. . Karplus CHAPTER VII SAMPLING AND QUANTIZING - THEORETICAL FOUNDATIONS 7.1 The Basic Problem In true hybrid computer systems, analog hardware is connected in a closed loop with digital hardware. At some point in the system it is there- fore necessary to translate data in analog form into a form compatible with the requirements of the digital circuitry; and at another point in the system, digital information must be converted into analog form. Data in the analog portion of the hybrid system are processed in the form of continuous DC voltages, the magnitudes of the voltages at any instant of time corresponding to the magnitudes of the dependent variables. In the digital portion of the hybrid loop, the dependent variable as well as the independent variable must be discretized, so that data are represented only at discretely spaced instants of time and are permitted to assume only one of a limited number : of amplitude levels as determined by the word-length. The basic steps involved in translating data from analog form to digital form and then back to analog form are illustrated in Figure 7.1 and include the following: ' 4 1. Sampling: the continuous analog variable is measured at discretely ee spaced intervals of time, and attention is limited to the magnitude of the data at those instants, Information as to excursions of the dependent variable between successive sampling instants is discarded or lost. 2. Quantizing and coding: the amplitude of each sample, usually an electrical voltage, is translated into a binary code using a specified number of significant figures. The variable is thereby forced to assume one of a mited number of levels or quanta, Information regarding the precise location of the dependent variable between quantization intervals is discarded or lost 3. Digital processing: the sampled, quantized, and coded data are manipulated in the digital portion of the hybrid system in accordance with a ai prescribed program,4, Decoding: the sampled, quantized, coded data are translated —_———___ into voltage form, such that each binary number is represented by a voltage pulse with an amplitude corresponding to that number. 5. Holding and filtering: the series of pulses is converted into a continuous analog sign@l by means of an extrapolating device which uses the signal velues at the sampling instants to construct a continuous voltage signal. In order to focus attention on the problems associated with the inter- faces between the analog and the digital portions of hybrid systems, it is assumed in the subsequent discussion that the digital processor does not affect the data in any way. This unit can be considered, for example, to be merely a digital memory which reads out a sequence of binary numbers identical to those previously read-in. The analog voltage output in Figure 7.1 should then be identical, except for a shift in time, to the analog voliage input, and any difference between these two continuous voltages can be ascribed to errors introduced by the conversion equipment. Clearly the operations of sampling and quantizing involve the loss or discarding of infor- mation inherent in the analog input and can be expected to contribute errors. Departures from the "ideal" in the decoding, the holding and the filtering units can be expected to compound these errors. _Instinctively, one would be tempted to expect that these errors could be reduced by reducing the length Of the sampling interval so that more samples are employed to represent the continuous signal, and by reducing the quantization interval so that more significant figures are employed in the digital representation. snaoe ment ccovensron niorratnatee comenszon — —~——— a vasa [RS] se [GE] me vanes || mos {gine pseoine es | Be so ects] Some [ray at tee FIGURE 7.1 OGUnfortunately, the present state of knowledge of the theoretical foundations underlying sampling and quantizing, does not permit the rigorous 1 and digital~analog conver: derivation of the errors inherent in analog-di sion. Nonetheless, a consideration of some of these theoretical concepts provides an insight into the role and relative importance of certain hybrid- system design-parameters in assuring accurate closed-loop operation. In this chapter, errors introduced by sampling are first considered. This is followed by a consideration of the specification of equipment to permit accurate desampling or conversion into analog form. In Section 7.4 the errors introduced by quantizing are briefly considered, while Section 7.5 is devoted to the problem of holding. More detailed information regarding these topics is to be found in References 1, 2,3. 