DIGITAL TRANS ge OF ANALOGS SGN 10.1 INTRODUCTION Communication systems are designed to handle the output of a variety of information sources. In th we considered analog com- i CW modulation schemes (AM. DSB, SSB, PM, and the use of digi communication systems such as the one hows in ‘re 10.15 for transmitting the output of analog information sour ae Digital transmission of analog signals is possible by virtue he sampling theorem which tells us that an analog signal can be scoring maperny *Ppropriate set of its samples and hence we need transmit ie ot the aaalse Values as they occur rather than the analog signal itself. ae saereiitia aural can be transmitted using analog pulse modul ease rs oportion 19 thlitude, width, or position of a Pulse waveform is varied in propor 2 values of the samples. The key distinction betusen. a some parameter of ne ad CW modulation is as follows: In CW modul ati ein analog pulse ednedulated wave varies continuously with the reset by a partic railation, some Parameter of each pulse is mo ("ple value of the message. Fi 505Digital Transmission of Analog Signals 506 s ‘Analog communication system ——m{ Channet LY Demodulator KO. Analog Analog ou input me i (a) — ‘Analog ‘Analog information modulation source scheme [<— Distal communicaton system —+| Digital modulation scheme Digitay output (o) Figure 10.1 Communication systems. (a) Analog communicati gystem for transmitting the output of an analog information source {b) Digital communication system for transmitting the output ofa discrete information source. Another method of transmitting the sampled values of an analog signal j round off (quantize) the sampled values to one of Q predetermined a and then transmit the sampled and quantized signal using digital motu tion schemes. The block diagram of a system that uses this scheme is shown in Figure 10.2. Here, the output X' (t) of the analog information source is converted to an M-ary symbol sequence {S,} through the processes of sampling, quantizing, and encoding. The M-ary sequence {S;} is transmitted using a digital communication system. The receiver output {$,} will differ occasionally from the input {S,} due to transmission errors caused by channel noise and ISI in the digital communication system. An estimate X(t) of X(0 is obtained from {S,} through the process of decoding, and digital to analog (D/A) conversion. The reconstructed waveform X(t) will be a noisy versioa of the transmitted signal. The noise is due to sampling and quantizing of Xi, Decoder and Digital 7 ie communication Fed SG See converter xie) ts, L a) {Sr} inuout pare ag Discrete Discrete Cont “Sonal Fendom random Sal eotence sequence Figure 10.2. Digital transmission of analog signals:oe 9 Theory ang "ractica — so7 ets mbol ol that occur in the digi ) a performance of this system is Mena! Onithun; avetal Te receiver omput. A. major portion ed by the sigeatiO® 8Ystem, ti? 8 a sampling, quantizing, and Gicodins this ¢ ben Moola. : The ka Jog output of an ij oding Tis d ¢ analog PI n information re: n Source j at are ug ane overt guitable for transmission over a digi into a inning of this chapter, we wit, Bital co, j a ont J tems | for tral i val Se eeeahe Gi ommunication -pressions for the signal-to- sysi aw oie expressions ‘ch ne Nols power fats rae Finally, we with t mparing th © rece} © perform: Wer output a id use ee | transmission schemes. We will also poi ince of digit; sal schemes for transmitting analog information” the advantages of w = In pulse communication systems, both analog and digi schemes are used. Analog pulse modulation, such as ae Pulse modulation plitude modulation and pulse position modulation, are ue pulse am- Pexponential CW (PM or FM) modulation schemes. Digital to linear (AM) modulation schemes such as pulse code modulation (PCM) cae a pulse (DM) have no CW equivalent. We will treat only the acu lation i é modulation schemes in this chapter. We begin our study with a review of sampling techniques. miaecet 10.2 SAMPLING THEORY AND PRACTICE In many applications (such as in sample data control systems, digital com- modulation systems that we are puters, and in discrete pulse and CW j currently dealing with) it is necessary and useful to represent an analog signal interms of its sampled values taken at appropriately spaced intervals, In ths section, we will first consider the representation of a low pass (bandlimite ) deterministic signal x(t) by its sampled values (kT) (k= oa ie 0,1,2,...), where T, is the time between samples. We bas te wel as (0 Concept of sampling to include bandpass determinisy sig reconstruction tandom signals. Finally, we will point out how the sampling an Of analog signals are carried out in practice. 