. Ifthe baseband signal is me) with spectral density MU) = Fm], we may deduce the spectral density of the sampled and held waveform in the manner of ‘Sec. $5 and in connection with flat-topped sampling. We need but to consider that the flat tops have been stretched to encompass the entire interval between instantaneous samples. Hence the spectral density is given as in Ea. (5.5-4) except with r replaced by the time interval between samples. We have, then, Fn), sampled and held] — Miu) O2f2fy (561) In Fig. 56-3 we have again assumed for simplicity that the band-limited signal m(2) has a flat spectral density of magnitude Mo. In Fig. 56-3a is shown the spectrum of the instantaneously sampled signal. In Fig. 5.6-3h has been drawn the magnitude of the aperture factor (sin x) (with x = @T/2), while in Fig. $6-3c is shown the magnitude of the spectrum of the sampled-and-held signal. These plots differ from the plots of Fig. 55-2 only in the location of the nulls ofthe factor (sin x)/x. In Fig. 56-3 the first null occurs at the sampling fre- quency Jf. We observe that, as a consequence, the aperture effect, which is responsible for the (sin x)x term, has accomplished most of the filtering which is required to suppress the part of the spectrum of the output signal above thehe 7 Figure 63 () Spectrum of instantaneously sampled signal mt) with me) having deta spectra shown in Fig. 83:22. (b) The magnitude of the apertare eet for. (2) Spectrum of sampled else bandlimit fy. Of course, the filtering is not perfect, and some additional filtering may be required. We also note that, asin the case of flat-top sampling, there will be some distortion introduced by the unequal transmission of spectral eom- ‘ponents in the range 0 to fy. If the distortion is not acceptable, then, as before, it ‘may be corrected by an x/sin x equalizer Most importantly we note in comparing Eq. (56-1) with Eq, (5.5-4) that, aside from the relatively small effect of the (sin x)/x terms in the two cases, the sampied-and-held signal has a magnitude larger by the factor Tyr than the signal of sample duration z. This increase in amplitude is, of course, intuitively to have. been anticipated. 5.7 QUANTIZATION OF SIGNALS ‘The limitation of the system we have been describing for communicating over Jong channels is that once noise has been introduced any place along the channel, wwe are “stuck” with it, We now describe how the situation is modified by sub- jecting a signal to the operation of quantization. When quantizing a signal m(¢) wwe create a new signal m,() which is an approximation to m(?). However, the quantized signal m0) has the great merit that iti, in large measure, separable from the additive noise ‘The operation of quantization is represented in Fig. 5.7-1. Here we contem- plate a signal mt) whose excursion is confined to the range from V, 0 V,. We hhave divided this total range into M equal intervals each of size S. Accordingly S, called the step size, is S=(Vq—¥,VM. In Fig 57-1 we show the specific ‘example in which M = 8. In the center of each of these steps we locate quantiza- fiom levels tg, my... m- The quantized signal m4) is generated in the following. way: Whenever m() is in the range Ao, the signal mt) maintains the constant level mg; whenever mie) isin the range A,, mt) maintains the constant level my: and so on. Thus the signal me) will at all times be found at one of the levels my. im soos mip. The transition in mg() from mls) = mg to m0) = m, is made abrupt. ly when mf) passes the transition level Lo, which is midway between my and my and so on. To state the matter in an alternative fashion, we say that, at every instant of time, mj) has the value of the quantization level to which mi) is closest. Thus the signal m0) does not change at all with time or it makes a quantum jump of step siz S, Note the disposition ofthe quantization levels in the range from V;, to Vi. These levels are each separated by an amount S, but the separation of the extremes V, and Vy each from its nearest quantization level is only 5/2, Also, at every instant of time, the quantization error mit) ~ m,() has a ‘magnitude which is equal to or less than S/2. We see, therefore, that the quantized signal is an approximation to the orig- inal signal. The quality of the approximation may be improved by reducing the size of the steps, thereby increasing the number of allowable levels. Eventually, with small enough steps, the human ear of the eye will not be able to distinguish the original from the quantized signal. To give the reader an idea of the number of quantization levels required in a practical system, we note that 256 levels can be used to obtain the quality of commercial color TV, while 64 levels gives only fairly good color TV performance. These results are also found to be valid when quantizing voice, ‘Now let us consider that our quantized signal has arrived at a repeater some- What attenuated and corrupted by noise. This time our repeater consists of a quantizer and an amplifier. Thete is nvise superimposed on the quantize levels ‘of m4). But suppose that we have placed the repeater at a point on the commu- nications channel where the instantancous noise voltage is almost always less ‘than half the separation between quantized levels. Then the output of the quanti- zet will consist of a succession of levels duplicating the original quantized signal and with the noise removed. In rare instances the noise results in an error in quan- tization level. A noisy quantized signal is shown in Fig. 5.7-2a. The allowable quantizer output levels are indicated by the dashed lines separated by amount S. ‘The output of the quantizer is shown in Fig. 5.7-2b. The quantizer output is the level to which the input is closest. Therefore, as long as the noise has an instanta- neous amplitude less than S/2, the noise will not appear at the output. One instance in which the noise does exceed $2 is indicated in the figure, and, correminds Figure S71 The opetion of quantization. somes “ Figure 7-2 (a gua signal ith ade noise (8) The signal after requantizaion. One instance ‘eteorded in which the nie eve solar hat an eto esl spondingly, an error in level does occur. The statistical nature of noise is such that even if the average noise magnitude is much less than 5/2, thete is always a finite probability that from time to time, the noise magnitude will exceed 5/2. ‘Note that it is never possible to suppress completely level errors such as the one indicated in Fig. 57-2. We have shown that through the method of signal quantization, the effect of additive noise can be significantly reduced. By decreasing the spacing of the repeaters, we decrease the attenuation suffered by mt). This effectively decreases the relative noise power and hence decreases the probability P, of an error in level. P, can also be reduced by increasing the step size S. However, increasing S results in an increased discrepancy between the true signal mit) and the quantized signal m0), This difference m() ~ m4() can be regarded as noise and is called quantization noise. Hence, the received signal is not a perfect replica of the trans- mitted signal m(t). The difference between them is due to errors caused by addi- {ive noise and quantization noise. These noises are discussed further in Chap. 12. 58 QUANTIZATION ERROR It has been pointed out that the quantized signal and the original signal from ‘hich it was derived differ from one another in a random manner. This difference or error may be viewed as a noise due to the quantization process and is called quantization error. We now calculate the mean-square quantization error &, where eis the difference between the original and quantized signal voltages, Let us divide total peak-to-peak range of the message signal mit) into M equal voltage intervals, each of magnitude $ volts. At the center of each voltage interval we locate a quantization level my, mz, .... my as shown in Fig, $8-la. ‘The dashed level represents the instantancous value of the message signal mi) at a time 1. Since, in this figure, mi) happens to be closest to the level my, the quantizer output will be m, the voltage corresponding to that level. The error is = mt) — my.Se} o Fare S84 (0) A rage of voltage over which gal mi makes excursions divided into M quan- Santon ranges eich of iz 8 The uuantzaton lvl ar acted the enter the rege re ‘210 voltage) a5 fonction ofthe nstantaeous value of the signal ml LJim dn be he probity that mi) Hes inthe voltage range m-— dmv? to m+ dm/2. Then the mean-square quantization error is se ranas ‘mn Fo [mina [stn nt dn 89 EO Sy eee eh we se yoyeyits Sa ae ee im) =f", etc, We may now remove {", f®, etc, from inside the integral sign. I we make the substitution x = m — m,, Eq. (5.8-1) becomes ane Bay sso | tae meses (58-20) = (s5 +92 4224S S95 (58-28) ‘Now /(S is the probability that the signal voltage m() willbe inthe frst quanti- zation range, "5 is the probability that m is in the second quantization range, etc. Hence the sum of terms in the parentheses in Eq, (S.8-2b) has a total value of unity. Therefore, the mean-square quantization error is (58:3) 59 PULSE-CODE MODULATION [A signal which is to be quantized prior to transmission is usually sampled as well ‘The quantization is used 10 reduee the effects of noise, and the sampling allows us to time-division multiplex a number of messages if we choose to do s0, The com= bined operations of sampling and quantizing generate a quantized PAM wave- form, that is, a tain of pulses whose amplitudes are restricted to a number of discrete magnitudes. We may, if we choose, transmit these quantized sample values directly. Alter- natively we may represent each quantized level by a code number and transmit, the code number rather than the sample value itself. The merit of so doing will be developed in the subsequent discussion, Most frequently the code number is con- verted, before transmission, into its representation in binary arithmetic, ic, base-2 arithmetic. The digits of the binary representation of the code number are transmitted as pulses. Hence the system of transmission is called (binary) pulse- code modulation (PCM). We review briefly some elementary points about binary arithmetic. The binary system uses only two digits, 0 and I. An arbitrary number NV is represent ced by the sequence .. kz yky,in which the k's are determined from the equation Naot hyd? + ky2t + hy 2? (59-1) with the added constraint that each k has the value 0 or 1. The binary represent- ations of the decimal numbers 0 to 15 are given in Table 59-1. Observe that to represent the four (decimal) numbers 0 to 3, we need only two binary digits ky and kp. For the eight (decimal) numbers from 0 to 7 we require only three binary Places, and so on. In general, if M numbers 0,1, .., M — 1 are to be represented, then an N binary digit sequence ky “*~ ky is required, where M = 2% ‘The essential features of binary PCM are shown in Fig, $9-1. We assume that the analog message signal m() is limited in its excursions to the range from =4 10 #4 volts. We have set the step size between quantization levels at 1 vot. Eight quantization levels are employed, and these are located at —3.5, —2.5,..-. +35 volts. We assign the code number 0 to the level at — 3.5 volts, the code number 1 to the level at —25 volts, etc, until the level at +3.5 volts, which is assigned the code number 7. Bach code number has its representation in binary arithmetic ranging from 000 for code number 0 to 111 for code number 7. In Fig. 59-1, in correspondence with each sample, we specify the sample value, the nearest quantization level, and the code number and its binary rep-~ on eno-c- eon o-or 7 . 5 ‘ 2 1 Simplevwiwe 13 Newest quantatontel 18 8828 05 -35 Code number 5 7 ‘ ‘ 2 ° Bray morertion It ott 00 Figure $91. message signal is cegulriy sampled. Quantstion levels are indicated, For each sample the quantize valu given andi ina representation is indicated resentation. If we were transmitting the analog signal, we would transmit the sample values 1.3, 36, 2.3, ete. If we were transmitting the quantized signal, we would transmit the quantized sample values 1.5, 3, 25, ete, In binary PCM we transmit the binary representations 101, 111, 110, ete. 5.10 ELECTRICAL REPRESENTATIONS OF BINARY DIGITS AAs intimated in the previous section, we may represent the binary digits by elec- trical pulses in order to transmit the code representations of each quantized level cover a communication channel. Such a representation is shown in Fig. 5.10-I Pulse time slots are indicated at the top of the figure, and, as shown in Fig. 5.10- la, the binary digit 1 is represented by a pulse, while the binary digit 0 is rep- resented by the absence of a pulse. The row of three-digit binary numbers given in Fig. 5.10-1 isthe binary representation of the sequence of quantized samples in Fig, 59-1, Hence the pulse pattern in Fig. 510-12 is the (binary) PCM waveform that would be transmitted to convey to the receiver the sequence of quantized samples of the message signal m(?) in Fig. 59-1. Each three-digit binary number that specifies a quantized sample value is called a word, The spaces between ‘words allow for the multiplexing of other messages. At the receiver, in order to reconstruct the quantized signal, all that is required is that a determination be made, within each pulse time slot, whether a pulse is present or absent. The exact amplitude of the important. There is an advantage in making the pulse width as wic since the pulse energy is thereby increased and it becomes easier to recognize a pulse against the background noise, Suppose then that we eliminate the guard time r, between pulses. We would then have the waveform shown in Fig. 5.10-1b, We would be rather hard put to describe this waveform as either a sequence of positive pulses or of negative pulses. The waveform consists now of a sequence of ‘ransitions between two levels. When the waveform occupies the lower level in a ele strarer ae yor 111 43@ 100-011 001000 uo h An 7 A Figure $1041 (c) Puts representaton of the Mnary numbers weed to code the stmples in Fig. 591 (by Representation by voltage levels aber than poleparticular time slot, a binary 0 is represented, while the upper voltage level rep- resents a binary I Suppose that the voltage difference of 2 volts between the levels of the waveform of Fig. 5.10-1b is adequate to allow reliable determination at the re- ceiver of which digit is being transmitted. We might then arrange, say, that the waveform make excursions between 0 and 2V volts or between —V volts and +¥ volts, The former waveform will have a de component, the latter waveform will not, Since the de component wastes power and contributes nothing to the reliability of transmission, the latter alternative is preferred and is indicated in Fig. 510-16, Sl THE PCM SYSTEM The Encoder ‘A PCM communication system is represented in Fig, $.11-1. The analog signal ‘ntt)is sampled, and these samples are subjected to the operation of quantization The quantized samples are applied to an encoder. The encoder responds to each such sample by the generation of a unique and identifiable binary pulse (oF binary level) pattern. In the example of Figs. $9-1 and $.10-1 the pulse pattern happens to have a numerical significance which is the same as the order assigned to the quantized leveis. However, this feature is not essential. We could have assigned any pulse pattern to any level. At the receiver, however, we must be able to identify the level from the pulse pattern, Hence it is clear that not only does the encoder number the level, it also assigns to it an identification code. ‘The combination of the quantizer and encoder in the dashed box of Fig. $.11-1 is called an analog-to-digital converter, usually abbreviated A/D con- verter. In commercially available A/D converters there is normally no sharp dis- tinction between that portion of the electronic circuitry used to do the quantizing and that portion used t0 accomplish the encoding, In summary, then, the A/D. converter accepts an analog signal and replaces it with a succession of code symbols, each symbol consisting of a train of pulses in which each pulse may be interpreted as the represcmtativa of a digit in an arithmetic system, ‘Thus the signal transmitted over the communications channel in a PCM system is referred toas a digitally encoded signal, sone i Te fore enn em seston Figere S114 A PCM communication stem, od Seer The Decoder When the digitally encoded signal arrives at the receiver (or repeater), the first ‘operation to be performed is the separation of the signal from the noise which has been added during the transmission along the channel, As noted previously, separation ofthe signal from the noise is possible because of the quantization of the signal. Such an operation is again an operation of requantization: hence the first block in the receiver in Fig 5.11-1 is termed a quantizer, A feature which eases the burden on this quantizer is that for each pulse interval it has only to make the relatively simple decision of whether a pulse has or has not been received or which of two voltage levels has occurred. Suppose the quantized sample pulses had been transmitted instead, rather than the binary-cncoded codes for such samples. Then this quantizer would have had to have yielded, in each pulse interval, not a simple yes or no decision, but rather a more compli- cated determination about which of the many possible levels had been received. In the example of Fig. 5.10-1, if a quantized PAM signal had been transmitted, the receiver quantizer would have to decide which of the levels 0 to 7 was trans mitted, while with a binary PCM signal the quantizer need only distinguish between two possible levels. The relative reliability of the yes or no decision in PCM over the multivalued decision required for quantized PAM constitutes an important advantage for PCM, ‘The receiver quantizer then, in each pulse slot, makes an educated and sophisticated estimate and then decides whether postive pulse or a negative pulse was received and transmits its decisions, in the form of a reconstituted or regenerated pulse train, to the decoder. (If repeater operation is intended, the regenerated pulse train is simply raised in level and sent along the next section of the transmission channel} The decoder, also called a digital-to-analog (D/A) con- verter, performs the inverse operation of the encoder. The decoder output is the sequence of quantized multilevel sample pulses. The quantized PAM signal is now reconstituted. It is then filtered to reject any frequency components |ying ‘outside of the baseband. The final output signal m'@) is identical with the input ‘m{t) except for quantization noise and the occasional error in yes-no decision ‘making at the receiver due to the presence of channel noise. 