7.2 Sampling The purpose of the sampling unit in the hybrid computer loop is to measure the values of the dependent variables at discretely spaced intervals of time, These data are then coded or translated into binary form and constitute the information fed into the digital portion of the hybrid system. In its most basic form, the sampling mechanism can be viewed as a simple switch, as shown in Figure 7.2. This switch is closed periodically at sampling instants T, 2T,...nT, where T is the sampling interval; the switch remains closed for a short interval of time y. A continuous analog signal applied to the input of the sampler therefore results in a series of approximately rectangular pulses whose amplitudes correspond to the input Seer where the analog signal controls the amplitude of a pulse train p(t). The pulse width 7 should be sufficiently small so that a negligible change in the analog signal takes place while the switch is closed. Thus, if the input to the sampler of Figure 7.2 is an analog signal f(t), the output is described by the relation Sampled fen Lee) eS rt) =D stat) [ wen) - utt-nt-n)] SE (.) n=0 Se 1-3| cournmons swruso | ‘evr | Saneiae our FIGURE 7.2 where u(t) is the unit step function and the expression in square brackets represents the pulse train p(t). In order to simplify the mathematical treatment of the sampled signals it is possible to consider the sampler as modulating the magnitude or area of a series of impulses (Dirac delta functions). With this assumption this equa- tion becomes rt) = J) fla) 6(t-nT) (7,2) n=0 a Well-developed mathematical techniques, including the z-transform can be applied to the analysis of impulse-modulated systems. In hybrid computing systems of the type discussed in this text, the sampling switch is activated periodically, resulting in uniformly spaced samples. This facilitates the synchronizing of the conversion operations with the operation of the digital computer and obviates a number of other difficulties. Jn general, however, it is not necessary that the sampling instants be equally, spaced. Infact, errors associated with sampling can be greatly reduced by, permitting the data being sampled to control the width of the sampling interval.Since nonperiodic, signal-dependent sampling is a nonlinear operation, the analysis of such systems is extremely difficult and this approach has not found application in hybrid computers thus far. The discussion of errors inherent in the sampling process is best — accomplished using Fourier transform techniques. The Fourier transform Fw) of a function of time f(t) is defined by the relation a Pw) =F ry eB tae Fons Ginefan (1.3) Equation (7.3) can be used to transform functions of time (which meet certain conditions) to functions of frequency, The inverse transformation is accom- plished by the relation _ 7 “bo Farin brmtrrc, wa jot isch. eaten ty = S Pla) ®t dw (7.4) In general, Fw) isa lex number, The magnitude of F(w), denoted by [F@)| is called the Fourier spectrum of it). Strictly speaking, periodic functions of time (such as cos wt) do not meet the conditions for Fourier transformation and their transforms cannot be obtained by direct application of Equation (7.3). However, note that the inverse Fourier transform of a pair of impulses in the frequency domain [sw+0,) +6 -0,} can be obtained Iatrre from Equation (7.4) as follows DZ fae ic jot 5 f t(t) = 5 S [su+a,)+6-v,)] eh dw Gor oN 1cre ap 1 cre om Top va) sete ret ae +3 5(v-w de? * aw jet ~jwt. a of) ot) 2 tg le) +e (ce ?) =F osu) (7.5) Consequently, the pair of impulses can be defined as the Fourier transform of cos wt. Ina similar way it is possible to obtain the time functions which correspond to periodic spectra, Several important spectra are defined in Table 7,1, Note that, as indicated above, continuous periodic functions of 1-5TABLE 7.1 £(t) ft A cosine waves u ys FP + A impulse constant $0 f F , 1 1 1 Yq vosine series of wae OSD Expulsce 7 2 @ 1 ft FP fa Me setnge \ nee . ane fie 7] 7time possess impulsive spectra while impulse functions of time have periodic spectra. Consider now the frequency domain characteristics of the sampled f signal indicated in Figure 7.3, Assume that the signal f(t) has a Fourier Assume that the signal f(t) hae a Fourie: \ 4 transform F(u). The output of the sampler is given by cenelormee ap eet ple) Samy ort pnt— (7.8) where the pulse train p(t) consists of a series of unit amplitude rectangular pulses of narrow width. Since p(t) is a periodic function with period T, it can be expanded in a Fourier series as py ©, el27Ht/T (Fonts Sets ) (7.7) a k where the C, are constant coefficients, Substituting in (7.6) the sampled function can be written as +20 jkw t L w=5 c,eurye* { sempbra fou (7.8) ig the sampling frequency. The "shifting theorem" of Fourier 7 ere w.