1 9.21 Sampling Theory The prings in Re teile of sampling can b ‘gure 10.3. The switch perio ‘ Jer shown Hy tching SMP ate of ed using BES cota explain pest ifts betwee” dically shi i |8 owe Electronic switeh ee Figure 10.3 Switching sampler, f. = 1/T, Hz staying on the input contact for 7 seconds and o, contact for the remainder of each sampling period. The out, sampler consists of segments of x(t), and x,(t) can be x(t) = x(t)s(t) where s(t) is the sampling or switching function shown in Two questions-that_need to be answered with the sampli in Figure 10.3 are: (1) Are the sampled ségments sufficient to describe 4 original signal x(t)? (2) If so, how can we reconstruct x(t) from s(t)? These questions can be answered by looking at the spectra (Fourier transforms) X(f) and X,(f) of x(t) and x,(t). Using the results derived in Chapter we can express s(t) as a Fourier series of the form __ N the Put x, Tepresenteq OF the Bound, : (10) Figure 10.4, 'ng scheme sh \ s()= Cot 3S 2G, cos most (ing where ' Co=7/T,, Cy =f.7 sinc[nf,r], and «, =2af, Combining Equations (10.2) and (10.1), we. can write x,(t) as x,(t) = Cox(t) + 2C,x(t) cos w,t +2C2x(t) cos 2a,t ++** (103) Input Output x(e) x,(t) = x(edste) Sampling funetion s(¢) cof sampling Figure 10.4 Sampling interpreted as multiplication. This tyP® is often called natural sampling.Sa in Mpling Theory Nd Practice 509 ansform of Equation (10.3) yields sop trate gout X.(f) = CoX (A) + CUX = f,) +X4 py) +CAXG = 2) + XY 4 2p 914! : (10.4 =OoXN+ D OX — nf, : nen (10.46) ne ation (10.4a) to find the spectrum of x, (t) given th us ws the spectrum of the sampler out, e wert re 10.5 sho Pectrum of . Put when the input x(t) is ze i nite : . Equation (10.4a) and from Figure me ollow’s : ing operation leaves the message s; i ec an ee dically in the frequency domain with a period of f.. We also wate it re rst term in Equation (10.4a) (corresponding to the first term in we that a0 3) is the message term attenuated by the duty cycle Cy of the sean pulse. Since the sampling operation has not altered the message ampli (10.56) that if fi >2f,, Pectrum intact, merely xi =o, he OT cin, (a) MOS in treaueney ath ‘of the samp! sage. (0) sami ring pulse * Figure 10.5 Sam; ition shown pling opera’ fae Cubput, f,>2f,. (¢) Sampled output, f- <7! “stumed to be much smaller than Te.)510 Dignan . be possible to reconstruct x(t) fro, mm Pca procedure for reconstruction is not ops , ( Mionships, it can be seen from Figure 19,5), P¥ious irogtt from X,({) by lowpass filtering. If we can filter X( X(f) calte Wihave recovered x(f)- Of course, stich recovery ig Pore, from y be then We indlimited and f,>2f. If the sampling rate f, <2f, ie xo) is : the signal overlap (Figure 10.5c) and x(ry cannot en the bam distortion from X«(). This distortion is referred to ma get Mem! ie the sampling frequency f, must satisfy lasing “4 fr22fe or T,<1/2f, spectru form Xs domain re! separated ™ the na ing frequency f.,,, = 2f: is called the Nygy;, (l03) ¢ minimum sampling ‘min Yquist eaten (10.5) is satisfied, x(t) can be recovered by passing’ (yt Mia jdeal lowpass filter with a bandwidth B, where B satisfies TOURH ay fe a(t - kT.) - (109) = D x(kT,)8(t — kT.) and ae cat Xs(f) = X(f) * Ss(f)"9 Theory Nd Practicg 511 Sf) = fy x 5 ~ ng, ere (10.11) of XO=L ES XG— mf) (10.12) .4b), We see that the Only difference ig 12) and (10.4) c Bastion (1002) ane (10.48) ate equal tof, in come constants reconstruction of x(t) from the or perfec! iat for PC us ‘uation (10.12), %8(t) We invol © the same ring x(t) from x(t), that is, x(t) must be S we had ie 2f, Then, we ean reconstruct x(t) from x(0) by tions to fe and fy ideal’ lowpass filter Hy(f) with a bandwidth B snl through a i shown in Figure 10.6. Next, if 7 let the filter psi ah 2f,. We state this result = fry i theorem: the flog The Uniform Sampling Theorem for Lowpass Signals If a signal x(t) contains no frequency components for Ifl> 4, thos, completely described by instantaneous values x(kT,) uniformly ca: itis with period T, < 1/2f,. If the sampling rate f, is equal to the Nyaa in time greater (f, =2f,), and if the sampled values are represented by Gea impulses, then the signal can be exactly reconstructed from its samples y y ideal lowpass filter of bandwidth B, where f, < B 2a wil 223+ is shown in Figure 10.8. The reader can easly veri) tT “sult in exact reconstruction, Proof of exact reco \ds on the ratio fad Bo Ie- ang ys fh Figure 