5.12 COMPANDING Referring again to Figs. 5.7-1 and 58-1 let us consider that we have established quantization process employing M levels with step size S, the levels being estab- lished at voltages to accommodate a signal mt) which ranges from a low voltage V;, to a high voltage V,. We can readily see that ifthe signal mi) should make excursions beyond the bounds Vy and V,, the system will operate at a disadvan- tage. For, within these bounds, the instantaneous quantization error never exceeds 45/2 while outside these bounds the error i larger. Further, whenever m(?) does not swing through the full available range the system is equally at a disadvantage. For, in order that m,() be a good approximation to mt) itis necessary that the step size S be small in comparison to the ‘range over which mdi) swings. As a very pointed example of this consideration ‘consider a case in which mi) has a peak-to-peak voltage which is less than $ and never crosses one ofthe transition levels in Fig. .7-1. In such a ease m() wll be fixed (de) voltage and will bear no relationship to mit). To explore this latter point somewhat more quantitatively let us consider that mi) is a signal, such as the sound signal output of a microphone, in which Vu=—Vi=V, ie, a signal without de components, and with (atleast approximately) equal positive and negative peaks. Further, for simplicity, let us assume that in the range +V the signal m(t) is characterized by a uniform prob. ability density. The probability density is then equal to 1/2V and the normalized Average signal power ofthe applied input signal is oot wa = mo dm = (20) The quantization note, a given by Bg, (58-36 2 nS (12) H the umber of quantation levels M, then MS = 2V so that Ms vas (123) 2 Combining Eqs. (5.12-1), (5.12-2), and (5.12-3) we have that the input signal-to- quantization noise power ratio is ae (5124) Eventually the received quantized signal will be smoothed out to generate an ‘output signal with power S,. If we have a useful communication system then pre- sumably the effect of the quantization noise is not such as to cause an easily per. ceived difference between input and output signal. Tu such a case the output ower S, may be taken to be the same as the input power ie, 8, ~ 5, so that finally we may replace Eq, (5.12-4) by (512-5) If there are M quantization levels then the code which singles out the closest ‘quantization to the signal m(t) will have to have N bits with M = 2". Hence Bg, (512-5) becomes Seams oom Brera? 612.6) In decibels we have [elem Equation (5.12-7) has the interpretation that, in a system where the signal is ‘quantized using an N-bit code (ic, the number of quantization levels is 2"), and ‘where the signal amplitude is capable of swinging through all available quamtiz tion regions without extending beyond the outermost ranges, the output signal- to-quantization noise ratio is 6N AB, In voice communication we use N= 8. corresponding to 256 quantization regions and S,/Ng = (68) = 48 dB. If the signal is reduced in amplitude so that not all quantization ranges are used then, SJNo becomes smaller, since No, depending as it does only on step S is not affected by the amplitude reduction, while S, is reduced. For example, if the amplitude were reduced by a factor of 2, the power is then reduced by a factor of 4 reducing the signal-to-noise ratio by 6 dB. Iti interesting to observe that the elfective number of quantization levels is also reduced by a factor of 2. Corre spondingly, the number N of code bits is reduced by 1. In summary, the depen: dence of S./Ng on the input signal power S, is such that as the number of code bits needed decreases, S,/Ng decreases by 6 dB/bit. It is generally required, for acceptable voice transmission, that the received signal have a ratio Sy/Ng not less than 30 dB, and that this minimum 30 dB. figure hold even though the signal power itself may vary by 40 dB. (The signal, in this case, is described as having a 40 dB dynamic range) In an &-bit system, at ‘maximum signal level we have 5,/Ng © Si/Ng = 48 dB. Now No is fixed, and depends only on step size. If we are\to allow S/N to drop to no lower than 30 dB then the dynamic range would be restricted to 48 — 30 = 18 dB. ‘The dynamic range can be materially improved by a process called com panding (a word formed by combining the words compressing and expanding). AS wwe have seen, 10 keep the signal-to-quantization noise ratio high we must use a signal which swings through a range which is large in comparison with the step size, This requirement is not satisfied when the signal is small. Accordingly before applying the signal to the quantizer we pass it through a network which has an input-output characteristic as shown in Fig. 5.12-1, Note that at low amplitudes the slope is larger than at large amplitudes. A signal transmitted through such a network will have the extremities ofits waveform compressed. The peak signal which the system is intended to accommodate will, as before, range through all available quantization regions. But now, a small amplitude signal will range through more quantization regions than would be the case in the absence of com- pression. Of cours, the compression produces signal distortion. To undo the dis (ortion, at the receiver we pass the recovered signal through an expander network. An expander network has an input-output characteristic which is the inverse of the characteristic of the compressor. The inverse distortions of com- pressor and expander generate a final output signal without distortion, ‘The determination of the form of the compression plot of Fig. 5.12-1 is a somewhat subjective matter. In the United States, Canada, and Japan a yelaw 10 logio 22° = 6N (512-7)Figure S121 An input-output charactrinic which provides compression compandor is used (see Prob. 5.12-1) and it differs somewhat from the A-law compander (see Prob, 5.12-2) used by the rest of the world. The “” and the "A" refer to parameters which appear in the equations for the compression and expansion characteristic, In implementing the compression characteristic, the analog signal mi) is left ‘unmodified and, instead, the step size is tapered so that the quantization levels are close together at low signal amplitudes and progressively further apart in regions attained as the signal inereases in amplitude, To see one way in which the step sizes are altered consider again an &-bit PCM system employing 2" — 256 Quantization levels. In uch a system, the A/D converter shown in Tig. 511-1 would ordinarily be an bit converter. Each input sample would generate an 8-bit output code identifying the closest quantization level. Ifthe total range of the input was V then the step size would be $ = 2V/28. Let us start however with @ 12-bit A/D converter so that the step size is 2/2" This 12-bit PCM signal is not applied to the communication channel directly but is instead applied to the address pins of a read-only memory (ROM) whose content is to be Aescribed. The ROM has 12 address pins and § output data pins. The signal transmitted isthe 8-bit data output of the ROM. The content of the ROM is as follows: As shown in Fig. 5.12-2, in each sue- ‘cessive memory location in the region where the step size is to be smallest (ve, step size = 2¥/2'? = A) there are written successive 8-bit code words. Hence in this region a change in analog signal of step size A will change the code word Presented for transmission by the amount A. This one-to-one correspondence between addresses and transmitted code applies as shown for the middle 64 addresses. For the next 64 addresses (32 on one side and 32 on the other side of the original 64) we arrange that at the memory locations specified by pairs of addresses, ic, for two adjacent addresses the same code word is written into the ‘memory. Hence the transmitted code word will change at every second addresses ‘change and correspondingly the step size is 2A. Next we arrange that for the next 12.4 input codewords 8. out Codewonds 76 ourpuT cove ors 16 ouTPUT, cove WoRDS: Figure 12-2 ROM characteristic for compression128 addresses (64 above + 64 below) the same code word is written into the loca- tions of four successive memory locations so that the step size is 4A, We proceed in this manner, each time doubling the number of successive memory locations Which contain the same code word. As can be verified, when we have used all 21? ss 4096 addresses we shall have generated 2° = 256 code words. At the re- ceiver we have a mechanism for generating the quantization corresponding t0 ‘each code word Aa ~ 40 96 —32 28-24-20 —16 12 — Baro} Figwe £123 Comparison of companded and uncompanded systems It is interesting to observe that for the smallest 64 levels, the input and ‘output signals are the same, ie, small signals are nor companded, One reason for this is that signals other than voice are often digitized using the same PCM system employed for voice. If a non-voice signal is subjected to the compression. algorithm the result is usually a degradation of performance, Therefore, to avoid this possibility, such non-voice signals are kept 40 dB below the peak level of a voice signal. In conclusion, we refer to Fig. 5.12-3 which compares the variation of output signal-to-noise ratio as a function of input signal power when companding is used, to the case of an uncompanded system. Note that the companded system, hhas a far greater dynamic range than the uncompanded system and that theoreti- cally the companded system has an output signal-to-noise ratio which exoveds 30 dB over a dynamic range of input signal power of 48 dB, while the uncom: panded system has a dynamic range of 18 dB for the same conditions. It should be noted however, that the penalty paid using companding is approximately 10 dB. Thus an 8-bit uncompanded system, when operating at maximum ampl- tude, produces a signal-lo-noise ratio of 48 dB. The same 8-bit system, using a compandor, yields a 38 dB SNR. 5.13 MULTIPLEXING PCM SIGNALS* We have already noted the advantage of converting an analog signal into a PCM waveform when the signal must be transmitted over a noisy channel. When large number of such PCM signals must be transmitted over a common channel, ‘multiplexing of these PCM signals is required. In this section we discuss the ‘multiplexing methods for PCM waveforms used in the United States by the ‘common cartiers (AT&T, GTE, ete) 5.13.1 The TI Digital System Figure 5.13-1 shows the basic time division multiplexing scheme, called the T1 Aigital system, which is used to convey multiple signals over telephone lines using wideband coaxial cable. ft accommodates 24 analog signals which we shalt reer to as, through s,,. Each signal is bandiimited to approximately 3.3 kHz and is sampled at the rate 8 kHz which exceeds, by a comfortable margin, the Nyquist rate of 2 x 33=66 kHz Each of the time division multiplexed signals (still analog) is next A/D converted and companded as described in Sec. 5.12. The resulting digital waveform is transmitted over a coaxial cable, the cable serving to ‘minimize signal distortion and serving also to suppress signal corruption due to noises from external sources. Periodically, at approximately 6000 fc intervals, the signal is regenerated by amplifies called repeaters and then sent on toward its ‘eventual destination. The repeater eliminates from each bit the effect of the dis tortion introduced by the channel, Also, the repeater removes from each bit any superimposed noise and thus, even having received a distorted and noisy signal, itAmpitude 3 Peo ise Pach os} generator | | sttitor Adjusaie | ""Spnoch ‘iter | ee a Fiore $19-2 A simple ines predictive decoder Pitch signals, These are used in connection with the modulator, and pulse and ois generators, as at the encoder, to provide an input to the adjustable filter. ‘The flter-parameter adjusting signals are also received and are used, as at the encoder, to adjust the filter characteristics for optimum voice regeneration ‘Not explicitly shown in Figs. 519-1 and 5.19-2, but nonetheless to be under- Stood is that, as in the simpler vocoder of Fig, .18-1, the transmitted signal must be the time-division multiplex of the individual signals to be transmitted. Typi- cally, 18 fiter-adjusting signals a, are employed. If, as before we sample at the rate 40 samples/s and encode each sample value into three bits, the bit rate Ris R=(18-+ 3)signals x 40 MEPIS 5 pitscample signal = 252 kbjs (519-1) REFERENCES. Lusky, RW 3. Sale and FJ. Weldon, Jt: “Principles of Data Communieation,” pp. 61-87, “McGraw-Hill Book Company, New York 196 * Losky, RW. J Saiz and. Welon, Je: "Prins of Data Communication: pp. 128-168, MeGiram-Hil Book Company, New York, 1968 Bal Telephone Laborato: “Telecommunications Transmission Enginering” Vol Il, AT&T, Western Flere Company. Teh, Pub Winton Solem, Nie ITT Jagr, F-:Detumodulation: A Method of PCM Transmision Using a I-unit Code, Pips Res Rep 7p. 442-46, 1952 Jayant. N-S, and F. Noll “Dipl Coding of Waveforms” Prete Hall In, Eaglewood Ci, | | PROBLEMS ‘S11 1s required to transmit telephone messages across the United States, 2 3000 mile run. The Sienal evel snot to te allowed to drop flow I milvl before amplistion and the sgn! moo be allowed to be larger than 15 sls order to avoid amplfer overload Assuming tha repeaters aye to e located with equal spacings, how many repeater willbe req? S12 A undpass signal has aspect range that extends fom 20 to $2 kx Find the acceptable range ofthe sampling frequency f ‘S13. A bandpass signal has 2 center Fegueney f and extends fom fo $ KHz (0 fo + KH. The signal is samplad at arate f,~ 25 His. Ax the center fequeny fy wai lom fg = Ske tafe = 50 [tt find the ranges of fr mich the sampling rate adequate S14 The signal) = os Sm + 05 cos 10K is instantaneously sampled. The interval betwen samples, (o) led he maximum allowable vale fo (6) tthe sampling signal i Se) = 8 F7.. Ae~ 0.1 the sampled signal rf) ~ HS) com sits of a wai of mpi cach wh lifeent irength oft) = fee fy At O14) Fd Tyg and (6) To reconstrct the signal e()& pated through a rectangslar low-pass filer. Find the ‘inimam filter bandh to reconstruc the signal without dsorton SILS. We have the signal ui) = cos 2fy1 + co 2x 2jgt + 6083 2yfyt, One intrest extends, however only to spectral component upto and insiding 2f. We therfore sample a he rae Sf ich s adequate forthe 2 component ofthe signal. {o) sampling i accomplished ty malping by am impulse rin in which the implses se of uit stength, wrt an expression for the sampled sgl (8) To recover the part ofthe signal of interest, the sampled signal is pase through «reo angular lowpass fer with passband extending fom 0 f0 sighy beyond 2 Wate an expretion forthe fer opus the par ofthe signal of intecest covered enaiy? we wan! fo eprde the first two terms of (without distortion, what operation mus! be performed athe very out?” ‘S14, The bandpas signal = cs 1Ovgt +605 Itigt + co 1249! sampled bya imple tain So) Sse t= HT. (@) ied he maximam ime between samples, to entre repeiction without ero, (0) Using the est cbtaind na obtain at expression or») = St) (©) The sampled signal fe 6 fltered by a restanguler low-pass iter with © bandwith B= 2 Obtain an expression for the er output (a) The sampled signals) ier by 4 rectangular bandpass filter extending from Yo to ‘y- Obtain am expression forthe fier output. 