= 2% omar, transforms now yields the Fourier transform of f*(t): s« (ina yd. Ne He Yidbowdg Equation (7.9) shows the effect of sampling in the frequency domain. too P%s)= J) C, Fw +kw,) ieee It will be recalled that the spectrum of f(t) is given by F(w). Therefore, the effect “OF-ampling is to generate an i number of sidebands separated by integral multiples of the sampling frequency and proportional in amplitude tothe input spectrum. This spectrum is illustrated in Figure 7.4. The side- bands generated by sampling are termed the "complementary signal" while ree ean een eee eee eee the central part, which is proportional to the Fourier transform of f(t) is called the "pure signal". The coefficients C, depend on the pulse width y 7 and form a decreasing sequence, so that the magnitude of the complementaryOC) at 7 COMPLEMENTARY SIGNAL COMPLEMENTARY SIGNAL FIGURE 7.3 COMPLUMMNTARY SIGNAL FURY SIGNAL COMPLEMENTARY SIGNAL FIGURE 7.4oO approaches infinity, If the impulse approximation is used to describe the sampling process, it can be shown that ard so that signals decreases as all the complementary signals are equal in magnitude, as illustrated in Figure 7.3. It is apparent, therefore, that the sampling process introduces unwanted frequencies. That is, the signal emerging from the sampler hae a frequency spectrum which extends to infinity, even when the analog input signal is band-limited, These additional frequencies represent errors which must subsequently be eliminated. ‘Thus, the sampling process gives rise to two major classes of errors: eid oat ad ee ee Errors due to the discarding or loss of information between sam- pring Kevvais, CFE” of Inge @ Errors due to frequency components added by the sampling process ‘The first category of errors can be controlled and eliminated by the correct ———EOAEAS—SErrre application of the sampling theorem described immediately below. | The elimina- tion of the second class or errors requires the inclusion of a suitable filter in the desampling or data reconstruction portion of the hybrid ayatem. ae Oe. Ooo ‘The fundamental theorem underlying this theory of sampling was first This theorem may be phrased in a number of different ways, each emphasizing a different implication, With reference to the above development, in which a function f(t) with a Fourier spectrum F(w) is impulse-sampled at a sampling rate 1/T, Shannon's theorem implies each of the following: 1, If Flw)=0 for |w|2 7/T, then an "ideal" desampk completely recovering f(t) from the sa nction. As is demonstrated in the next section, an "ideal desampler is not physically realizable, however. Mir 2. te pw)=0 tor — mex’ | by specifying the values of the signal at instants of time separated by no more 20,4,’ 2 complete description of f(t) is obtained max’ than (7/w,, ,,.) seconds. MW) 3. The sampling frequency must be at least twice aa great as the fre- quency of the highest Fourier component Umax of f(t). eeEeE——~——ES~—EIErs vO 7-9fed “Ss Ws. If the sampling frequency is less than two times the frequency of the highest Fourier component u,.,, errors are introduced into the sampled signal. These errors cannot be removed by subsequent filtering or other manipulation. [V/\ 5. & desampting device, even if operating in an "ideal!" manner is unable to distinguish frequency components of f(t) which are © higher than one-half the sampling frequency from frequency components of f(t) which are “crower Hai GHE=TaIT the sampling frequency. The higher frequency compo- nents are therefore misinterpreted as lower frequency components and the desampled output is in error, An appreciation of these statements may be gained by considering two sinusoids, one having a frequency € above /T, and the other having a fre- quency € below r/T, where ¢ is a small number, Following a development by D. T. Ross,” it is easy to show that cos [(A+e) t+0]-cos[(Z-e) t-0] where @ is a phase angle, whenever time t is a multiple of the sampling period, This will occur whenever t=kT. Recognizing thet cos(x) = cos (-x) cos(x) = cos(x+ 2kn) the left hand side of Equation (7.4) becomes = = a cos | (5 +e) xr +e] = cos [:tk + ¢kT +6] = cos [-mk - €kT - 8] = cos (-1k - € kT - 6 + 2rk] = cos [7k - €kT - 6] : cos| (% se) er- | In other words, provided that the two sinusoids are sampled only at t=, 27, 3T..., the