10.10 Sampling of nonbandiimi aS. | fed | 1 Signals, 10.3 QUANTIZING OF ANALOG SIGNALS In the preceding sections we established the fa 7 signal can be adequately represented by its sam ae aN analo such as speech waveforms or video waveforms ae values, Me, ety range and hence the samples are also continuous in a Continuous aya continuous amplitude samples are transmitted paar lita receiver cannot discern the exact sequence of transmitted vy cha of the noise in the system can be minimized by representin Values. The g a finite number of predetermined levels and transmitting rai Samples jy discrete signaling scheme such as discrete PAM. Nov, if bles Using 4 between the levels is large compared to the noise perturbations, te simple matter for the receiver to decide precisely which specific = = transmitted. Thus the effect of random noise can be virtually ainsi Representing the analog sampled values by a finite set of levels is called quantizing. While sampling converts a continuous time signal to a discrete time signal, quantizing converts a continuous amplitude sample to a discret amplitude sample. Thus sampling and quantizing operations convert te output of an analog information source into a sequence of levels (or symbol, hat is, the analog source is transformed to a discrete (digital) soure, equence of levels can be transmitted using any one of the a ignaling schemes discussed in the preceding chapters. An example mantizing operation is shown in Figure 10.11. resens The input to the quantizer is a random. process X(t) that ta xo utput of an analog information source. The rand KckT) a ampled at an appropriate rate and the sampled vane che preset 9 one of Q allowable levels, m1, M2- ++» ma according a ( } X,(kT,) =m if xy XKT) <¥ a= 0, XO HR ule:SE signals 619 ‘Actual value of the signal *4In) Giantiving co Quantized value ‘of the signal geec (xkT,)) [cone | {X_(kT)) put levels of the mf igure 10.11 OU quantizer. antizing operation; Mm ™Mz---* m; are the seven oul ‘The output of the quantizer is a sequence of levels, shown in Figure 10.11 252 waveform X,(t), where X,(t) = XakT)> kr, dx ~a+(i-1)a 2 a3 (a) a2 =T since QA = 2a. Now, the output signal power S, can be obtained using Equation (10.27): S,= > cmi?(3) 2 a “ay and hence the average signal to quantizer noise power ratio is S. Pa — Or N, Qe-1 02) =Q*, when Q>1523 q (Sal Naan = 20 : 78) indicates that th log. Q s 0. i s that th iy avtt® Cf quantizer levels. If FNC elity of the nme Toon the output Xq can be hed nee Oren incre (10.285) imo , of levels (Q) is determined p: le a8 near acl of g ssi with @, 1 If the variance is 7%, then the preceding expression becomes 03) N, ~(2.2)0%Q** ne 5, 3 = ok, am Now, if we assume X to have zero mean, then S, mE(X YH oH (103) S,|Nq ~ (0.45)Q'* sizer . in Equation (10.33) can be used to determine the num ber of quan. ; : ‘se powe! needed to achieve a given average signal to quantizer noise P40.4.1 The PCM System A PCM communication system is shown in Figure 10.18. The analo, x(t) is sampled and then the samples are quantized and encod 8 signal purposes of analysis and discussion we wil] assume that the corral t 2 binary sequence. In the example shown in Figure 10.17 the binary code has a numerical significance that is the same as the order assigned to the quantized levels. However, this feature is not essential. We could have used arbitrary ordering and codeword assignment as long as the receiver knows the quantized sample value associated with each code word. The combination of the quantizer and encoder is often called an analog to — a32 Digital Transmission of Analog Signals 5: g Xgl =X (04 eye) ass AID converter, Lowna ie Matched filter Sli) = Xo ltl + ng lt) + molt) Figure 10.18 Block diagram of a PCM system. na(t) is the noise due to quantizing and no(t) is the noise due to bit errors caused by channel noise. digital (A to D or A/D) converter. The sampler in practical systems is usually a sample and hold device. The combination of the sample and hold device and the A/D converter accepts analog signals and replaces it with a sequence of code symbols. A more detailed diagram of this combination, sometimes called a digitizer, is shown in Figure 10.19. ‘The digitally encoded signal is transmitted over the communication channel to the receiver (shown in Figure 10.20). When the noisy version of this signal arrives at the receiver, the first operation performed is the separation of the signal from the noise. Such separation is possible because of the quantization of the signal. A feature that eases this task of separating