51-7. The bandpass signal) = cor 1004 ¢ 4698 Tey t 608 12a is sampled by an impulse SEAM. Te me ia ft) = Sn en by rests ‘S14. Let us view the waveform «() = 008 yt a8 4 bandpass sen occupying an arbitrary narrow frequeney band On this ass we nd that the required samping rate, 0 Discus, ‘52-1. The TDM system shown in Fig. 52-1 i used to multiplex the four signals m0) = 28 ey rma = OS cos ogi my) = 2 cot Bagh and ma) = 8 dt (@) IF each signal i sampled a the Same sampling rate, aeulate the minimom sampling mie. (0) What isthe commutator sped in revolutions pe second (0) Design a commutator which wl allow each ofthe four signals toe sampled a «rte fuser hans requited satis the Nga enteron forthe inv signal, $22. Three signals mm, abd m, ate tobe multiplexed m, and m have « KHz bandwidth, and my has a 104Hr bandwith. Design a commutator switching sem 5 tha each sina is amped at Ra Nyquist ate,SSM. Show thatthe response ofa rectangular low-pass fir, with a bandwith, 10 the impube function 1 t~ Rin ty sin nt — WY = att Wf) Acsume hatin ite passband the ers HT) ~ S32. Four signal, m= 1 om 298, ml) = Un gt, ml) — Leos yt and mul) —t sin (me ae Sampled every 12 se by the saping onion sn 4) The signa ate then timesdivsion multiplexed. The TDM signi fered by &rectangulrlon-pss fer having mancwidth f= Aad then deommtated. (a) Sketch the four outpts of the decomtator, (6) Each ofthe four output igal iter bya rectangular low-pass filer having band wih Show that the four signal are reconstrcted without ron ‘533. The fur signals of Prob. $3-2 ze seme, indicated in that problem, and timedivson ‘molptete. The TDM signal is red by rectangular low-pass ler bevng» bandwidth. = 3 A then decommutated. Sketch the outpat atthe deoommutator switch segment where the samples ‘of m()should appear and show that mi) cant be revered S41 The signal t= cos +008 Foyt amped by using narra sompting. (o} Determine the minim sampling rat (0) Sketch ef) = Shs 0) train of ples having unit height, ecuring at the rte and Si) = foe nT ~ 72 £1 = + FP, cos nt 4 bum) ma togie lve logic level volige Figee 621 Means of generating a DPSK signalFigure 63.2 Logic waveforms titrate the response Mit an input ‘and with this table we can easly verify that the waveforms for dt), Mt — 7), and ‘b{a) are consistent with one another, We observe that, as required, Mt — 7) is indeed 2) delayed by one bit time and that in any bit interval the bit tis given by Kt) = dle) @ He ~ T). In the ensuing discussion we shall use the symbolism ‘a{k) and H() to represent the logic levels of dt) and (0) during the kth interval Because of the feedback involved in the system of Fig. 63-2 there is a dif culty in determining the logic levels in the interval in which we start to draw the ‘waveforms (interval | in Fig. 63-2), We cannot determine bt) in this fist interval of our waveform uniess we know ik = 0). But we cannot determine 0) unless wwe know both d(0) and H(—1), ete. Thus, to justify any set of logic levels in an initial bit interval we need to know the logic levels in the preceding interval. But such a determination requires information about the interval two bit times earlier and 0 on. In the waveforms of Fig. 6.3-2 we have circumvented the problem by arbitrarily assuming that in the fist interval (0) = 0. It is shown below that in the demodulator, the data will be correctly determined regardless of our assump- tion concerning HO). We now observe that the response of Hi) to d() is that He) changes level at the beginning of each interval in which d(t)= 1 and Hf) docs not change level when d(q) = 0, Thus during interval 3, d3) = 1, and correspondingly b(3) changes at the beginning at that interval, During intervals 6 and 7, d(6) = d(7) = 1 and. there are changes in Ht) at the beginnings of both intervals, During bits 10, 11, 12, id 13 a(t) = I and there are changes in H() at the beginnings of each of these intervals. This behavior is to be anticipated from the truth table of the exclusive~ OR gate. For we note that when dit) = 0, He) = H(¢ ~ T,) so that, whatever the initial value of ~ Tp, it reproduces itself. On the other hand when dit) = 1 then t) = At — T). Thus, in each successive bit interval 6() changes from its value in the previous interval. Note that in some intervals where at) = 0 we have ‘¢)= 0 and in other intervals when a{e) = 0 we have H(t) = I. Similarly, when 4 = | sometimes Ho) = 1 and sometimes Ho) =O. Ths there i no correspon dence between the levels of de) and A, and the only invariant feature of the system is that a change (sometimes wp and sometimes down) in hi) occurs whens ever di) ~ I, and that nochange in wil occur whenever d) = Finally, we note thatthe waveforms of Fig. 62 are drawn on the asomp- Aion that interval I, AO) =O Ass easly vera if not intuitively apparent tte had astimed 0) =I, the invariant feature by which we have characterized ‘the system would continue to apply. Since h(0) must be either (0) =0 or 140) 1 tere being no ater postin, our res is raid quite generally. I homer, we hal stared with BO) = the levels 0) and 0) would have been inverted ‘As is seen in Fig. 63:1 Ae) i api to balanoed modulator to which i also applied the carrier /2P, cs ay. The modulator output, which isthe rans titted signal is Sora) = 0) PP, cos ot = 4 JP, 008 ot ean “Thus altogether when di) =0 the phase of the cari does nor change atthe beginning of the bit interval while when i) = I there isa phase change of mag mie A method of recovering the data bit team from the DPSK signal is shown in Fig 63-3 Here the received signal and the received signal delayed by the bit time ae applied toa moti. The mliper output i HOWE ~ TAP, 00s (apt + 6 cos Loult ~ ) + 8) = nome Tor fom 0 +008 [204{+—B) + 20]} (6229 and is applied to a bit synchronizer and integrator as shown in Fig, 62-1 for the BPSK demodulator. The first term on the right-hand side of Eq. (6-2) is, aside from a multiplicative constant, the waveform H()¢ ~ TZ) which, as we shall se is precisely the signal we require. As noted previously in connection with BPSK, and so here, the output integrator will suppress the double freauency term, We should select «% % 50 that 0 Ty = 2nx with nm an integer. For, in this ease we shall have cos 9 % = +1 and the signal output will be as large as possible. HiJBF, cos toot +0) Schone Tolintearator ond (eruption [—bireynenonize oa ton Oey i Figure 63.3 Method of covering data rom the DPSK signal WTF, cos (nde ~ 7) + 8)Further, with this selection, the bit duration encompasses an integral number of lock cycles and the integral ofthe double-frequency term is exactly zero. The transmitted data bit do) can readily be determined from the product o{o1Mt — Ti). I le) = 0 then there was no phase change and B(t) = He — 7) both being +1V or both being —1V. In this case Wat — Ty) = 1. If however, (a) == 1 then there was a phase change and either Me) =1V with it~) IV or vice versa, In either case AIC — Ty) = —1 The differentially coherent system, DPSK, which we have been describing. hhas a clear advantage over the coherent BPSK system in that the former avoids the need for complicated circuitry used to generate a local carrer atthe receiver, To see the relative disadvantage of DPSK in comparison with PSK, consider that during some bit interval the received signal is so contaminated by noise that ina PSK system an error would be made in the determination of whether the transmitted bit was a 1 or a 0. In DPSK a bit determination is made on the basis ofthe signal received in two successive bit intervals. Hence noise in one bit inter- val may cause errors to two bit determinations, The error rate in DPSK is there- fore greater than in PSK, and, as a matter of fact, there is a tendency for bit errors to occur in pairs. It is not inevitable however that errors occur in pairs, Single errors are still possible. For consider a ease in which the reccived signals in th and (k + I)st bit intervals are both somewhat noisy but that the signals in the (k— pst and (k-+ 2ind intervals are noise free. Assume further that the kth inter val signal is not so noisy that an error results from the comparison with the (&— Ist interval signal and assume a similar situation prevails in connection with the (K + 1st and the (k + 2)nd interval signals. Then it may be that only single error will be generated, that error being the result of the comparison of the kth and (k + 1st interval signals both of which are noisy. 64 DIFFERENTIALLY-ENCODED PSK (DEPSK) As is noted in Fig. 63-3 the DPSK demodulator requites a device which operates at the carrier frequency and provides a delay of T,. Differentally-encoded PSK climinates the need for such a piece of hardware. In this system, synchronous demodulation recovers the signal (0), aad the decoding of bf) to generate att 8 done at haseband, « “To abD— 4) = HOB KET) LY % » Figure 644 Rasehand decoder to obtain) fom Mi (tne ww: 01101100 Me: 01101100 ® AL)=HRVOME—1 1.011010 ondenor mm o1tt1100 N se-m — arnsa100 ® a= wome-—1 1000010 “ a Figure 64.2 Errors in diferent two ore encoded PSK occur in pis, ‘The transmitter of the DEPSK system is identical to the transmitter of the DPSK system shown in Fig. 63-1. The signal 2) is recovered in exactly the ‘manner shown in Fig. 62-1 for a BPSK system. The recovered signal is then applied directly to one input of an exclusive-OR logic gate and to the other input is applied e — 7, (see Fig. 64-1). The gate output will be at one or the other of its levels depending on whether A) = B(¢ — 1) or Ht) = A(t — Ty). In the first case ‘did not change level and therefore the transmitted bit is a(t) =0. In the second ease dt) = 1 We have seen that in DPSK there is a tendency for bit errors to occur in pairs but that single bit errors are possible, In DEPSK errors always occur in pairs. The reason for the difference is that in DPSK we do not make a hard deci- sion, in each bit interval about the phase of the received signal, We simply allow the received signal in one interval to compate itself with the signal in an adjoin- ing interval and, as we have seen, a single error is not precluded, In DEPSK, a firm definite hard decision is made in each interval about the value of bt), If we ‘make a mistake, then errors must result from a comparison with the preceding and succeeding bit. This result is illustrated in Fig. 64-2. In Fig, 64-20 is shown the error-free signals Wk), Mke— 1) and d(l) = ML) @ Hk — 1). In Fig. 64.26 we hhave assumed that 5(k) has a single error. Then b(k — 1) must also have @ single error. We note that the reconstructed waveform d(k} now has two errors, 65 QUADRATURE PHASE-SHIFT KEYING (QPSK) We have seen that when a data stream whose bit duration is 7; is to be transmit- ted by BPSK the channel bandwidth must be nominally 2f, where f, = 1/7. Quadrature phase-shift keying, as we shall explain, allows bits to be transmitted using half the bandwidth, In describing the QPSK system we shall have occasion to use the type-D fip-flop as a one bit storage device, We therefore digress, very briefly, to remind the reader of the essential characteristics of this flip-flop.Type-D flip-top, The type-D fip-lop represented in Fig. 65-1a has a single data input terminal (D) to which a data stream a(t) is applied. The operation of the Aipxiop is such that at the “active” edge of the clock waveform the logic level at D is transferred to the output Q. Representative waveforms are shown in Fig. 65-1(0) We assume arbitrarily thatthe negative-going edge ofthe clock waveform is the active edge. At the active edge numbered 1 we find that dtt) = 0. Hence, after a short delay (the delay is not shown) from the time of occurrence of this edge we shall find that Q=0. The delay results from the fact that some time is required for the input data to propagate through the flip-flop to the output Q. (On the basis of the waveform d(f) shown, we have no basis on which to determine Q at an earlier time) At active edge 2 it appears that the clock edge occurs precisely at the time when dio) is changing: If such were indeed the case, the response of the flip-flop would be ambiguous. As a matter of practice, the change in d() will occur slightly after the active edge. Such isthe case because, rather inevitably, the change in dt) §s itself the response of some digital component (gate, flip-Mop, etc) to the very same clock waveform which is driving our type-D flip-flop. (The delays referred to are normally not indicated on a waveform diagram as in Fig. 65-1 because these delays are ordinarily very small in comparison with the bit time T,.) In any event, the fact is that at active edge 2 we have d(t) = 0 and Q remains at ‘The remainder of the waveforms are easily verified on the same basis. Observe that the Q waveform isthe a(t) waveform delayed by one bit interval 7, The rele. ‘ant point about the flip-flop in the matter of our present concern is the follows ano—o 7 o Flee 65:1 (3) Tyre fip-op symbol. (b) Waveforms showing fip-op chaacteriticn ing: Once the flip-flop, in response to an active clock edge, has registered a data bit, it will hold that bit until updated by the occurrence of the next succeeding active edge. QPSK Transmitter The mechanism by which a bit stream b(?) generates a QPSK signal for transmis- sion is shown in Fig, 6.5-2 and relevant waveforms are shown in Fig. 65-3. In these waveforms we have arbitrarily assumed that in every case the active edge of the clock waveforms is the downward edge. The toggle flip-flop is driven by a clock waveform whose period isthe bit time T,. The togale flip-flop generates an ‘odd clock waveform and an even waveform, These clocks have periods 27. The active edge of one of the clocks and the active edge of the other are separated by the bit time 7,. The bit stream Ki) is applied as the data input to both type-D flip-flops, one driven by the odd and one driven by the even clock waveform. The flip-Nps register alternate bits in the stream b(t) and hold each such registered bit for two bit intervals, that is for a time 27. In Fig. 65-3 we have numbered the bits in H(7). Note that the bitstream (0) (which isthe output of the Rip-flop driven by the odd clock) registers bit 1 and holds that bit for time 27, then regis- ters bit 3 for time 27, then bit S for 27, etc. The even bit stream b, (1) holds, for times 27, each, the alternate bits numbered 2, 4, 6, ete The bi steam (0 (which, as ease ake to be yf) = 41 vol i super impel on a caries cost and the Bt seam false von no | O%Bioe (e) (oven VT tetfop [— e/a Ee nn | Pieibe (@) {od9) mae VS Figure 652 An oe QPSK transite.Clock ie mM TL ook Ir Sock vee 4 eee ew 16 7 ns Mo Fee 65.3 Waveforms forthe QPSK trnemiterof Fig. 652 superimposed on a carrer /F; sin ot by the use of two multipliers (i, bale anved modulators) as shown o generate two signals 5) and 3,0) These signals are then added to generat the transmitted outpt signal rg () whichis, VP, bylt) sin at + /P, belt) 608 cot (65-1) As may be verified, the total normalized power of (0s Pr 'As we have noted in BPSK, bit stream with bit time Ty multiplies a carer, the generated signal has « nominal bandwidth 2x 1/7. In the waveforms by () and hy (?) the bit times are each 1/27, hence both s,(0 and s,() have nominal bandwidths which are half the bandwidth in BPSK. Both «,() and s,() occupy the same spectral range but they are nonetheless individually identifiable because ofthe phase quadrature of their carriers When by =I the signal s,(0 = Ps sin gt, and 340) = ~/F, sin ot when =1. Correspondingly, for 6,0) = +h (= 4/P, cos oot. These Tout sianals have been represented as phasors in Fig. 65-8: They are in mutual phase ma FP sino igre 65.4 Phasoe diagram for sinusoids in Fig 62 quadrature. Also drawn are the phasors representing the four possible output signals v9(0) = 5,(0)+ 5,(0) These four possible output signals have equal ampli- tude ./2P, and are in phase quadrature; they have been identified by their corre: sponding Values of b, and b,. At the end of each bit interval (ie. after each time 1) either b, oF b, can change, but both cannot change at the same time. Conse- quently, the QPSK system shown in Fig. 6.2 is called offer or staggered QPSK and abbreviated OQPSK. After each time T,, the transmitted signal, iit changes, changes phase by 90° rather than by 180" as in BPSK, Non-offset QPSK Suppose that in Fig. 65-2 we introduce an additional flip-flop before either the ‘odd or even fip-flop. Let this added flip-flop be driven by the clock which runs at the rate f,. Then one or the other bitstreams, odd or even, will be delayed by one bit interval. As a result, we shall find that two bits which occur in time sequence (ie, serially) in the input bit stream 2) will appear at the same time Ge, in parallel) at the outputs of the add and even fip-lops, In this ease b,(¢) and b(t) can change at the same time, after each time 27,, and there can be a phase change of 180° in the output signal. There is no difference, in principle, between a staggered and non-staggered system, In practice, there is often a significant difference between QPSK. and ‘OQPSK. At each transition time, 7; for OQPSK and 27, for QPSK, one bit for OQPSK and perhaps two bits for QPSK change from IV to —1V or —1V to IV. Now the bits (0) and b,(t) can, of course, not change instantancously and. in changing, must pass through zero and dwell in that neighborhood at least briefly. Hence there will be brief variations in the amplitude of the transmitted waveform. ‘These variations will be more pronounced in QPSK than in OQPSK since in the first case both h,0) and b,() may be zero simultaneously so that the signal ampli- tude may actually be reduced to zero temporarily. There is a second mechanism through which amplitude variations ate caused at the transmitter. In QPSK as in BPSK a filter is used to suppress sidebands. It turns out that when waveforms which exhibit abrupt phase changes, are filtered, the effect ofthe filter, atthe time of the abrupt phase changes, is (© Cause substantial changes again in the ample tude of the waveform. Here too, we expect larger changes in QPSK. where phase changes of 180° are possible than in OQPSK where the maximum phase change is 90" The amplitude variations can cause difficulty in QPSK communication systems which employ repeaters, ic, stations which receive and rebroadcast signals, such as earth satellites, For such stations generally employ output power stages which operate nonlinearly, the nonlinearity being deliberately introduced because such nonlinear stages can operate with improved efficiency. However, precisely because of their nonlinearity, when presented with amplitude variations, they generate spectral components outside the range of the main lobe, thereby undoing the effect of the band limiting filter and causing interchannel interference. Further filtering to suppress the effect of amplitude variation has an effect on the phase of the signal and it is, of course, precisely the phase which conveys the signal message ‘Symbol Versus Bit Transmission In BPSK we dea! individually with each bit of duration 7;. In QPSK. we lump {vo bits together to form what is termed a symbol. The symbol can have any one of four possible values corresponding to the two-bit sequences 00, 01, 10, and 11. We therefore arrange to make available for transmission four distinct signals. At the receiver each signal represents one symbol and, correspondingly. two bits When bits are transmitted, as in BPSK, the signal changes occur at the bit rate When symbols are transmitted the changes occur at the symbol rate which is ‘one-talf the bit rate, Thus the symbol time is T, = 27, ‘The QPSK Receiver AA receiver for the QPSK signal is shown in Fig. 65-5, Synchronous detection is Fequired and hence itis necessary (0 locally regenerate the carriers cos wy and sin wot. The scheme for carrier regeneration is similar to that employed in BPSK. In that eatlier case we squared the incoming signal, extracted a waveform + Pn em yt Toro Ft smog te 2] | WL Tor "suger i es senting i E08 gt Figwe 65.8 A OPSK recsver at twice the carrier frequency by filtering, and recovered the carrier by frequency dividing by vo, In the present case, it is required that the incoming signal be raised to the fourth power after which fitering recovers a waveform at four times the carrier frequency and finally frequency division by four regenerates. the carrier. In the present case, also, we require both sin @9f and cos 1 Its left as a probiem (see Prob. 6.5-3) to verify that the scheme indicated in Fig. 65-5 does indeed yield the required waveforms sin «¢ and 60S 6% 1 ‘The incoming signal is also applied to two synchronous demodulators con- sisting, as usual, of a multiplier (balanced modulator) followed by an integrator. ‘The integrator integrates over a two-bit interval of duration T, = 2% and then dumps its accumulation, As noted previously, ideally the interval 27; = T, should encompass an integral number of carrier cycles. One demodulator