magnitudes of each of the samples are exactly identical. his is illustrated in Figure 7,5. Note that although the frequency of the QOO FIGURE 7.5 two cosine waves is considerably different, they have exactly the same magni- tudes at the sampling instants. In the desampling process the function with frequency (n/T’ + e) will be mistaken for the function of frequency (1/T - €). Tequency is 4 Fourier component of f(t), the shape of the es! Wi fer markedly from that of the original function. The Sampling Theorem effectiv mpliny rate, i.e. the frequency with which the switch in Figure 7.2 is closed. This frequency must be at least twice the highest frequency of interest. [But this By itself 1s not enough, | All frequency components of f(t) which are greater Than one-half the sampling frequency must be removed prior to they will introduce errors into the data, Accordingly, a sampler must always be preceded by a filter having a cut-oif frequency somewhat below one-half the sampling frequency. Occasionally the analog system generating f(t) may be counted upon to perform this filtering action automatically, {,e, the analog system may have a bandwidth smaller than one-half the sampling frequency. Tn general, however, a special filter is required 7.3 Desampling As demonstrated in Figures 7.3 and 7.4, the Fourier spectra of sam- pled functions contain all the frequencies of the unsampled functions as well as complementary components located outside the region |w|=*/T. The 7-11reconstruction of the continuous signal from the sampled signal then involves It is therefore necessary to provide a low-pass filter to eliminate all comple - mentary signals, while leaving the pure signals entirely unaffected. The desired transfer characteristics H(w) of this filter in the Fourier domain are shown in Figure 7.6. If the actual response characteristics differ ‘from the ideal, the filter either alters the pure signal or permits some of the comple- mentary frequency components to pass through. pga |H(«>)] |P*(eo)] “3 =r ° x am ws L T T T T 1 Cc ) ve FIGURE 7.6 Inverse Fourier transformation of H(v) yields the time domain expres- sion h(t) for the response of the filter to an impulse applied at time t = 0. First expressing H(w) mathematically HW) = T for jw] s ala = Ofer |ul> 2+ lwl> Applying Equation (7.4), nit) = f Ble) * aw (7.10) and substituting for Hl) -0/T a/T 0 + moll oars” re asl” ot ay . eo La/T LITor it r/T it a/T nit) “3 < 7 which becomes upon application of Euler's equation ves sink t snvelot® h(t) = —2 5 Cahir (7.11) t ais Since there is no restriction upon the time variable t in Equation (7.11), the ideal filter is required to respond to an impulse at time t = 0 by generating & nonzero output for negative time. That is, the filter is required to respond before the excitation is applied. This involves prediction, and such a filteris evidently not physically realizable. It is therefore necessary to employ filters having "nonideal" characteristics. It appears therefore that the desampling _Process invariably introduces errora or fails ta remove all the erroneous “signal components introduced in the sampling process. Practical approxima- tions to the required desampling unit are considered in Section 7.5. 74 Quantizing ‘The translation of an analog variable into digital code necessarily involves quantizing, such that the analog voltage at €ach sampling instant is _siutgned-one ora lialled number Gf levelo or quanta, The number of Tevela~ “af levels oF quanta. levels, available depends upon the number of significant figures in the binary number used to represent the analog voltage. This number of significant figures depends in turn upon the word length or register size of the digital equipment being used. Since the analog voltage can assume any value within the dynamic range of the analog equipment, usually from -100 volts to +100 volts, quanti- zation involves loss of information and can be expected to introduce errors into the hybrid loop. Although in some respects the process of quantizing ig similar to that of sampling, as described in Section 7.2, the description of errors incurred in quantizing and the specification of a maximum quantiza- tion interval is far more complex and has received far less constructive attention on the part of theoristsIn the absence of reli esti re introduced by quantizing, hybrid computing-system designers have generally chosen to specify word lengths such that the per-step round-off errors committed in the digital calculations are at least as small as errors expected in the analog portion of the hybrid