signal and noise is eames ae area L-—2] Comparator Analog input — End of {amp voltage conversion ecoae [| senerator command Binary counter Clock signal = = = Codeword Parallel Transmit 2 re Teorey converter PCM output Figure 10.19 Elements of a PCM transmitter.533 matched filter receiver synchronization ie H Sample and rraioa | ie output Figure 10.20 Elements of a PCM receive, thot during each bit interval the receiver (matched filter) has only to make the sanple decision of whether a 0 or a 1 has been received. The Telative reliability of this decision in binary PCM over the multivalued decision required for direct Q-ary PAM is an important advantage for binary PCM. After decoding a group of binary digits Tepresenting the codeword for the quantized value of the sample, the receiver has to assign a signal level to the codeword. The functional block that Perfo: i A converter as a Qk Revd ‘o reject any frequency components lying outside of the baseband. * reconstructed signal X(t) is identical with the input X(t) except for the ‘antization noise n, a(t) and another noise component no(t) that results from ‘coding errors due to the channel noise. Figures 10,18- 10.20 do not show signal companding components, and timing "covery networks, 10, ite Bandwidth Requirements of PCM ar singe M requires the transmission of several digits ne oe than the Mestage 48 @PParent that the PCM bandwidth will be mus seobiaied 38 taloye’ P2CWidth, A lower bound on the bandwidth can apes occur at Tate f(g IRE Message bandwidth is. then the quantized nn svt “C2E) samples Per second, If the PCM system uses534° Digital Transmission of Analog Signals (M-ary transmi ission) to represent the Q quantizer levels, then ea would consist of digits, where y =logu(Q), M=2yf,. Recallin ei gttiate discrete baseband PAM signalling we need a bandwidth > 7/2 Hy, wy. 7 for the bandwidth of the PCM signal as tain Boem > Vfx (10,34) For binary PCM, the bandwidth required is greater than or equal to filo As an illustration of the bandwidth requirements let us consider tine 9@2@ transmission of telephone-quality voice signal. While the average voice .°! trum exceeds well beyond 10 kHz, most of the energy is concentrated in ia range 100 to 600 Hz and a bandwidth of 3 kHz is sufficient for intelligibility a standard for telephone systems, the voice signal is first passed throug 3 kHz lowpass filter and then sampled at f, = 8000 samples per second, Each sample is then quantized into one of 128 levels. If these samples are trans. mitted using binary PCM, then the bandwidth required will be larger than (8000)()(log: 128) = 28 kHz, which is considerably greater than the 3kHz bandwidth of the voice signal. 10.4.3 Noise in PCM Systems It is shown in Figure 10.18 that the output X(t) in a PCM system can be written as X(t) = Xolt) + ng(t) + not) (10.35) where X(t) = kX(t) is the signal component in the output; ng(t) and no(t) are two noise components. The first noise waveform n,(t) is due to quantization and ‘the additional noise waveform no(t) is due to the additive channel noise. The overall signal-to-noise ratio at the baseband output, which is used as a measure of signal quality, is defined as (8) =a a (10.36) Nio EX{ng(t)}}+ Ef{no(t)} The average noise power at the output, E{{ng(t)/7} and E{{no(t)}?}, can be calculated as follows. Quantization Noise in PCM Systems. If we assume that ideal impulse sampling is used in the PCM system, then the output of the sampler isq "99 Signals a 535 t)= X X(t) (t) ot ~kr) 4 signal Xie 69 then be expressed ag wed § Patt wr x, (0) > S(t = kT.) 1 = XW) BBE KT) +X (0) ~ XOLD 8¢¢~ pp 2 SIXT )8(t ~ kT) + ; zt Y+ ea(kT,)8(t — kr) ‘ ror introduced by the id 1) is the en the quantizing operat we a ed in Chapter 3, we can obtain the power sient ace Using the els G.*=-2 Fe nity of eg as ca) = 7, Elea(kT,)} (10.37) ing. that Bleg(kT.)=0 and Exeq(kT,)eq weed error due to quantizing, EXe{(kT,)}, will depend on the si i sand the method of Seale Oi Purposes, let fa san" quantizer operating on X(t) having a uniform pdf pia oat Then we have Pat over the interval LK +)TD=0. The mean Efe%(kT,)} = A712 where A is the step size, and a1 oo 4(8) If we ignore the effects of channel noise temporarily, then the noise com- ponent n,(t) has a power spectral density G6) = GPR OP where Ha(f) is the transfer function of the lowpass filter used for reconstc- tng the signal. Assuming f, = 2f, and Hp(f) to be an ideal lowpass filter with a bandwidth f,, we have G.f) MflIn the transmissi able Portions of TaNsmission al has high sample to sample correlati icture (video) information, appreci the «i i « feround information containing Very little tonal Vari: “ral describe tions, if we use PCM, the codewor ti ations. In such situa- P ‘ds describe the value of the average tackground level; if these tonal values do Not change ay rae ppreciably, then we are essentially transmitting repeated sample values, One way to improve the situation is to send only the digitally encoded differences between successive samples. Thus a picture that has been quantized to 256 levels (eight bits) may be transmitted with comparable fidelity using 4-bit differential encoding. This reduces the transmission bandwidth by a factor of 2. PCM systems using differential quantizing schemes are known as differential PCM (DPCM) systems. A differential PCM system that is particularly simple to implement be When the. difference signal is quantized into two levels. The output of a Guantizer is represented by a single binary digit, which indicates hae the sample to sample difference. This PCM system is known as a PCM tion (DM). Delta modulation systems have an advantage over Ma : and M-ary DPCM systems in that the hardware required sale. the Wansmitter and demodulation at the receiver are much si Was Delta Modulation Systems ; tem is show" in a he functional block diagram of a delta modulation Vy Xx(0 is oom ized into 2, nore Hue X(T) Oe qua Aprediet Me transmitter, the sampled val iy — XRTD Ictes d value X(kT") and the difference X(Quantizer + Uxth) = 0 g (a) Transmitter” (6) Receiver Figure 10.22 Discrete time model of a DM system. (a) Transmitter, (b) Receiver. Sampling rate = f; = 1/T;. one of two values +A or —A. The output of the quantizer is encoded usin one binary digit per sample and sent to the receiver. At the receiver, fe decoded value of the difference signal is added to the immediately Preceding value of the receiver output. The operation of the delta modulation scheme shown in Figure 10.22 is described by the following equations: X(kT) = X(k- DTD (10.44) where X((k — 1)T) is the receiver output at t = (k —1)T! and X(T) = X(kTD + (X(kTD — XkTY)q = Xk - DT) £A (10.45) The delay element and the adder in Figures 10.22a and 10.22b can be replaced by an integrator whose input is an impulse sequence of period T’, and strength +A. This results in the system shown in Figure 10.23. The operation of the delta modulation scheme shown in Figure 10.23 may be seen using the waveforms shown in Figure 10.24. The message signal X(t) is compared with a stepwise approximation X(t) and the difference signal Y(.) = X(t)— X(t) is quantized into two levels +A depending on the sign of Hard s(t) = 5(t—kT;)- limiter i Low ——} Integrator }>} pass FL ¥eqle) L__] Xt) fitter | X00 Difference —| filter amplifier Xe) | fitter YaalA) Ke) = Xie) Figure 10.23 Hardware implementation of a DM system, (a) Modulator. (b) Demodulator.>. — SNalg 544 Slope Overload Figure 10.24 Delta modulation waveforms; X(t + X(t) = X(t), : giflerence- The output of the quantizer is sampled to rode t ce Yio(t) =D A senkX (kT) — XEKTI|I(t— kT : y .46) stepwise approximation X(t) is generated by passi _ He ae in Equation (10.46) through an intra it Aetee impulse with a step rise. Since there are only two possible impulse weighs Yet)» this signal can be transmitted using a binary waveform. The demodu. jator consists of an integrator and a lowpass filter. Jnapractical delta modulation system, the lowpass filter in the receiver will, by itself, provide an approximate measure of integration. Hence we can i diminate the receiver integrator and depend on the filter for integration. At | the transmitter, the sampling waveform sg(t) need not be an impulse wave- form but a pulse waveform with a pulse duration that is short in comparison with the interval between pulses. Furthermore, the transmitter integrator need not be an ideal integrator—a simple RC lowpass filter will be adequate. These simplifications reduce the complexity of the hardware in DM systems con- siderably. Some of the problems that occur when we usé delta mo italy, an analog signal can be seen in the waveforms shown in a | Ietus assume that X(t) < X(t) so that the first impulse “ : When this impulse is fed back through the integrator a P change in X(t) of height A. This proces’ continues (rir al the recelet inet und 0) eeneds (0). During the $1005 (pA tes Utput will differ considerably from ! ins or behavior when jet nodulation y the messase SEN 7) remus Tcpate P period, X(1) exhibits a hunting bem nting leads to idling noise. The sampling modulation to transmit542 Digital Transmission of Analog