uses the cartier c0S wf and the other the carrier sin wp. We recall that when sinusoids in phase quadrature are multiplied, and the product is integrated over an integral number of cycles, the result is zero, Hence the demodulators will selectively respond to the parts ofthe incoming signal involving respectively bi) oF ba(0) Of course, as usual, a hit synchronizer is required to establish the heginnings and ends of the bit intervals of each bit stream so that the times of integration can be established, The bit synchronizer is needed as well to operate the sampling switch, At the end of each integration time for each individual integrator, and just before the accumulation is dumped, the integrator output is sampled. Samples are taken alternately from one and the other integrator output at the end of each bit time 7, and these samples are held in the latch for the bit time Ty. Bach indi- vidual integrator output is sampled at intervals 27. The latch output is the recovered bit stream h(t) ‘The voltages marked on Fig. 65-5 are intended to represent the waveforms of the signals only and not their amplitudes. Thus the actual value of the sample voltages at the integrator outputs depends on the amplitude of the local carrier, the gain, if any, in the modulators and the gain in the integrators. We have hhowever indicated that the sample values depend on the normalized power P, of the received signal and on the duration T, ofthe symbol. The mechanism used in Fig. 6.5-5 to regenerate the local carriers is a source ‘of phase ambiguity of the type described in Sec. 6.4. That is, the carrier may be 180° out of phase with the carriers at the transmitter and as a cesull the denne lated signals may be complementary to the transmitted signal. This situation can bbe corrected, as before, by using differential encoding and decoding as in Figs. 641 and 64-2. ‘Signal Space Represent In Sec. 1.26 we investigated four quadrature signals. Equation (1.26-2) repeated here, is ol) iP, cos[ oot + 0m +] m=0123 652)‘These signals_were then represented in_terms of the two orthonormal signals 44(0 = /2/T) 08 gt and y(t) = /Q/T) sin cot. The result in Eq, (19-22), repeated here is 1 = [ VPFT conto 0g] fem aaa ~ [versions 93] 2 singe 6539 ‘The QPSK signal v4) in Eq, (65-1) can be put in the form of Eq, (6:3) by setting bg V2 cos Om 4 1) 7 (65-4a) and b= ~ JFsin m+ 1 (65-46) Thus va(l) = VE bate) ~ /Eybftust) (65-5) T,. Now Fig. 1.9.9 can be redrawn as shown in Fig. 65-6, to show the geometrical representation of QPSK. The points in signal space corre- sponding to each of the four possible transmitted signals is indicated by dots From each such signal we can recover two bits rather than one, The distance of @ signal point from the origin is ./E, which is the square root of the signal energy associated with the symbol, that is E, AQT). AS we have noted earlier Figte 65.6 The four QPSK signa drawn in sgn space. and will verify in Sec. 11.14, our ability to determine a bit without error is mea- sured by the distance in signal space between points corresponding to the differ- tent values ofthe bit. We note in Fig. 65-6 that points which differ in a single bit are separated by the distance 1/P T= 2/6 (6546 where By is the energy contained in a bit transmitted for a time Ty. This distance for QPSK is the same as for BPSK (see Eq, (62-11). Henee, altogether, we have the important result that, in spite of the reduction by a factor of two in the band- width required by QPSK in comparison with BPSK, the noise immunities of the two systems are the same. 66 M-ARY PSK In BPSK we transmit each bit individually. Depending on whether b() is logic 0 fr logic 1, we transmit one or another ofa sinusoid for the bit time T,, the sinus- ois differing in phase by 2x/2 = 180". In QPSK we lump together two bits Depending on which of the four two-bit words develops, we transmit one oF another of four sinusoids of duration 27,, the sinusoids differing in phase by amount 27/4 = 90°, The scheme can be extended. Let us lump together N bits so that in this N-bit symbol, extending over the time N7, there are 2% = M possible symbols. Now let us represent the symbols by sinusoids of duration NT, = T, Which difer from one another by the phase 2x//M. Hardware to accomplish such ‘Mary communication is available. ‘Thus in M-ary PSK the waveforms used to identity the symbols are rat) BP, €08 (gt + bq) (m=O Lea M= 1) (6641) with the symbol phase angle given by Om 40% (662) The waveforms of Eq, (66-1) are represented by the dots in Fig. 6641 in a signal space in which the coordinate. axes are the. orthonormal waveforms 41 () = V/2/T) cos aot and_u,(t) = \/(2/T,) sin ot. The distance of each dot from the origins VE, = J.T, From Eq, (661) we have vl = (J/7P, 008 4) 608 9t ~ (FF; sin 4) sin og (663) Defining p, and p, by Pe BP, 008 by (66-4a) = SIP, sin by (66-4)y sin [Famer Fit Geom preston f May PSK gale Eq. (66-3) becomes alt) 08 19 — p, sin wot (66-5) Both p, and p, can change every T,= NT; and can assume any of M possible values. The quantities p,, p,, and , " power es Pa» and dy are random processes. The power spectral densities of p, and p, ate given by Eq” (226-4) as rece = 27,1. 208"G, (UM) 666) aod 2P, 1, si 3, Ge) 66-7) However, since ¢ is uniformly distributed cos? 6 = Sia B= 4 (66.8) so that Gin il) (669) ‘As we have already noted, when signals with spectral density given by Eq. (66-9) are multiplied by a carrier, the resultant spectrum is centered at the carrier fre~ quency and extends nominally over @ bandwidth (66-10) We thus note that as we increase the number of bits N per symbol the bandwidth becomes progressively smaller. On the other hand as we can see from Fig. 661 the distance between symbol signal points becomes smaller. We readily calculate (asing the law of cosines) that this distance is J¥E, a IM) = SINE STP) (66.1) where E, is the symbol energy P, x (NT) = P,T,= NE, and E, = P, Ty is the cenoray associated with one bit. Thus as we increase N, Le, as we increase the duration of the symbol, the bandwidth decreases, the distance d decreases and. 2 We shall see, the probability of error becomes higher. Such is the case for all increases in N except forthe increase from N = 1 (BPSK) to N = 2(QPSK) M-ary Transmitter and Receiver ‘The physical implementation of an M-ary PSK transmission system is moder- ately elaborate. Such hardware is only of incidental concern to us in this text so ‘we shall describe the Mary transmitter-receiver somewhat superficially. ‘As shown in Fig. 66-2, at the transmitter, the bit stream B() is applied to a serial-to-parallel converter, This converter has facility for storing the N bits of a symbol. The WV bits have been presented serially, that is, in time sequence, one after another, These N bits, having been assembled, are then presented all at once fon N output lines of the converter, that is they are presented in parallel. The con- verter output remains unchanging for the duration NT, of a symbol during which time the converter is assembling # new group of NV bits. Each symbol time the converter output is updated, ‘The converter output is applied to a D/A converter. This D/A converter gen erates an output voltage which assumes one of 2” = M different values in a one to-one correspondence to the M possible symbols applied to its input. That is, a n Shuzo 9 Sioet alee 2 a size | omeu o | etter sos i nd Fy a Figure 66:2 Mary PSK tani2 = pcos eat Kars onetoane cronies oie semble Stel The pase Change oe ry “icra town Fig 663 is cima 0 he ont QPSK. reser bonivihs Bop common o Bef fr OFSK, TSK sytem Tre 67 QUADRATURE AMPLITUDE SHIFT KEYING (QASK) In BPSK, QPSK, and M-ary PSK we transmit, in any symbol interval, one Flgre 663 Mary PSK recive. signal or another which are distinguished from one another in phase but are all of the same amplitude. In each of these individual systems the end points of the signal vectors in signal space falls on the circumference of a citcle. Now we have noted that our ability to distinguish one signal vector from another in the pre sence of noise will depend on the distance between the vector end points. It is hence rather apparent that we shall be able to improve the noise immunity of a system by allowing the signal vectors to differ, not only in their phase but also in amplitude. We now describe such an amplitude and phase shift keying system, Like QPSK it involves direct (balanced) modulation of carriers in quadrature (ie, | fading channel). ( | | | |integrators have an integration time equal to the symbol time T; and, of course, as usual, symbol time synchronizers (not shown) are required, Finally, the orig- ‘nal input bits are recovered by using A/D converters. 68 BINARY FREQUENCY-SHIFT KEYING In binary frequency-shift keying (BFSK) the binary data waveform a(?) generates a binary signal urslt) = /2P, €08 Log t+ dA] (68-1) ered) = +1 or —1 corresponding to the logic levels 1 and 0 of the data wave- foom. The transmitted signal sof amplitude /2P, and is either Pyraalt) = 5H(0 = V/2P, 605 (wy + 9} (68-2) or Uarselt) = s4(t) = V/2P, £08 («9 — OMe (68-3) and thus has an angular frequency 0» + oF @y ~Q with © 2 constant offset from the nominal carrier frequency @). We shall call the higher frequency ‘yf = O05 +) and the lower frequency ,( = «2» — Q. We may conceive that the BFSK signal is generated in the manner indicated in Fig. 68-1. Two balanced modulators are used, one with carrier wy and one with carrier «,. The voltage values of p(t) and of pt) are related to the voltage values of a) inthe following, ao | pat | puto siv | +iv [ov wl oov | siv “Thus when a) changes from +1 to —1 py changes from 1 to O and p, from 0 t0 At any time either py of py, is 1 but not both so that the generated signal is either at angular frequency oy oF at >, JP nd | VIF. rt cos ont adder Loe seal Lind eorone VE Tru Figure 641 A representation of mancer in which a BESK signal canbe generated Spectrum of BFSK In terms ofthe variables py and p, the BFSK signal is /2P, py cos (ory + By) + V/3P, py, cos (w,t +04) (68-4) where we have assumed that each of the (wo signals are of independent and random, uniformly distributed phase. Each of the terms in Eq, (68-4) looks like the sighal /2P, Hi) cos «agt which we encountered in BPSK [see Ea, (62-9] and for which we have already deduced the spectrum, but there is a important difference. Inthe BPSK case, is bipolar, ie it altemates between +1 and ~1 while inthe present case py and pate unipolar, alternating between +1 and 0 We may, however, rewrite Py and py asthe sums ofa constant and a bipolar vari able, that is Parse) edt) = 4+ $ pelt) (68-50) pu =$44 P00 (68-54) In Fg. (68-5) py and pi, are bipolar, alternating between +1 and —1 and are complementary. When py is + 1. p= —L and vice versa, We have then 00s (o,t + 6) [rnc tone +0) + |B rcos tout +60) (68.6) ‘The first two terms in Eq, (68-6) produce a power spectral density which consists of two impulses, one at fy and one at fz. The last two terms produce the spec- trum of two binary PSK ‘signals (see Fig. 62-2a) one centered about fy and one about f,. The individual power spectral density patterns of the last two terms in Eq, (68-6) are shown in Fig. 68-2 for the case fy — fi, = 2f,. For this separation between fy and f, we observe that the overlapping between the two parts of the spectra is not large and we may expect to be able, without excessive difficulty, to distinguish the levels of the binary waveform ai). In any event, with this separa- tion the bandwidth of BFSK is BW(BFSK) = 4f, 68.7 hich is twice the bandwidth of BPSK, Receiver for BFSK Signal A BESK signal is typically demodulated by a receiver system as in Fig. 68-3. The signal is applied to two bandpass filters one with center frequency at fy the other at f,. Here we have assumed, as above, that fy ~ fy, = 29/2) = 2. The filter frequency ranges selected do not overlap and each filter has a passband wide ‘enough to encompass a main lobe in the spectrum of Fig. 68-2. Hence one filterPower special density Gal) BW = a Figure 682 The power specenl dense ofthe individual terms in Ea (68-8. fens Bn th Envelope rr py saacer Trine VIP, costo + 00) ao Fie 7 Envelope t+ then re Loh Bole Figee 693 A receives fora BFSK sgn will pass nearly all the energy in the transmission at fy the other will perform similarly for the transmission at f,. The filter outputs are applied to envelope detectors and finally the envelope detector outputs are compared by a compara. tor. A comparator isa circuit that accepts two input signals. It generates @ binary ‘output which is at one level or the other depending on which input is larger ‘Thus at the comparator output the data d() will be reproduced, ‘When noise is present, the output of the comparator may vary due to the systems response to the signal and noise. Thus, practical systems use a bit syn- chronizer and an integrator and sample the comparator output only once at the end of each time interval Geometrical Representation of Orthogonal BFSK We noted, in M-ary phase-hift Keying and in quadratureamplitde shit keying, that any signal could be represented as Cu) + Cy us(). There a) and ae the orthonormal vectors i signal spac, that iwi) = 42/7, cos eet sod 1440 = J/2/, sin gt. The funtionsw, and pare orthonormal over the symbol interval, and the symbol isa singe bit, 1, ~ Ty. The coefcents Cy and Cy are constants. The normalized energies associated with Cyn) and with Cf) are respectively C7 and C} and the total signal energy is C? + C3. In M-ary PSK {and QASK the orthogonality ofthe vectors ay and u results fom thet phase duedrature. Inthe present case of BFSK its appropriate thatthe orthogonality Should result from a special selection of the frequencies of the unit vectors Accordingly, with m and integers, let us establish unit wetors 2 -f : Fonte a in which, as usual, f, = 1/7. The vectors u, and u, are the mth and nth harmo= nics of the (fundamental) frequency fy. As we are aware, from the principles of Fourier analysis, diferent harmonics (m + n) are orthogonal over the interval of the fundamental period T, = If Hf now the frequencies fy and f, in a BFSK system are selected to be (assuming m > n) and of Sa = hy (6.8-10a) t= me (6.8-106) the the coresponding signal vectors are sH0= JE ut 110) l= Ju 10h“0 "(0 Flgure 68-4 Signal space represent- ve sao ion of orthogonal BFSK. The signal space representation of these signals is shown in Fig. 68-4, The signals like the unit vectors are orthogonal. The distance between signal end points is therefore a= JE, (68.12) Note that this distance is considerably smaller than the distance separating end points of BPSK signals, which are antipodal. Geometrical Representation of Non-Orthogonal BFSK When the two FSK signals sp(t) and 5,0) are not orthogonal, the Gram-Schmidt procedure can still be used to represent the signals of Eqs. (68-2) and (68-3). Let us represent the higher frequency signal sy(0) as: sult) = J2P, cos oyt sim) OStST (68-130) and the lower frequency signal s,()as: = SPP, cos ot = sim) +s.) OStST 6813) ‘The representation of these two signals in signal space is shown in Fig. 68-5. Referring to this figure we see that the distance separating sy and s,s (S14 Si2F + Shy rsx in — 2844512 + 532 + Sha (68-14) In order to determine diysx when the two signals are not orthogonal we must evaluate 5,,,52,and 5,5 using Eqs. (68-13). From Eq, (6.8-13a) we have: a a 1. [sot opt [1 $8228] 68.15) Using Eq, (68-135) we first determine s,: by multiplying both sides of the equa- tion by u(t) and integrating from 0 t-< Ty. The result is: VF, [nin cos ged sno Se -20E] ie sil ton * Gn Foy “To arrive at this result we have used Ea (68-130) where : (0 = (JP 608 ay (oats Finally, 23 is found from Eq. (68-136) by squaring both sides of the equation and then integrating from 0 to 7. Since u, and u, are orthogonal, the result is: om terdan [ts 27] oon ‘The distance d between sy and s, given in Eq. (68-14) can now be deter- mined by substituting Eqs. (68-15), (68-164), and (68-176) into Eq, (68-14). The result is poe) S820uT] _ ap, [snow - oT onal! ) - 2 [Saleem sin (oy + oat] (n+ DT 20m 7 sin 20, Ty ane = T, Figere 6855 signal space representation of BFSK when sand 0) are mot oehogoralEquation (68-18) can be simplified by recognizing that: sin 209 2y sin 204 Fy oT | sin (on + oF | < (04 + OT, onal coordinate axes, The signal vectors are then parallel to these axes. The best ‘we can do pictorally isthe three-dimensional case shown in Fig, 610-3. As usual, ‘and as is indicated in the figure, the square of the length ofthe signal vector is the normalized signal energy. Note that, as in Fig. 68-4, the distance between signal points is a= JE, = JiNE, (6.10-3) [Note that this value of dis greater than the values of d calculated for M-ary PSK. with the exception of the cases M = 2.and M = 4. Its also greater than d in the case of 16-QASK.f tem Pectin ue Reoinn vin Fgore 610-3 Geometrical representation of orthogonal M-ary FSK (M = 3) when the faquencin areslected to generate ovhogonal ign 6.11 MINIMUM SHIFT KEYING (MSK) In discussing minimum shift keying, we shall want to make a number of compari- sons between MSK and QPSK. One of these comparisons will concern the spectra of the two systems. For this reason we review briefly some matters con. cerning the spectrum of QPSK. In offset QPSK the transmitted signal is [see Eq, (65-19) Norse) = VB, bt) 08 wot + /P, bie) sin cage To find the power spectral density of this signal we start with the power spectral density of the baseband waveform b,(0). The waveform p(t) = 4/P, bft) is a random sequence of rectangular waveforms, flat topped for symbol duration T.= 7h, and having an amplitude Pls poner speaal desi GU) w given by the general formula (Eq, (2.24-14)) ™ a ar oun = ADE 11-9 where P(f) is the Fourier transform p(t), We readily calculate that the two-sided Power spectral density of (ti, with f, = 1/T, = fy2 and P, = Ey/T, (611-3) “The spectral density of V/P, 640 cos otis generated by translating the two- sided pattern of Eq. (6.11-3) to +f, and also to —f, cach such translated pattern being reduced in magnitude by 4 because the multiplication of \/P,b,(0) by cos gt translates one-half of the voltage spectrum to fo and the other hall t0 yy. (See See 32) Thus the voltage ofeach term is reduced by 2 andthe power by 4. The power spectral density of the second term in Eq. (1-1), that i the term ¥/P, hi) sn gti dential to the density ofthe frst Finaly we note that the two terms are not correlated since 0) and bi) ae quite independent of one another. Hence the total power density i twice the density gonerated by eller {erm Alvogether then we find ‘sin 2n(f — fof ‘sin 2e(f + fol] omit 6 Sera | “rsa } «9 Earlier. upon examining the pattern for the power spectral density (see Fig. 62.2) we took the attitude that the bandwidth of QPSK (QPSK and ‘OQPSK yield the same result is B = fy hecavse such a bandwidth is adequate t0 encompass the main lobe. The main lobe contains 90 percent ofthe signal energy Sill, the not inconsiderable power outside the main Tabe is a source of trouble when QPSK isto be used for multichannel communication on adjacent carirs. If, say, we establish additional channels at carrier frequencies fj = fo +f, then the side lobe associated with the ist channel, having @ peak value at frequency a+ 3, willbe a source of serious iterchannel interference, These sie lobes, 3 is tobe noted in Fig. 62-2, are smaller than the mai lobe by only 14 4B, This dificulty, ie, the wide spectrum of QPSK, is de to the character ofthe baseband signal. This signal consists of abrupt changes, and abrupt changes sive rise to spectral components ot high frequencies. In short, the baseband spectral range is very large and mukilication by a career translates the spectral pattern without changing its form. We miaht try to alleviate the dificult by passing the bascband signa through a low-pass iter to suppress the many sidelobes, Sich fering wll cause intersymbol interference. The problem of interchannelinter- ference in QPSK isso serious that regulatory and standardization agencies such a8 the FCC and CCIR will not permit these systems to be used encept with bandpass fiering at the cartier Frequency (Ge, at the transmitter outer) to suppress the side tes. The filtering which we have just desribed doesnot, in certain situation, nee- essarly resolve the problem of intrchannel interference. We shall discus the ratte qualitatively: We reall hat QPSK (staggered or not) ia system in which the signal is of constant amplitude, the information content being borne by phase changes. In both QPSK and OQPSK there are abrupt phase changes im the signal. In QPSK these changes can occur a the symbol rate, = 1/27, and can be as large as 180°. In OQPSK phase changes of 90" can occ atthe bit rte ‘Now it turns out that when such waveforms with abrupt phase changes, are Al tered to suppress sidcbands, the effet of the filter, atthe times of the abrupt phase changes, isto cause substantial changes inthe amplitude of the waveform, Such amplitude variations can cause problems in QPSK communication systemsal I () I Ayey Nargwband q Fire 615.2 QPR decor takes place; each duobinary decoder being similar to the decoder shown in Fig. 6.14-3h except that they operate at f/2 rather than at f,, The reconstructed data df) and d{1)is then combined to yield the data d(). PROBLEMS {621 The data consis ofthe it ream GO1010O1100, Assume that he bit ae sequal to the trier frequency fan 0 pal) 662.2 Calculate and plo the poer , contained inthe power spec density Gi) in Bq, (628) 8 ‘function of the equency range Oto Nae tht whenf'=0 Py = and when) ininte Py =P, 63.1. The bit sueam dt sto be transite using DPSK. 10 ip ODI0LQD110, determine Show that MME ~ Tp yl the orginal data 4641. The bit seam i i oe eanmited sing DEPSK, 1s 001010011010, determine Show that after decoding sing the dteuit of Fig. 641 the date) resverd. Show tha if the fourth it in Mo isin erzr, then the fourth ad data wl alo ben ron {6541. The bitstream Hs to Be tana sing OQPSK. If Me) 091010011010 sete nt ace Ea, 65.0, Assume, = 12 = f (652. Repeat Prob 651i the od bit steam is delayed by Tso that OPSK i generated, {6S:3, Very thatthe stoi shown in Fig. 655 yields sng and cot tach with the sae phase tthe recived signal 54. Very Eq 65.5) Show thatthe minimum distance betwaeh gal ven by Eq (656) {6st ary PSK is usd 1 mediate the team OOIDIDDIIOIO. Sketch the iano waveoem aah ASE, = 3 ~ Jo (66-2. Repeat Prob. 61 Ibary PSK isu. Assume f= f= fo ‘6. Verity Fa (669) aking neo he et that has he drt vals given hy Fa (64.2 and docs no ake onal ales from = ton 64. Very Eq 66.10, (668. IPA becomes aren Eg (6611), show tat dapprouches zero 74, Vesity Ep. (67-23 (67-9, 67.2. The bit sen 001010011010 is to be waveform. Asm f= 4 =f 67 Very Ba (67), 674 Verity Eq. 67.40, 618, Verify Eq 62.12, 6746, Vesy Ba, (67-19. Calealate the average pomer contained inthe signal vag) atthe fe quem a 468-1. The bitstream 0O101001100 ict be tans using BFSK. Sketch the Warsmited wave form, Assume fe = S898 Fy ~ 2 2. One way to obiin the power spectral density of a waveocm i to Gist obtain the aloe ion function” fs) The power speces! density is the Fourer trasiowm of RU Since eo) = Flt + (0) Find Ro) for he BFSK signa of. (68-6) and very Fig. (68-2, (1 = 0 find Ro a0 the power spectra deny. 68.3. Verify Eq 8-121 64, Very Eas. (68-15, 6S. Verify Eqs (68-16) and (65.170) 446 Ves Ea, (6819). What isthe relationship between ff and fn oder that Ea, (10) re to Fe (6812, (610-1. Consider sing 419 PSK (a) Show thatthe frequencies ate separated by hey ae each octhogont (bh) Caleuate the bandwidth B under the canton ao) Show tht th bandh ee ight ofthe valve requires By Eq (610-1 Explain the difference, () Determine the rat of bandwidth of ary FSK and ‘ary FSK when the fequencies ate separated by f- Repeat or 6114. Verity Ea (11-60) {6412 Very E6118 ong Eq, (61-12, {61128 1100101001100 teh Fy) Asse in a, (611-12) thet m= 5. SATA Repeat Prob 611i = 3 6118. Verity Ea (611-18) 1146, Verify that in Fig. 611-5 the waveforms sf) and) are a8 shown and that the output hh GIDL. Verily Table 624 (6122. Verity Eos. (6.12.7) and (1248), ransmited sing IGQASK. Sketch the transit123, very Fag (1240 and 6124) 6124 Plt Guan GHP 4 anton eqn sig Ea. (1240 and (121 ‘te, Vert Ba. S1, Show tha if mo dbinry as y= 8k In ig corer sae ch tne uate ha d= JW coe ag + SFR n gt then th eer seme oct and snot cam be covered in the reiner by ering the Soa yn S12 Rope 6151 and ae ech ered by» “hick wl owpse roto SShnametine motel and op, = ABA 1) ate ede CHAPTER SEVEN MATHEMATICAL REPRESENTATION OF NOISE Al the communication systems discussed in the preceding chapters accomplish the same end. They allow us to reproduce the signal, impressed on the communi- cation channel atthe transmitter, at the demodulator output. Our only basis for comparison between systems, up to this point, has been bandwidth occupancy, convenience of multiplexing, and ease of implementation of the physical hard. ware, We have neglected, however, in our preceding discussions, the very impor- tant and fondamental fact that, in any real physical system, when the signal voltage arrives atthe demodulator, it will be accompanied by a voltage waveform which varies with time in an entirely unpredictable manner, This unpredictable voltage waveform is a random process called noise. A signal accompanied by such a waveform is described as being contaminated or corrupted by noise, We ‘now find that we have a new basis for system comparison, that is, the extent to Which @ communication system is able to distinguish the signal from the noise and thereby yield a low-distortion and low-error reproduction of the original signal. In the present chapter we shall make only a brief refefence to the sources of noise which are discussed more extensively in Chap. 14. Here we shall be con- cerned principally with a discussion of the mathematical representation and sta tistical characterizations of noise. 7.1 SOME SOURCES OF NOISE ‘One source of noise is the constant agitation which prevails throughout the uni verse at the molecular level. Thus, a piece of solid metal may appear to our gross view to be completely at rest. We know, however, that the individual molecules atsGnarter EIGHT NOISE IN AMPLITUDE-MODULATION SYSTEMS In Chap. 3 we described a number of different amplitude-modulation commur cation systems. In the present chapter we shall compare the performance of these systems under the circumstances thatthe received signal is corrupted by noise. 8.1 AMPLITUDE-MODULATION RECEIVER [A system for processing an amplitude-modulated carrier and recovering the base- bband modulating system is shown in Fig. 8-1. We assume that the signal has suffered great attenuation during the course of its transmission over the commu- nication channel and hence is in need of amplification. The input to the system might be a signal furnished by a recsiving antenna which receives its signal from a (ransinilting antenna, The carrier of the received signal is called radion jrequency (RF) carter, and its frequency is the radio frequency f,. The input signal is amplified in an RF amplifier and then passed on to a mixer. In the mixer the modulated RF carrier is mixed (Le, multiplied) with a sinusoidal waveform generated by a local oscillator which operates at frequency fue. The process of ‘mixing is also called heterodyning, and since, as is to be explained, the hetero- dyning local-oscillator frequency fin is selected to be above the radio frequency ste system is often referred to asa superheterodyne system. ‘The process of mixing generates sum and difference frequencies. Thus the mixer oulput consists of a carrier of frequency fine +f ANd a CAPTICT fg ~ fa Each carrier is modulated by the baseband signal to the same extent as was the input RF carrer. The sum frequency is rejected by a filter. This fier is not shown NOISE IY AMPLITUDE-MODULATION SySTEMS 347 Loca estes RE = mere | wnt | sae |_.f Poem setter ree Figere 81-1 A receiving tem for an amplide-modulate signal explicitly in Fig. 81-1 and may be considered to be part of the mixer. The difference-frequency carrier is called the intermediate frequency (IF) cartier, that 18, fe = Se —fr- The modulated IF carrier is applied to an IF amplifier. The process just described, in which a modulated RF carrier is replaced by a modu- lated IF cartier, is called conversion. The combination of the mixer and local oscillator is called a converter The IF amplifier output is passed, through an IF carrier filter, to the demodulator in which the baseband signal is recovered, and finally through a baseband filter. The bascband filter may include an amplifier, not explicitly indi- cated in Fig. 81-1. IP synchronous demodulation is ued, a synchronous signal source will be required, ‘The only absolutely essential operation performed by the receiver is the process of frequency translation back to baseband. This process is, of course, the inverse of the operation of modulation in which the baseband signa is feequency- translated to a cartier frequency. The process of frequency translation is pet formed in the system of Fig. 8.-1 in part by the converter and in part by the demodulator. For this reason the converter is sometimes referred to as the first detector, while the demodulator is then called the second detector. The only other ‘components of the system are the linear amplifiers and filters, none of which would be essential if the signal were strong enough and there were no need for multiplexing,348 PRINCIPLES OF COMMUNICATION SYSTEMS It is apparent that there is no essential need for an initial conversion before demodulation. The modulated RF carrier may be applied directly to the demodu- lator. However, the superheterodyne principle, which is rather universally incor Porated into receivers, has great merit as is discussed in the next section. 82 ADVANTAGE OF THE SUPERHETERODYNE PRINCIPLE: SINGLE CHANNEL, A signal furnished by an antenna to a receiver may have a power as low as some tens of picowatts, while the required output signal may be of the order of tens of ‘atts. Thus the magnitude of the required gain is very large. In addition, 10 mini mize the noise power presented to the demodulator, ters are used which are no wider than is necessary to accommodate the baseband signal. Such filers should be rather lat-topped and have sharp skirts tis more convenient to provide gain and sharp flat-topped filters at low frequencies than at high. BY way of example, in commercial FM broadcasting, the RF cartier frequency is in the tange of 100 MH, while at the FM receiver the IF frequency is 10.7 MHz. Thus, in Fig. 8.1-1 the largest part, by far, ofthe required gain is provided by the IF amplifier, and the critical filtering done by the IF filter. While Fig. 8.141 suggests a separate amplifier and filter, actually in physical receivers these two Usually form an integral unit, For example, the IF amplifier may consist of a umber of amplifier stages, each one contributing to the filtering. Some filtering will also be incorporated in the RF amplifier. But this fitering is not eritcal. Tt serves principally to limit the total noise power input to the mixer and thereby voids overloading the mixer with a noise waveform of excessive amplitude RF amplification is employed whenever the incoming signal is very small, This is because of the fact that RF amplifiers, such as masers, are low-noise devices: je, an RF amplifier can be designed to provide relatively high gain while generating relatively little noise. When RF amplification is not employed, the signal is applied directly to the mixer. The mixer provides relatively litte gain and generates a relatively large noise power. Calculations showing typical values of gain and noise power generation in RF, mixer, and IF amplifiers are presented in Sec. 14.14, Multiplexing ‘An even greater merit of the superheterodyne principle becomes apparent when we consider that we shall want to tune the receiver to one ot another of a ‘number of diffrent signals, each using a different RF carrier. If we were not to take advantage of the supetheterodyne principle, we would require a receiver in which many stages of RF amplification were employed, each stage requiring tuning. Such tuned-radio-frequency (TRF) receivers were, as a matter of fact, commonly employed during the early days of radio communication. It is difficult ‘enough fo operate at the higher radio frequencies; itis even more difficult to ‘OIE IN AMPLITUDE-MODULATION systiMs 39) ang-tune the individual stages over a wide band, maintaining atthe same time a reasonably sharp flat-topped filter characteristic of constant bandwidth In a superhet receiver, however, we need but change the frequency of the local oxcillator 10 g0 from one RF carrier Frequency to another. Whenever fis se $0 that fog —fy =f the miner will convert the input modulated RF carriet to a ‘modulated carrier at the IF frequency, and the signal will proceed through the demodulator to the output. OF cours, itis necessary to gang the tuning of the RF ampli to the frequency control of the local oscillator. But again this anging is not critical, since only one or two RF amplifers and fiters are employed. Finally, we may note the reason for selecting fa. higher than fi. With this higher selection the fractional change in fa, required to accommodate a. given range of RF frequencies is smaller than would be the ease for the alternative selection 83 SINGLE-SIDEBAND SUPPRESSED CARRIER (SSB-SC) ‘The receiver of Fig. 81-1 is suitable for the reception and demodulation of all types of amplitude-modulated signals, single sideband or double sideband, with and without carrier. The only essential changes required to accommodate one type of signal or another are in the demodulator and in the bandwidth of the IF carrier filter. Hence, from this point, our interest will focus on the section of recei= ver beginning with the IF filter and through to the output, The signal input to the IF fiter is an amplitude-modulated IF carrier, The normalized power (power dissipated in a f-ohm resistor) of this signal is $,. The signal arrives with noise. Added, is the noise generated in the RF amplifier and amplified in the RF amplifier and LF amplifier. The IF amplifiers and mixer are also sources of noise, ie. thermal noise, shot noise, etc, but this noise, lacking the gain of the RF amplifier, represents a second-order effect. (See Sec. 14.11.) We shall assume that the noise is gaussian, white, and of two-sided power spectral density 9/2. The IF filter is assumed rectangular and of bandwidth no wider than is necessary to accommodate the signal. The output baseband signal has a power S, and is accompanied by noite of total power Ny, Calculation of Signal Power With a single-sideband suppressed-carrer signal, the demodulator is a multiplier as shown in Fig. 83-1a. The cartier is A cos 2. For synchronous demodu- lation the demodulator must be furnished with a synchronous locally generated carrier cos 2nf.t, We assume that the upper sideband is being used: hence the carrier filter has a bandpass, as shown in Fig. 83-16, that extends from f. tof, + fu, Where fy is the baseband bandwidth, The bandwidth of the baseband filter extends from zero to fy as shown in Fig. 83-1c Let us assume that the baseband signal is a sinusoid of angular frequency‘coir asebang iar Hef tie gt) Nose sper power ‘eet = Oe ap) OE a eT Figure 83:1 (a) A synchronous demodulator operating on a siglesieband sngle-tone signal () ‘The bandpass ang ofthe crete (e The pasebané of the lowpass basband te, Sol S Jue. The cartier frequency is f, and, since we have assumed that the upper sideband is being used, the received signal is sft) = A c0s [2m f + dé) 31) ‘The output of the multiplier is 50 = 50 608 0,1 = 400s (2H, +f) + 40s aft (32) Only the difference-trequency term will pass through the baseband filter. There- fore the output signal is sa 4 $008 2afat (833) “hich ie the modatng signal amplified by 4. “The inp signal powers set 634) while the output signal power is 83-5) Thus 036) [NOISE IN AMPLITUDE-MODULATION SYSTEMS. 