Signals much higher than the Nyquist rate and hence the rectangular il ally be will normally form can be filtered or smoothed out by the receiver filter idling noise wave! erioading. A serious problem in delta modulation schemes ari ae the rate of rise overloading. When X(t) is changing, X(1) and oF follow X(t) in a stepwise fashion as long as successive samples of X(t) go tet differ by an amount greater than the step size A. When the difference ig Treater than A, X(1) and X(1) can no longer follow X(t). This type of srerload is not determined by the amplitude of the message signal X(t) but ever by its slope as illustrated in*Figure 10.25; hence, the name slope overload. c : ‘To derive a condition for preventing slope overload in DM systems, let us assume that X(t) = A cos(2zf,t). Then, the maximum signal slope is ax()] _ XO) = Ada The maximum sample to sample change in the value of X(t) then is A2af,T'. slope overload, this change has to be less than A, that is, 2af.TiA © as f 0, and the integral of Gi(f) over ding f = 0 is infinite. Fortunately, baseband filte range of frequencies inclu have a low-frequency cutoff f > 0; further, f; is usually very small compar to the high-frequency cutoff f,. Hence Binoy =2 f : Ga(f) af -Pf[1 1] w 1 Je, _ 2M Pf : whi (10 since f, filter Shi distributor 7 Xy(e) — Xy(t) (a) Figure 10.30 Baseband filtering of a TDM waveform. it has no direct relation to the original messages, passes through the correct sample values at sampling times. X,(t) is obtained by lowpass filtering of the interleaved sample sequence. At the receiver, X,(t) is sampled and the sample values are distributed to appropriate channels by the distril sampling frequency is close to the Nyquist rate, that is, f,~ fz, then the bandwidth of the filtered TDM waveform is Mf, Hz, which is the same as the bandwidth of the FDM waveform. 10.5.3 Asynchronous TDM In the preceding discussion of TDM systems we had assumed that the signals being multiplexed have comparable bandwidths and hence the sampling ne for each signal is the same. However, in many applications the signals to be ‘time-division multiplexed have different bandwidths and hence they baal be sampled at different sampling frequencies. In these situations, we eh multiplex these signals using the technique described previously, ™ employs a common clock rate for all the channels. «need One method of combining a group of asynchronously sampled time multiplexed signals uses elastic store and pulse stuffing. An. elastic s vision storageTime- Divigi, Division Muttipie xing . ssential for multiplexin, iol is in such a manner that the it ttonous si rn seo the rate at which it was read a at order. Data can be recorded ont (his BR adjusting the tape speed during rep mt ae puffer into which data can be read ing . One ¢ the ta ample of such a lay Tead o sean ther exay uit at a one rat, Mple is a rate f elasti © and read out a on the use 7 a store and pulse stuffing t Vi and tr ‘0 the earth. S of a nu rime is j + Let us su number Oe s each lasting @ ent ea of one second are et that three ey and that their signals are sampled and stored in thre tformed simul- eo pevices- Let us assume that the tinec signal: € Separate digital : 7 s are sai gor 000 samples per second, respectively, and the wltine rates 2000, are encoded an d: sits obit PCM codewords. At the end of each 1-sec ii ht for one second during which time ai of me eae experiments caste ie oton each storage devi dao ol ring trans! . levice can be emptied ame rate (5000 samples per second), ynehistily tect . ultiplexed and a single TDM signal can be transmitted to earth. There is i major problem associated with this procedure. The first 2000 words of each signal can be multiplexed without any trouble. During the multiplexing of the next 2000 words there is no contribution from the first signal, and during the ist 1000 words there is no contribution from the first and second signals. However, because of noise, the receiver will be reading words when no words d 2 are being ‘transmitted. To avoid this erroneous slots corresponding to signals that filled with dummy sequences of bits. These ded so that the receiver se stuffing since missing from channels 1 an inerpretation of noise as signal, the time have already terminated are dummy sequences are carefully chosen and enco nique is called pul 10 spaces provided for the recognizes them without difficulty. This tecl itrequires that digits or pulses be stuffed int message bits. (See Figure 10.31.) Frame 1 CER: Dummy data (stuffing! tina ul Figure 10.31 example of Puls?