361 ‘We may readily see that, even though Eq, (83-6) was deduced on the assumption of a sinusoidal baseband signal, the result is entirely general, For suppose that the baseband signal were quite arbitrary in waveshape. Then the single-sideband nal generated by this baseband signal may be resolved into a series of harmo- nically related spectral components. The input power is the sum of the powers in these individual components, Next, we note, as was discussed in Chap. 3, that superposition applies to the multiplication process being used for demodulation. ‘Therefore, the output signal power generated by the simultaneous application at the input of many spectral components is simply the sum of the output powers that would result from each spectral component individually. Hence $, and S, in Eqs. (6.3-4) and (83-5) are properly the total powers, independently of whether a single or many spectral components are involved, Calculation of Noise Power We now calculate the output noise N,. For this purpose, we recall from Sec. 78 that when a noise spectral component at a frequency fis multiplied by cos 2rf,t, the original noise component is replaced by two components, one at frequency J. +f and one at frequency f; ~f, each new component having one-fourth the power of the original. The input noise is white and of spectral density n/2. The noise input to the multiplier has a spectral density G,, as shown in Fig. 83-2a, The density of the noise after multiplication by cos 23,1 is Gyz as is shown in Fig. 8.326, Finally the noise transmitted by the bascband filter is of density Gay a8 in Fig. 8.3-2c. The total noise output is the area under the plot in Fig. 83-2¢, We have, then, that ne Nem Mga 63-7 oes 7 Figere 82:2 Spectral densies of moses in SSB demodulator. (a) Dentiy ., of mote input to multi ie. () Density, of ose output of malkper. Density Gof oie oupet of hesetond fierLe 62 PRINCIPLES OF COMMUNICATION SYSTEMS Use of Quadrature Noise Components tis of interest to calculate the output noise power Ny, in an alternative manner using the transformation of Eq, (7.11-2); n{) = nt) c08 2a — ge) sin Daf (63-8) We now apply Eq, (83-8) 0 the noise output of the IF filter so that nt) has the spectral density G,, as in Fig, 83-2a, The spectral densities of n(@) and mt) are {see Eqs. (7.12-7) and (712.8) GD = We observe that for << fus Gull. +0) Gq f) and Gal f) areas shown in Fig. 83:3 “Multiplying me) by eos 2, «yields rt) 608 2rfet = n(f) 008? Inf, — mit) sin Inf t 008 Inf t Ant) + U( c0s 4a — dno) sin daft (83-10) ‘The spectra of the second and third terms in Ea, (83-10) extend over the range 2, ~fu to 2. +Jy and are outside the baseband ‘filter, The output noise is, therefore, N= Gale) Gul +) 83.9) n/2, while Gs, ~f) = 0, so that nf = n0 63.1) ‘The spectral density of nis then Gy = 4Gy, = Hi/2) = n/8. Hence as before, as, shown in Fig 83-2c, the spectral density G,, is n/8 over the range — fy tO fu, and the total noise is again N, = nfy/4 Calculation of Signal-to-Noise Ratio (SNR) Finally we may calculate, using Eqs. (83-6) and (8.3-7), the signal-to-noise ratio at the output, SYN, We have S_ $4 N, hls the ‘The importance of S/N, is that i€ serves as a figure of merit ofthe performance of communication system. Certainly, as S/N, increases, it becomes easier to dis- tinguish and to reproduce the modulating signal without error or confusion. If a (83.12) 7 Figwe833 Power spectral denies of G, snd 6, NOISE I AMPLITUDE-MODULATION SYSTEMS 353 system of communication allows the use of more than a single type of demodu- lator (say, synchronous or nonsynchronous), that ratio S/N, wil serve as a figure cof merit with which to compare demodulators. We observe from Eq. (83-12) that (© increase the output signal-to-noise power ratio, we can increase the transmitted signal power, restrict the baseband frequency range, or make the receiver quieter, 84 DOUBLE-SIDEBAND SUPPRESSED CARRIER (DSB-SC) When a baseband signal of frequency range Jy is transmitted over a DSB-SC system, the bandwidth of the carrier filter must be fy rather than fy. Thus, input noise in the frequency range f, ~ fy to f+ fy Will contribute to the output noise, rather than only in the range J; t0 f+ Ju as in the SSB case. Calculation of Noise Power ‘This situation is illustrated in Fig. 84-10, which shows the spectral density Gui) of the white input noise after the IF filter. This noise is multiplied by cos a, ‘The multiplication results in a frequency shift by +f, and a reduction of power in the power spectral density of the noise by a factor of 4, Thus, the noise in region of Fig. 84-10 shifts to regions d shown in Fig. 84-1b, Similarly regions a, band, c of Fig. £4-1a are translated by -bf, and are also attenuated by 4 as shown in « o nth, 2,44, Any Bethy Figure 841 Special densities of noise in DSB demodulation.) Density Gy of noise filer (8) Densty 6, of oe output of baseband te utp of F1S PRINCIPLES OF COMMUNICATION SYST Fig. 84.16, Note that the noise-power spectral density in the region between fy and + fy is 7/4, while the noise density in the SSB case, as shown in Fig. 83-2c is only 1/8. Hence the output noise power in twice as large as the ‘output noise power for SSB given in Eq, (83-5). The output noise for DSB after ‘baseband filtering is therefore Tap) = te 641) Calculation of Signal Power ‘We might imagine that for equal received powers, the ratio S/N, for DSB would be only half the corresponding ratio for SSB, We shall now see that such is not ‘the case, and that the ratio SJ/N, isthe same in the two cases. Again, without loss in generality, let us assume a sinusoidal baseband signal of frequency f_ < fy. T0 keep the received power the same as in the SSB case, that is, $; = 42/2, we write (0) = /7 A cos Daft cos 2nf.t 0s (2a(fe + fg0-+ 4 cos at A942) ae eae ay-4 as as in Eq. (8.3-4), In the demodulator (multiplier) s(t) is multiplied by cos w,¢. The upper- sideband term in Eq, (84-2) yields a signal within the passband of the baseband filter given by Vi ‘The received power is then ss) = om hat 44) The lower sideband term of Eq, (84-2) yields 4 (=A cos nt (45) sate) ws fr ) ‘Observe, most particularly in Eqs. (84-4) and (84-5), that s(?) and s{(9) are in phase and that hence the output signal is sft) = 510+ 0) A cos 2afat 649 Vi which has a power een NOISE IN AMPLITUDE-MODULATION SYSTEMS. 385 rather than S, = 42/8 = Sy/4 as in Eq, (83-5) for the SSB case. Thus we see that ‘when a received signal of fixed power is split into two sideband components each ‘of half power, as in DSB, rather than being left in a single sideband, the output signal power increases by a factor of 2. This increase results from the fact that the contributions from each sideband yield output signals which are in phase. A doubling in amplitude causes a fourfold increase in power. This fourfold increase, due to the inphase addition of s, and s,, isin part undone by the need to spit the input power into two half-power sidebands. Thus the overall improvement in output signal power is by a factor of 2 On the other hand, the noise outputs due to noise spectral components sym- ‘metrically located with respect to the carrier are uncorrelated with one another ‘The two resultant noise spectral components in the output, although of the same frequency, are uncorrelated. Hence the combination of the two yields a power which is the sum of the two powers individually, not larger than the stm, as is the cease with the signal. Calculation of Signal-to-Noise Ratio Returning now to the calculation of signal-to-noise ratio for DSB-SC, we find from Eqs. (84-1) and (84-7) that (848) exactly as for SSB-SC. Arbitrary Modulating Signal In the discussion in the present section concerning DSB we have assumed that the baseband signal waveform is sinusoidal. As pointed out in Sec. 8.3, this assumption causes no loss of generality because of the linearity of the demodu- lation. Nonetheless it is often convenient to have an expression for the power of @ DSB signal in terms of the arbitrary waveform mi) of the baseband modulating signal. Hence let the received signal be 5h) (t) cos 2nf,.t (84-9) The power of st) is 5, = SH) = m7 eos? Daft = Ami) + SP) cos | (8.4.10) Now m() can always be represented as «sum of sinusoidal spectral components. [OF inerest, albeit of no special relevance inthe present dieussion, is the fact that if m(t) is bandlimited to fy, m7(¢) is bandlimited to 2fy. See Prob. 8.4-1.] Hence mt) cos 4x: consists of «sum of sinusoidal waveorns inthe frequency range 2 Jy. The average valve of such a sum is zero, and we therefore have S850 = dm) 641)QQ es 2356 PRINCIPLES oF comMUNtCATION SYSTEMS When the signal s(i) in Eq. (84-9) is demodulated by multiplication by ‘cos 2af,t, and the product passed through the baseband filler, the output is (0) = m2. The output signal power is (e412) (8413) that is, the same result as given in Eq, (84-7) for an assumed single sinusoidal ‘modulating signal. Use of Quadrature Noise Components to Calealate N, {tis again interesting to calculate N, using the transformation of Eq, (7.11-2): n{t) = n,(0 608 2af,t ~ mgt) sin 2nfet e414) ‘The power spectral density of n,0) and nj) are [see Eqs. (7.12-7) and (7.12-8)) GN = Gal) = GAS. +1) Gul DY (84-15) In the frequency range |f 1 < fu, Gai(fe +I) = Galle ~S) = n/2. Thus GIN=GN=n Misha 416) (This result was also derived in Example 7.12-1) “The result of multiplying nfo) by cos 2 ¢ yields nt) cos 2af,t = 4n40) + 4nd) cos 4nf.t — Ind sin dnt (84-17) Baseband filtering eliminates the second and third terms, leaving nde) = 3a) (8418) ‘The power speciral density of ns then GAN = 26 = thu sS hu 6419) ‘The output noise power Nis, therefore, ‘This result is, of course identical with Eq. (84-1), which was obta sidering directly the effect on a noise spectral component of a multiplication by 608 2 NOISE IN AMPLITUDE-MODULATION SvSTEMS 387 85 DOUBLE SIDI AND WITH CARRIER Let us now consider the case where a carrier accompanies the double-sideband signal. Demodulation is achieved synchronously as in SSB-SC and DSB-SC. The carrier is used as a transmitted reference to obtain the reference signal cos «,¢ (see Prob. 8.5-1), We note that the carrier increases the total input-signal power ‘but makes no contribution to the output-signal power. Equation (8-4-8) applies Girectly to this case, provided only that we replace S, by S™, where SE is the power in the sidebands alone, Then (e5-1) Suppose that the received signal is Sk) = ALT + mio)] cos 2af¢ cos 2nf,t + Amie) 008 Daf 5.2) where m(t) is the baseband signal which amplitude-modulates the carrier A 00s 2rft. The cartier power is A2/2. The sidebands are contained in the terma ‘Amt) cos 2rf,t, The power associated with this term is (4*/2)m(t) where m() is the time average of the square of the modulating waveform. We then have that the total input power 5} s given by £ O+m@] es) Eliminating 4?, we have as4 or, with Eq. 5-0), R55) In terms of the eartier power P, = 42/2, we get, from Eqs. (85-3) and (8.55 that Soo P = mi) 85-6) One 656) If the modulation is sinusoidal, with mf) = m cos 2nf,t(m @ constant). then s(t) = A(l + m 608 Daft) COS 2nf.t 57 In this case m™(@ = m?/2 and 658)ASB PRINCIPLES OF COMMUNICATION SYSTEMS ‘When the carrier is transmitted only to synchronize the local demodulator wave- form cos 2rf.1, relatively little carrier power need be transmitted, In this case m> 1 m/(2-4 m2) = 1, and the signal-to-noise ratio is not greatly reduced by the presence of the carrier. On the other hand, when envelope demodulation is used (Sec. 34), itis required that m < 1, When m = 1, the carrier is 100 percent ‘modulated. In this case m#/(2.+ m) = 4, so that of the power transmitted, only one-third isin the sidebands which contribute to signal power output A Figure of Merit We observe that in each demodulation system considered so far, the ratio Synfy appeared in the expression for output SNR [sce Eqs. (8.3-12), (84-8), and (8.5-5)] This rato is the output signal power S; divided by the product nfy- To give the Product nfy some physical significance, we consider it to be the noise power Nw at the inpul, measured in a frequency band equal to the baseband frequency. Thus (85.9) The tatio Sy/nfy is, therefore, often referred to as the input signal-to-noise ratio SyNy_ It needs to be kept in mind that Ny is the noise power transmitted through the IF fiter only when the IF filter bandwidth isfy. Thus Ny isthe true input noise power only in the case of single sideband. For the purpose of comparing systems, we introduce the figure of merit 7, defined by 13 3Ne 5:10 ‘The results given above may now be summarized as follows 1 SsB-sC san 1 Dsp-sc (65:12) m0) 5 - psp 5-19 T+ mo) : DSB with sinusoidal modulation (8-14) A point of interest in connection with double-sideband synchronous demodu- lation is that, for the purpose of suppressing output-noise power, the carrer filter of Fig. 83-1 is not necessary. A noise spectral component at the input which lies outside the range f+ fy will, ater multiplication in the demodulator, lie outside the passband of the baseband filter. On the other hand, if the carrier filter is climinated, the magnitude of the noise signal which reaches the modulator may bbe large enough to overload the active devices used in the demodulator. Hence, such carrier filters are normally included, but the purpose is overload suppression rather than noise suppression. In single sideband, of course, the situation is difler- ent, and the carrer filter does indeed suppress noise. 86 SQUARE-LAW DEMODULATOR We saw in Sec. 36 that a double-sideband signal with carrier may be demodu- lated by passing the signal through a network whose input-output characteristic is not linear, Such nonlinear demodulation has the advantage, over the linear synchronous demodulation methods, that a synchronous local carrier need not bbe obtained. This eliminates the rather costly synchronizing circuits, In this section we discuss the performance and determine the output SNR of a nonlinear demodulator which uses a network whose output signal y (voltage or current) is related to the input signal x (voltage or current) by ae (864) in which 4 is a constant, As shown in Fig. £6-1, this nonlinear network, which constitutes the demodulator, is preceded by a bandpass IF filter of bandwidth 2fy and is followed by a baseband low-pass filter of bandwidth fy We discuss this square-law demodulator in patt for its own intrinsic interest, but also because it exhibits an important characteristic which is not displayed by the linear synchronous demodulators. We have previously adopted the quantity 7 =(SJN,\(SYNu) as @ figure of merit for the performance of demodulator in the presence of noise. We observed [Eqs. (85-11) to (8.5-14)] that this figure of merit is not a function of the input signal-to-noise ratio S/N. Therefore, if the input S/N decreases, say, by a factor a, the output S,/N, will also decrease by a ‘The nonlinear demodulator also has a range where the figure of merit y is inde yemintne met [a] _[rncmee Bema] tetan vet ant wace Ban eth anh ; i et batt Sri! whe hth hte bt Fier saps oe Figere 6.1 The square law AM demodulator[364 princinss oF COMMUNICATION SYSTEMS Equation (86-22) is plotted (solid plot) in Fig. 64 with the variables ‘expressed in terms of their decibel (4B) equivalents, ie, the abscissa is marked off in units of 10 log (P,/Ny}. Above threshold, when P./Ny is very large, Eq, (8.6- 22) becomes 0 86.23) Below threshold, when P,/Ny < 1, Eq, (8.6-22) becomes. wolf) 8.457 (2) (8029 For compro, Eq (8623 and (8.62) tae ao tech tel Fle A We Sore pte ofa hohe tae Rls eae Sgt a cmapening vo Pe fe re Te ane Son Asymptote for me Eeigss23) Sy sommate or B-no eteenany Ay the phenomenon of threshold NOME IN AMPLITUDE-MODULATION svSTEMS 365 chosen arbitrarily to be the point at which the performance curve falls away by 1dB as shown, On this basis it turns out that the threshold occurs when P./Ny = 46 dB or when P, = 29Ny, 8.7 THE ENVELOPE DEMODULATOR We again consider an AM signal with modulation |m)| < 1, To demodulate this DSB signal we shall use a network which accepts the modulated carrier and pro. vides an output which follows the waveform of the envelope of the carrier. The diode demodulator of Sec. 36 is a physical circuit which performs the required ‘operation to a good approximation. As usual, as in Fig. 8.31, the demodulator is preceded by a bandpass fiter with center frequency , and bandwidth 2fy, and is followed by a low-pass baseband filter of bandwidth fy. It is convenient in the present discussion to use the noise representation aiven in Eq, (711-2) (0) = nt) 608 1, ~ ng) sin og @7-1) I the noise mt) has a power spectral density n/2 in the range If — fol < fr and is zero elsewhere as shown in Fig. 84-1, then, as explained in Sec. 7.12, both 1.0) and nj) have the spectral density 9 inthe frequency range — fy t0 fy. ‘At the demodulator input, the input signal plus noise is si + (0) [1 + mo} c05 0, + nt) cos ot — nf) sin ot (8.7-2a) ALI + mie)} + nQ@)} 608 at ~ ng) sin (87-26) where A is the carrier amplitude and mit) the modulation, In a phasor diagram, the first term of Eq. (87-26) would be represented by a phasor of amplitude ALI ++ mio] + m0), while the second! term would be represented by a phasor per- pendicular to the first and of amplitude no). The phasor sum of the two terms is then represented by a phasor of amplitude equal to the square root of the sum of the squares of the amplitudes of the two terms. Thus, the output signal plus noise ust prior to baseband filtering is the envelope (phasor stm): 520 + Al) = (CALL + ma) + nO + MBE (87.30) APLL + ml]? + DALE + mice) + nto + ng}? (87-38) We should now like to make the simplification in Eq, (8.7-3b) that would be allowed if we might assume that both |n,()| and |njt)| were much smaller than the carrier amplitude 4, The difficulty is that n, and n, are noise “waveforms” for Which an explicit time function may not be written and which are described only in terms of the statistical distributions of their instantaneous amplitudes. No ‘matter how large A and how small the values of the standard deviation of n{0) or ‘no, there is always a finite probability that |n.(0)|, [n4é)|, oF both, will be com-$466 PRINCIPLES OF COMMUNICATION SYSTEMS parable oor even are has An the oer band, fhe anda deviations rin) and nf) are ech Sale than A the hed tat aa proach or eced 4 rvather mel For erp seman, Rie eon tortion, the probably tat) pete han twee the andar) dere ftom sony 04 ang ony OOUGS th eset nes a soar Seer tion, Hence 11 /adT} <4 and we asume tat (91, the aseunption wal a Assuming then that |n,(t)| € A and |nJ¢)| < A, the “noise-noise” terms n3(0) and ni) mabe dropped esting is with he nopesoaton Salt) + malt) = {A2LL + mith]? + ALL + mlehno}? (8.7-4a) 2ne(t) ® cm fe ES ana Using now the further approximation that (1 + x)" = 1 + x/2 for small x we hhave finally that SHO + m0 = ALL + mk + 2) ers, The output-signal power measured afer the baseband fier, and neglecting de terms, is S, = Amp. Since the spectral density of n,(0) = n, the output-noise power after baseband filtering is N, = 2nfy. Again using the symbol Nu = nfy) to stand for the noise power at the input in the baseband range fy. and using Eq, (8.5-3), we find that SUN, _ vs 8.7-6 SYNu PO : The result is the same as given in Eq, (8.5-13) for synchronous demodulation, To make # comparison with the square-law demodulator, we assume m7() 1. In this ease, as before, S, = P,, and Eq, (8.7-6) reduces to Eq, (8.5-6). Hence we have the important result that above threshold the synchronous demodulator, the square-law demodulator, and the envelope demodulator all perform equally well, Provided m'(t) <1 ‘Theeshold Like the square-law demodulator, the envelope demodulator exhibits a threshold. AAs the input signal-to-noise ratio decreases, a point is reached where the signal- to-noise ratio at the output decreases more rapidly than at the input. The caleu- lation of signal-to-noise ratio is quite complex, and we shall therefore be content to simply state the result! that for S/Ny < ,and m=) <1 Se _ mle) a nor Ga) an Equation (8.77) obviously indicates « poorer performance than indicated by 4. (87-6, which applies above threshold [NOISE Wy AMPLITDE- MODULATION svsTeMS 367 Since both square-law demodulation and envelope demodulation exhibit 3 threshold, a comparison is of interest. We had assumed in square-law demodu: lation that m(Q) <1. Then, as noted above, 5, = 4*/2 = P, the cartier power tn eg (22) mo Hi (Pe)? a GR) a Which isto be compared with Eq. (86-24) giving SJN, below threshold for the square law demodulator ‘The comparison indicates that, below threshold, the square-aw demodulator performs better than the envelope detector, Actually this advantage of th square-law demodulator is of dubious value. Generally, when a demodulator i ‘operated below threshold to any appreciable extent, the performance may be s poor as to be nearly usless. What is of greater significance is that the compa Son suggests that the threshold in squarelaw demodulation is Tower than the threshold in envelope demodulation. Therefore a square-law demodulator wil ‘operate above threshold on a weaker signal than will an envelope demodulator. In summary, on strong signals all demodulators work equally well excep that the square-law demodulator requires that m’() 1 to avoid baseband-signa distortion. On weak signals, synchronous demodulation does best since exhibits no threshold, When synchronous demodulation is not feasible, square law demodulation does better than envelope demodulation. It is also interesting (0 note that voice signals requite a 40-dB output signal-to-noise ratio for high duality. In this case both the inear-envelope detector and the square law detector operate above threshold s, N, REFERENCE, 1. Denon, W, and W. Root: "Random Signals and Nowe.” McGraw-Hill Book Company, New York, 1958 PROBLEMS ‘BLA () A superbeterodyne reciver sing an IF froquency of 455 KHz is tuned to 60 KH. I i found that the reeiver picks up a tanamisson ffom a tanumiter whote carrier frequency i 1560 KH Suggest areas for this undesired reception and sogget remedy (het fequenies (680 kt and 1560 kta are rere toa image requencles Why") ‘A341, Lat gi) be a waveform characeriad by a power spectral density Gif). Assume GLP) = 0 fo 712. Show thatthe imeraverage vale of) ces By. 200 8.» 832. As noted in See. 3.10, if) isan abirary baseband waveform, a recived SSB inal may be rien 40) = mt eae 2a t+) Df Here A) i rived fromm) By ahiing By 90° the Phase of every secral component in ma —_ (a) Show hat mands) have the same power spel dense and that mG) = AYO, (6) Show tha ifm) has a spectrum which extends fom zero frequeney lo 8 maximum fre quency fs then m0 (and ma) ll Bane spesen which extend rom aro Fequeny 10 fy368 PRINCIPLES OF COMMUNICATION SYSTIIS (Show that the normaizd power 5, of st) i WD) Prot 431) (2) Caeuate the normalized power S, ofthe demodulated SSB signal, ie, the signal mult ied by cos 2, and then passed tough sbascnd liter Show that §,~ aie and Bens tha 5,5, = 4 [Nove This probem estabishes more generally the res given in Eq. (6) winch was ‘rived om the ass ofthe aumption that the modulating aveorm ta sino) AB. Prove that Eg (811)5 correct by sketching the power spect densi of Ea, (83-10) 34. A haseband canal mis eons wing SSB asin Prob, 83-2. Asse tha the power epee tea densi of mis AAG, (Hint: Use the resuts of oad? 0 late J Nhe Find: {o) The input signal power. (0) The output sana power. (6) If white gaussian foie with power spctal density 2 is aed to the SSB ig, id the coutpat SNR. The baseband iter cule OF tf Ju ‘AUK. A received SSB signal has «power spectrum which exendi ove the requency rang rom = Mtr 10 +f = 1 MHz. The siptal is accompanied by noise with unilorm power special density 10° watt {a The noise is exprested as 8) =m) os 2a t~ nsf Find he power special ensies ofthe quadrature components mand mf of the noe in the petal rane specie ele tS 1% The Signal plus its accompanying nie it motpied by a local cater cos 2. Plt the owe petal density ofthe noite the ouput of the mul lit 'c) The signa plus sos, aller maltipiation, ie peed through a baseband iter and an ami ‘er which promdes x voltage gain of 10. Plot the power spectal density of he nose at the ampli output and ella the oa noe output pomes. AL Show that mi) in Eq (810) bandied t0 fy RAZ Repeat Prob 83-44 DSB rater than SSB modulation i employed RAR A carer of amplitude 10:mV at fi 50 percent amplinide modulated by a sinsoidal wave form of frequency 720 He Its accompanied by thermal nie of two-sided power spect deity a f= 1000 =f, =, 41000 ° 70h 1, F741000 Fe 1-00 "4500 Figore P43 _sreeememenrmem tien ee eR er [OIE AMPLITUDE: MODULATION SYSTEMS 369 y2.= 10"? waty. The sal plot nose ie passed through the er shown. Te signa is deme Ind by meliptiation wi’ local ater ofampltue 1 vo. (o) Find the ovtpet signal power. (6) ind the outpt nase poet ‘RS4. The signal (e+ mil eos 1 syachronouay detected. The reference Sigal coro wd Yo ‘molly the icoring ego obtained by pang the spt signal throughs narombamd Be o Sandwidth 8, 8 shown in Fig PSS (a) Caleta, (0) Caleaatesy) e to the iput signal alone Calelte he nose poser accompanying rit. {6} Comment onthe efx of he noisy elerence onthe output SNR n10.60 = 2 [aie [-O—0 pb feterece fetmunteanuct Rea, Figere PES 852. Verily Eqs. (85-5] and (858) ‘48.3. (a) Show that the output SNR of DSB-SC signal which it synchronously detected is indepen: ‘ent ofthe IF bandwith; ie, Eq (848) independent ofthe IF bandit, {by Show thatthe output SNR ofan SSB signal which s synchronously detected is dependent ‘on the TF bandwidth Todo thi consider an IF bandwidth which extends fom fg ~ fy Jy ce > B'> 0, Cala the output SNR esing tis bandwith RSA. Inthe reid ampitade-modolated signal 2} = A(1 +m) 0s 2a i as the power "spect density Ga) specie in Prob. 83-4 The received signal accompanied by noise of poet ‘spectral deny 9/2. Calla the output signs-to-noe ati GL. Verily Fa. (86-8) by showing graphy that all the nelected terms have spectra falling tute the tans 15h 162, Given 2K + 1 spectral components af nese waveform space by intervals Af Show that the umber of pair of components separates by a frequency p Ais p= 2K ~~ p (Le veil the di sion lading to Ba (8615. 863. Io a DSB irmsinin, a cater of equency 1 MHz and of ampltode 2 vole amplitude ‘modulated to the extent of 10 percent bya snusldal baseband signal of requency 5 kl. The sigma ‘corp by white nls of two-sided power spectral Jensity 10-* wat/Hz. The demodulator ia[310 PRINCIPLES OF COMMUNICATION sys device whoxe inpuoutpat characterise is given by t, = 3e?, whee and eee respectively the ‘utput and inpt voltage The T Ste, before the demodlator, has a rectangle traser character ‘si uiy gui and LOH bandwith By ero, the I filter ane so that its center requene) ston Mite (6) Calulte the signt waveform at the demodulator output and cakulat its normaied (0) Cat the noise power tthe desnodulatoroutpet and the sigal-io-nise ratio. 64. Plt (in) (SN, versus PN in Eq (8.522) 20d show thatthe IG threshold acers when 6S. Assume that WG) = 01 and that SYN, = 30 W,. Use Ea. (1018-6) and plot NV/AN at function of Sind, wit [+ 18,8 = 4 (this coxespond to choosing t+ 2G = which represents fe sony good design W143, Find the thveshold extension posible for f= Sand 51 + Gy = 8 ASA, Refer to Fig. OLS. Show that if. de 0 noi it momentarily placed to «= ITI ot T/A then when the isturance subsides fg wl turn fo ty = Ta $018.2, Design a sine-ottage converter having the characteris that 20 past samples at wed to compute a that al sample resive equal weight, 40465, Show thats 9) f+ 1/9) and the PLL i djs 1 have a bandwidth 24, the PLL rl ele low iter eecursing a he rate with peak deviations ss haw 10.6.2 Repeat Prob, 1016 forthe Costas lop shown in Fi. 1016-2. CHAPTER ELEVEN DATA TRANSMISSION A data transmission system using binary encoding transmits a sequence of binary digits, that is, I's and 0's. These digits may be represented in a number of ways, For example, a 1 may be represented by a voltage V held for a time 7, while a zero is represented by a voltage — V held for an equal time, In general the binary digits are encoded so that a 1 is represented by a signal s(t) and a 0 by a signal sy(0), where s,() and s,(@) each have a duration T. The resulting signal may be transmitted directly or, as is more usually the case, used to modulate a carrier The received signal is corrupted by noise, and hence there is a finite probability that the receiver will make an ertor in determining, within each time interval, whether a 1 or a 0 was transmitted, In this chapter we make calculations of such error probabilities and discuss ‘methods to minimize them. The discussion will ead us to the concept of the ‘matched filter and comrelator. 11.1 A BASEBAND SIGNAL RECEIVER Consider that a binary-encoded signal consists of a time sequence of voltage levels + or —¥. Irthere is a guard interval between the bits, the signal forms a sequence of positive and negative pulses. In either case there is no particular interest in preserving the waveform of the signal after reception. We are inter- ested only in knowing within each bit interval whether the transmitted voltage was +V or —V. With noise present, the received signal and noise together will yield sample values generally different from +V. In this case, what deduction shall we make from the sample value concerning the transmitted bit?Suppose that the noise is gaussian and therefore the noise voltage has a probability density which is entirely symmetrical with respect to zero volts. Then. the probability that the noise has increased the sample value is the same as the probability that the noise has decreased the sample value. It then seems entirely reasonable that we can do no better than to assume that if the sample value is positive the transmitted level was +¥, and if the sample value is negative the transmitted level was —V-It i, of course, possible that at the sampling time the noise voltage may be of magnitude larger than V and of a polarity opposite 10 the polarity assigned to the transmitted bit. In this case an error will be made as indicated in Fig. 11.1. Here the transmitted bit is represented by the voltage -+V which is sustained over an interval T from ¢, tot. Noise has been superim= posed on the level +V so that the voltage » represents the received signal and noise. If now the sampling should happen to take place at a time ¢=1, + AV, an error will have been made. We can reduce the probability of error by processing the received signal plus noise in such a manner that we are then able to find a sample time where the sample voltage due to the signal is emphasized relative to the sample voltage due to the noise. Such a processer (receiver) is shown in Fig. 11.1-2. The signal input during a bit interval is indicated. As a matter of convenience we have set ¢ = 0 at the beginning of the interval. The waveform of the signal st) before ¢ = 0 and after f= T has not been indicated since, as will appear, the operation of the receiver during each bit interval is independent of the waveform during past and future bit interval, The signal s(t) with added white gaussian noise n{t) of power spectral density n/2 is presented to an integrator. At time ¢ = 0+ we require that capacitor C be uncharged. Such a discharged condition may be ensured by a brief closing of switch SW, at time t= 0 —, thus relieving C of any charge it may have acquited during the previous interval. The sample is taken at the output of the integrator by closing this sampling switch SW. This sample is taken at the end of the bit interval, at t= T. The signal processing indicated in Fig, 11.12 is described by the phrase integrate and dump, the term dump referring to the abrupt discharge of the capacitor after each sampling. inthe determination of transmted voltage level Figure 111-2 A receiver for a binary coded sina Peak Signal to RMS Noise Output Voltage Ratio. ‘The integrator yields an ovtput which is the integral ofits input multiplied by URC. Using ¢ = RC, we have [ [so + mo) ‘The sample voltage due to the signal is oT) = [era isn sf) (1-2) ‘The sample voltage due to the noise is n(T)=* [me ae re) ‘This noise-sampling voltage »,(T) is a gaussian random variable in contrast with nf), which is a gaussian random process. The variance of n,(T) was found in Sec, 7.9 [see Ea, (79-17) to he at m= (14-4) and, as noted in See. 7.3, nT) has a gaussian probability density ‘The output of the integrator, before the sampling switch, is (0 + ni). As shown in Fig. 11.1-3a, the signal output s,() is a ramp, in each bit interval, of duration 7: At the end of the interval the ramp attains the voltage s{T) which is +VT/ or ~VT/e, depending on whether the bit is a | or a 0. At the end of each interval the switch $1, in Fig, 11.1-2 closes momentarily to dis charge the capacitor so that s,() drops to zero. The noise nf), shown in Fig. 111-3, also starts each interval with m(0)=0 and has the random value nT) at the end of each interval. The sampling switch SW, closes briefly just before the closing of SW, and hence reads the voltage WAT) = 57) + nfT) es)mace) 1.3 (9) The Sa ouput and (the nie ouput ofthe integrator of Fig We would naturally like the outpot signal voltage to be as large as possible in comparison with the noise voltage. Hence a figure of merit of interest is the signal-to-noise ratio [my 2 ral (116) This result is calculated from Eqs (111-2) and (111-4). Note that the signalto- noise ratio inreases with increasing bit duration and that it depends on V°T- svhich isthe normalized energy ofthe bit signal. Therefore, a bit represented by a natrow, high amplitude signal and one by 8 wide, low amplitude signal are quay effective, provided V"T is kept constant Ieisinstrvtive to note that the integrator fers the signal and the noise such tat the signal voltage ineeass linearly with time, while the standard deviation {rms value) of the noise incresien more slowly, as a/T. Thus, the inegrtor enhances the signal relative to the nose, and this enfancement increases with time as shown in Eq, (11.18). 11.2 PROBABILITY OF ERROR Since the function of a receiver of a data transmission is to distinguish the bit 1 from the bit O in the presence of noise, a most important characteristic is the probability that an error will be made in such a determination, We now calculate this error probability P, for the integrate-and-dump receiver of Fig. 111-2 We have seen thatthe probability density of the noise sample nT) i gan jam and hence appears asin Fig. 112-1. The density is therefore given by sentTy) = (21 Jina where 02, the variance, is of = ni(T) given by Eq, (11.1-4). Suppose, then, that during some bit interval the input-signal voltage is held at, say, —V. Then, at the sample time, the signal sample voltage is sT) = ~VT/z, while the noise sample is nT). If n(T) is positive and larger in magnitude than VTi, the total sample voltage n(7) = s(T) + nT) wil be positive. Such a positive sample voltage will result in an error, since as noted earlier, we have instructed the receiver to int pret such a positive sample voltage to mean that the signal voltage was +V during the bit interval. The probability of such a misinterpretation, that is, the probability that nT) > VT/s, is given by the area of the shaded region in Fig. 112-1. The probability of ertor is, using Eq, (11.21) em [° ptoar ana Ceram oan : an Jah Defining x m(TV/3e, and sing E114, Ba, (112-2) may be ewrten a onde ete (GT) Sete (BY ates) zee) m3 in which E, = V°T is the signal energy of a bit. I the signal voltage were held instead at + during some bit interval, then it is clear from the symmetry of the situation that the probability of error woud again be given by P, in Eq. (11.2-2). Hence Eq, (11,2-3) gives P, quite generally Moot) Figwe 112-1 The gausian probability density of the noise sample m7)3 Figure 11.22 Variton of P vers En The probability of error P,, as given in Eq, (11.2-3), is plotted in Fig. 11.2-2 Note that P, decreases rapidly as Ey increases. The maximum value of P, is 4 Thus, even ifthe signal is entirely lost in the noise so that any determination of the receiver isa sheer guess, the receiver cannot be wrong more than half the time ‘on the average. 113 THE OPTIMUM FILTER In the receiver system of Fig. 11.1-2, the signal was passed through a filter (ie, the integrator), so that at the sampling time the signal voltage might be empha sized in comparison with the noise voltage. We are naturally led to ask whether the integrator is the optimum filter for the purpose of minimizing the probability of error. We shal find that for the received signal contemplated in the system of Fig. 11.1-2 the integrator is indeed the optimum filter. However, before returning specifically to the integrator receiver, we shall discuss optimum filters more gen- erally ‘We assume that the received signal is a binary waveform. One binary digit (bit) is represented by a signal waveform s,(t) which persists for time T, while the other bit is represented by the waveform s(t) which also lasts for an interval T. For example, inthe case of transmission at baseband, as shown in Fig. I1.1-2, sit) = +, while st) = —V; for other modulation systems, different waveforms are transmitted. For example, for PSK signalling, \() = A cos wot and 53() = =A 608 wt; while for FSK, 542) = A €08 (0%) + Oirand s,(0) = A cos (a»p — Oy. AAs shown in Fig. 11.3-1 the input, which is s(0) or s(t), is corrupted by the addition of noise (0). The noise is gaussian and has a spectral density G(s). [In most cases of interest the noise is white, so that G(f) = m/2. However, we shall assume the more general possibility, since it introdivees no complication to do 50] The signal and noise are filtered and then sampled at the end of each bit interval. The output sample is either nT) = s(T) + nJT) oF eT) = 5:(T) nT). We assume that immediately after each sample, every energy-storing element in the fter has been discharged. We have already considered in Sec. 2.22, the matter of signal determination in the presence of noise. Thus, we note that in the absence of noise the output sample would be 17) = 5,,(T) oF 5,3(7). When noise is present we have shown that to minimize the probability of error one should assume that s(t) has been transmitted if of) is closer to 84\(T) than {0 s4,(T). Similarly, we assume s3(0) has been transmitted if 1(T is closer to 5,3(T). The decision boundary is there= fore midway between s,,(T) and s,,(T). For example, in the baseband system of Fig. 11.1-2, where s,4(7) = V/s and §,3(T) = ~VT/t, the decision boundary is n{T) = 0. In general, we shall take the decision boundary to be T+s.AD 2 ef) ay ‘The probability of error for this general case may be deduced as an extension of the considerations used in the baseband case. Suppose that 5,(T) > s,3(T) and that 5,(0) was transmitted. If, at the sampling time, the noise nT) is positive and larger in magnitude than the voltage difference ¥sq\(T) + s.a(T] — 8,a(7), an error will have been made. That is, an error [we decide that 50) is transmitted rather than 5,0] will result if 32) Hence the probability of ror i 7 ami) (3a) sora oa, 20 Son as. ap ayo a Pentn — of rm Ld BL in. fry 5 z oT) Figure 141A seciser for binary coded signallingWwe make the substittion x a n(TH\/2ay, Ba (11.3.3 becomes erte Se") (i134) Note that for the case 5,(T) = VT/t and s,.(T) = —VT/r, and, using Eq, (I1.1- 4), Bq, (11.3-4) reduces to Eq, (11.2-3) as expected, ‘The complementary error function is a monotonically decreasing function of its argument, (See Fig. 112-2) Hence, a isto be anticipated, P, decreases as the difference s,(T) ~ 5,,(T) becomes larger and as the rms noise voltage , becomes smaller. The optimum filter, then, isthe filter which maximizes the ratio 2a =s4T) ca We now calculate the transfer function Hi) ofthis optimum filter. As a matter of ‘mathematical convenience we shall actually maximize y? rather than y Calculation of the Optimum-Filter Transfer Function H(f) ‘The fundamental requirement we make of a binary encoded data receiver is that it distinguishes the voltages s\() + n(t) and s,()+ n(t). We have seen that the ability of the receiver to do so depends on how large a particular receiver can make 7. It is important to note that + is proportional not to s(t) nor to s3() but rather to the difference between them, For example, in the baseband system we represented the signals by voltage levels + and ~V. But clearly, if ovr only interest was in distinguishing levels, we would do just as well (use +2 volts and Ovolt, of +8 volts and +6 volts, et. (The +V and ~V levels, however, have the advantage of requiring the least average power to be transmitted) Hence, while 5,0) oF s,(9 is the received signal, the signal which is to be compared with the noice, ie, the signal which ie relevant in all our error probability calevlations, isthe difference signal w= 50) 113.6 Thus, for the purpose of calculating the minimum error probability, we shall assume that the input signal to the optimum filter is p(t). The corresponding. ‘output signal of the filter is then. Pat) (113-7) We shall let P(f) and P,(f) be the Fourier transforms, respectively, of p(t) and Pa. IHC) is the transfer function of the filter, PAS) = HPL) (113-8) and nays [* raner arm [ menmnet ar 1391 ‘The input noise to the optimum filter is ni). The output noise is n,) which has a power spectral density G,(f) and is related to the power spectral density of the input noise GA) by GL = HDPGN (113.10) Using Parsevat's theorem (Eq, 113-5), we find that the normalized output noise power, ie, the noise variance 03, is £ ona [" IPGL ar (naan 9) and (11.3-11) we now find that AT) _ fa HPL (ee) JEeTHPGLD a Equation (11.3-12)is unaltered by the inclusion or deletion of the absolute value sign in the numerator sinee the quantity within the magnitude sign p,(T) is a Positive real number. The sign has been included, however, in order to allow further development ofthe equation through the se ofthe Schwarz inequality The Schwar: inequality tates that given arbitrary complex functions X(J) and YJ) of a common variable then From Eqs (11 Foxomnal’ sf ixmea [nea aisay ‘The equal sign applies when X= KY) (sta where K is an arbitrary constant and Y*() is the complex conjugate of ¥(/} ‘We wow apply the Schware inequality (o Eq, (11.3-12) by making the identi cation x= JG HU) osas and Y= Tap mem an346 Using Eqs, (11.3-15) and (11.3-16) and using the Schwarz inequality, Eg.(11.3-13) wwe may rewrite Eq, (113-12) as DE ROMOE 6 syne quanor. using Bq. (113-16), pT). ” Be (ene oa sf tren a-f Gay (113.18) The ratio p(TVo} will attain its maximum value when the equal sign in Eq, (11.3-18) may be employed as isthe case when X(f) = KY*(/), We then find from Eqs. (11.3-15) and (113-16) that the optimum filter which yields such a maximum ratio p3(Tya3 has a transfer function PD inet Hin= KEIR (113.19) Correspondingly, the maximum ratio is, from Eq. (11.318), a i Pune 5 [ @ daw dew Gal) a In succeeding sections we shall have occasion to apply Eqs. (11.13-19) and (11.13.20) to-a number of cases of interest 114 WHITE NOISE: THE MATCHED FILTER AAn optimum filter which yields a maximum ratio pi(TVe2 is called @ matched filter when the input noise is white. In this case G,(f)= n/2, and Eq. (113-19) hecomes Hify=K ) nur (114-1) ap ‘The impulsive response of this filter, ie, the response of the filter to a unit strength impulse applied at ¢ fe) = FLAN = f Pape PeTe df (114-20) ok _ Prien ay (11.426) AA physically realizable filter will have an impulse response which is real ie, not ‘complex. Therefore f(s) = h*(). Replacing the right-hand member of Eg, (11.4-2b) by its complex conjugate, an operation which leaves the equation unaltered, we have 9 = 7K [" ppemnr-nay (11430) 2K wr 9 (114.35) -+vereeeummemnnaremnneemnnensmnesmmmnenmcwd Finally, since pt) = 50) — s(t) [See Eq. (11.3-6)), we have wy rp ser an ay worn pcos un tae 0 Fialyhe com a= ingen impute oat ote mated ther cosa comes Figere 144 The signals (0 (0 (9 hand (6) =) ~ 24) Ud) tated aboot the faint = (The waveform i tresited 0 "he gh hy aman 7able. We may note in passing, that any additional delay that a fier might intro- duce would in no way interfere with the performance of the filter, for both signal and noise would be delayed by the same amount, and at the sampling time (ochich would need similarly 10 be delayed) the ratio of signal to noise would remain unaltered, 115 PROBABILITY OF ERROR OF THE MATCHED FILTER The probability of error which results when employing a matched filter, may be found by evaluating the maximum signal-to-noise ratio [p2(TV2]u, given by Fg. (113-20), With G,(f) = m/2, Eq. (11.3-20) becomes [BP] $2 [7 inne a ais From Parseval’s theorem we have In the last integral in Ea, (11.5-2), the limits take account of the fact that pl) per- sists for only a time 7, With p() = s() — s,(, and using Eq. (115-2), we may write Eq, (115-1) as [2] 2 [toto ~ sso (15-30) 2[ [sows ['aoa—2 [omsnd rss oj la + Ba 2B) (118339) Here F,, and E,, are the energies, respectively, in s(t) and s,(@, while Ej. is the energy due to the correlation between s,(0)and s3(), Suppose that we have selected s,(2), and let x(t) have an energy E,s. Then it ‘can be shown that if s,() is to have the same energy, the optimum choice of s,(t) sx = —s4(0) (154) The choice is optimum in that it yields a maximum output signal p(T) for @ eiven signal energy. Letting s;() = —sy(0), we find z, -E and Eq, (11.5-2c) becomes (11s) le Rewriting Eg. (11.3-4b) using p(T) p= ete [PAT aio, ‘Combining Eq, (11.56) with (115-5), we find that the minimum error probability (P nie otresponding to a maxiraum value of p3(T a3 is up} \" cen (158) sa T) ~ Sys) we have (1s.6) (aunt! We note that Eq, (11.58) establishes more generally the idea that the error probability depends only on the signal energy and not on the signal waveshape, Previously we had established this point only for signals which had constant voltage levels. We note also that Eq, (11.5.8) gives (P,jae for the case of the matched filter and when s,(t) = —s3(0. tn See. 11.2 we considered the case when s(t) = +¥" and s(t) ="~V and the filter employed was an integrator. There we found [Eq, (11.2-3)] that the result for P, was identical with (P,)qi, given in Eq, (115-8) ‘This agreement leads us to suspeet that for an input signal where s\()= +V and s4(0) = —V, the integrator is the matched filter. Such is indeed the ease. For when we have si(0) and = vo ostsT (115-90) vo OstsT (11.98) the impulse response of the matched filter is, from Eq. (11.44), HO 2B g(t 9-70) (115-10) The quantity s,(7 =) —s,(T — 1) is a pulse of amplitude 21” extending from 0 107-= T and may be rewritten, with uf?) the unit step, w= ® arte wssin ‘The constant factor of proportionality 4K V/y in the expression for A) (that is, the gain of the filter) has no effect on the probability of error since the gain affects signal and noise alike. We may therefore select the coefficient K in Eq, (115-11) so that 4KV/n = I. Then the inverse transform of Mf), that is, the transfer fune- tion ofthe filter, becomes, with s the Laplace transform variable, (15-12)The first term in Eq (I1.5-12) represents an integration beginning at ¢ = 0, while the second term represents an integration with reversed polarity beginning. at r= T. The overall response of the matched filter is an integration from 1 to t=T and a zero response thereafter. Ina physical system, as already 1, we achieve the effect of a zero response aher t= T by sampling at $0 that so far as the determination of one bit is concerned we ignore the response after t= T 11.6 COHERENT RECEPTION: CORRELATION We discuss now an alternative type of receiving system which, as we shall se, is identical in performance with the matched filter receiver. Agsin, as shown in Fig. 1146-1, the input is a binary data waveform s,() or 5,0) corrupted by noise (0. The bit length is T. The received signal plus noise (0) is multiplied by a locally generated waveform sy(t) ~ s(0). The output of the multiplier is passed ‘through an integrator whose output is sampled at ¢ = T. As before, immediately after each sampling, atthe beginning of each new bit interval, all energy-storing elements in the integrator are discharged. This type of receiver is called 2 correla. to, sige we ae correlating the esived signal and noise with he waveform 5) ‘The output signal and noise of the correlator shown in Fig. 11.61 are sD = i, SHOTS (0) — (0) de (116-1), na) = [mots ~ ston ae (116.2) ‘where s(t) is either 5,1) or 5,0), and where ris the constant of the integrator (ie., the integrator output is I/+ times the integral of its input). We now compare these ‘outputs with the matched filter outputs, stent) a we 24a Sent eat THH MAT) sai oy = Fire 11441 A coberen sytem of signal reception 1 H() isthe impulsive response of the matched fite, then the output of the ‘matched filter 0) can be found using the convolution integral (see Sec. 1.12). We have vin [ramya= [uma ey ‘The limits on the integral have been changed (0 0 and T since we are interested in the filer response to a bit which extends only over that interval. Using Eq, (114-4) which gives Wo) for the matched filter, we have uo =2E for =~ sg = (164 oe ee Substituting Eq. (11.6-5) into (11.6-3), we have v9 = 28 [nats 14 4 S(T 04 Ma (1166) Since o(2) = s() + n(2),and 2,(0) = 5) + nt, seting ¢ = T yields 167 wun =2 [nae 4a a ues Thus 57) and n,{7), as calculated from Egs.(11.6-1) and (11.6-2) for the eor- relation receiver, and as caleulated from Eqs, (11.6-7) and (11.6-8) for the matched filter receiver, are identical. Hence the performances ofthe two systems are identi- cal ‘The matched filter and the correlator are not simply two distinet, independent techniques which happen to yield the same result. In fact they are 1Wo techniques, of synthesizing the optimum filter ho) 11.7 PHASE-SHIFT KEYING ‘An important application of the coherent reception system of Sec. 11.6 is its use in phase-shft keying (PSK), Here the input signal is cos wot i, or sl) = =A c08 aot (172 s(t)modulating baseband signal. For SSB modulation W is the bandwidth of both the modulating baseband signal and the modulated carrier. However, all of the ‘other eases are doubled sideband systems and hence: a wed (11.201) B being the (two-sided) bandwidth of the modulated carrier. For the sake of having a uniform basis of comparison, we have in every case tabulated the error probability P, 0, se Pas ‘Shannon's iit om wesk 1M =4 (QPSK, MSK) “asymptote value for M +20 Figure 1.204 BinHe ws. Ey for Probaity af eeor = 10°% Sn Tn connection with a communication system which is to be designed, it is ‘common place that, in addition to such practical constraints as cost, volume, ete there be a specification on allowable error rate, In such circumstances, plots as shown in Fig, 11.201 are very useful. Here P, is fixed at P, = 10~*. As we shall see in Sec. 13.7, there is an ultimate limit (0 the performance of a communica tions system. This limit, deduced by Shannon, is given by the inequality Eevn[1+(2(2) and is independant of the error probability. We have included a plot of Ea (11,20-2) in Fig, 11.20-1, In Fig, 11.20-1, as well as for any given error probability. we shall always find that all modulation schemes yield plots which lie to the right of Shannon's limiting plot Figure 11.20-1 shows that QAM more closely approaches Shannon's curve than does MPSK, indicating that QAM is a more efficent system for operation in a white gaussian noise environment, Note that when we compare MPSK with MESK (see Table 11.20-1 and Fig. 11.20-t) we find that as M increases the band: Width of MPSK decreases while the SNR required to obtain P, = 10-* increases However, with MFSK the bandwidth inereases While the SNR decreases. In all ‘eases however we note that the slope of each curve is postive, so that an increase in AWW requires an increase in Fyn if a constant bit error rate is to be main tain, i, for a constant error rate (11.202 (11209 If the error rate were, for example, P, = 10~* each curve would shift approx imately 1 dB to the right, except of course, for Shannon's curve which is indepen- dent of P, REFERENCES 1 Stein S. and J Jones: "Modern Communistion Principles,” McGraw-Hill Book Company, New York, et 2. Schwartz M. W. R. Benet, and S. Stein: "Communication Systems and Techniques” MeGra 5. Worenea Jy an. Jneabe:" Commaniaion Engineering" Jahn Wiley and Sons, New York 1966, PROBLEMS 1.4, (0) Find he power spectral deny Gf noite which hasan autocorrelation function (0 The noise in (i apie 0 a icegrtor at = 0. Find the mean square valve [RATIP of the nie out o he negrtor al? = 7€2-The mis ina acompani sgl which oni f ihr the vole + ¥ othe ose ¥ snd fora tne T- Al tine = "Tn the ao the neste ete siglo theme aie vlog 1142, A recived sip A) $Y i eld for antral, The sal & scompanid by white isin soe of perspec! densy 2. The reed signal io be proceed ip 18 However. as an afjexiton othe equred egrator we aloe RC ero oa Nand wiih Cte he alo for sch he a se volgy the npg el De ‘masini Fov thistle off elt the ga o-mse a and compare wh Eg 18a) {hh ani when an ite et Show tht fr he RC weve game aio ‘oat G8 sar then or gato 1124, 4 imal which can nme one ofthe vokage + or ~P is ransmite. Connie that he probity of tsnting + Ff ne he ebay Hanming is The spa ezopaned by ite pon noe { Asume that he etl sla for desi beeen the wo posible nig i ahr than owes, Writean exe forte profi tah eor dy Rt he made !stote wn dnp oe el ini 1) Find Foch tha he probabiy of e072 simu and eel the creping probability of error, 7 as 12.2.4 anal shih can tak on he votaes 47.0, = wth equ eto aie Wien resid it isembeded in white gaussian nae, Theveciverintegrtes he ial and ee or aime Pe Wie an expres othe test vole 1s ht the probaiiy of ere indepen eso wich garam 2.8 A reed signal ier 42¥ of —2V held fora ime 7. The sili corp by wie {Bian ase of power spectral ety 10-* vl Hie oral posed by an meet an Srp reive, wht he mii ine T during wich signal tbe susie i he pa ity of error is not to exceed 10°47 m N34 A uanmier ante the ene wih ual prubiiysthe chanel nie has he power seta denny Ou) = Gal * Ud (Find the ane fncton Hohe mathe fier, and comment on its rear 1) Find he vera prota of enor when sige athe te {0} An inegrr employed ther tan the opm ited its Pand compat wih 2 WS. & sil ieither 56) =A 08 ft of 4) 0 foram intral Tf wth nm eget ‘he nol cvrped by iene wih Of) = 92 Fin the ander on fhe meted Ser oh spat Wate am exresion othe sbatiity fr , 2. Repeat Prob. 1151 he seal ia) = +l ~ on fh Wo, Compare the outs fhe MF andthe coeltor, when the net signal either ¥. a4 funcon nf te «for 01 T_Asue whe pusan ce Arete utp he same oa tat when =? 6A. A sgl i) = 41/7) fr O15 T The seal corte by white usin ne of rower aptly 10" ee {c) Draw the sii water he ou of mace ler ese. 10 he pofay eo Pt Be ao lage than 10" fn te nium Alora ner war! 1.71. Very Eq (1125, 1184. Pot Eg (118) EJ M2 Hie fequeny fet 2 ie FSK sales OF = nx, and sae oops (ol Prove he stoment ( Caoute (0) Plt es En and compare withthe eu gen in B18, AA Pot he Pin inary FSK sea Non of Set By = 18. 14941, Plot Eq (11.9.1) serus En IMA, Mary PSK involves choosing signal ofthe Form (ot yt + )forM values of {o) Show how to shore te 50 tha he probability oferor enh the me. (6) Find the creat detec (6) Obtain am expreson forthe probabiity of ero. 11.2 Verify the entvies in Table 1.1 AA. Very Bq (11122), 4, Refer 1 Ea. (11.150 and &) Determine oy ~ om sch hat the uit vectors) and se orthonormal 11132. (0) Hay) = JWT cos nt 40) and ft) IT cor ot + 9) where © and 6 are Independent random variables uniformly distributed between ~ t= 6 #

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