system. Thus, if the analog computer and the analog circuits have a specified accuracy of 0.1% of full acale, at least 10 or 11 binary digits are carried in the digital equipment. Ina 10 bit word, the least significant digit corresponds to one part in 1, 024 of full scale. Since most digital computers have word length considerably in excess of this 10 or 11 bit figure, quantization problems can usually be ignored in designing hybrid systems. There are instances, however, in which a considerable economy in equipment can be effected by holding the word size to a minimum, In order to give the reader a qualitative insight into the ‘quantization problem, @ brief theoretical discussion, based upon a more detailed treatment by D. T. Ross” is presented below. ‘The action of the quantizer may be regarded as the changing of the continuous analog signal into a staircase function, as shown in Figure 7.7 for a very coarse quantizing process. Note that the quantized function £!(t) Lv is limited to five ordinate values corresponding to the midpoints of five A quantization intervals, q. Whenever f(t) lie within one of these intervals, the quantizer reports the ordinate value of the midpoint of the interval. In ‘' V/. a hybrid computer loop, the quantized f'(t) is eventually dequantized, resulting in a conventional analog signal, The quantization problem involves then the consideration of any differences between the original analog function f(t) and the analog function generated by quantizer-dequantizer action, just as the sampling problem involved the analysis of the difference between an input analog function and an output analog function, generated by sampling and desampling the input function. In dealing with sampled functions, it is found expedient to make extensive use of the Fourier transform domain. Errors introduced by the sampling process are characterized as functions of frequency, and the sampling theorem is likewise based on frequency concepts, To apply the concepts developed in the discussion of sampling, a variable capable of playing the same role as frequency must be defined for the quantization process.£"(t) alee ee 4 FIGURE 7.7 ‘The basis for quantization system analysis lies in statistical concepts. A statistical function termed the density di ion, w(t), is employed to characterize the analog signal, and attention is limited to the properties of this distribution function ae ene ee I u(t) is plotted against £, w(t) df is proportional to the fraction of the value of f(t) which lie in the interval ftof+Af. Figure 7.8 shows the density distribution function associated with a specific analog signal. Evidently an infinite variety of analog transients can have the same density distribution function. Unlike Fourier series and Fourier integral representa- tions, therefore,| density distributions do not constitute unique specification: ——— Of @ continuous variable Density distributions may change in time or they may be constant, as may be the case for example if the analog signal exhibits periodicity. eee In the latter case, the analog function is termed stationary, convenience, density distributions are usually normalized so that the area nder the w(t) u vs, fouvelseqalminiy, — —S=~CS ( TL dasribuhom) u(t) £ For — £(t) FIGURE 7.8If attention is limited to probability density distributions, the time variable t in sampling becomes analogous to the ordinate variable f in a quantizing. To develop an integral transform analogous to the Fourier ennai 5 transform, a new variable @ is defined to correspond or be analogous to the frequency variable w in sampling. The transform of the probability density distribution is termed the characteristic function and is defined by we) “ff we) 3% at (7.12) The inverse operation becomes a w= f wee) el aa (7.13) Note the similarity of Equations (7. 12) and (7.13) and Equations (7.3) and (7.4). Utilizing these concepts, a quantizing theorem analogous to the sam- c ) pling theorem can be formulated. Letting i(t) = continuous analog signal f(t) = quantized analog signal w(t) = density distribution of £(t) w! (f) = density distribution of £1 (t) Wa) = transform of w(f) q* quantization interval the Quantization Theorem can be stated as WW 1. 1 we) = o for Ja] 2 #/q, then wit) can be completely recovered from wi (f). That is, provided the quantization interval q is small enough, errors introduced by the quantizing process can subsequently be compensated or eliminated. mW 2. If Wie) #0, for je] 2 7/q, erroneous signals will be introduced into the data, signals which cannot be removed by subsequent filtering or ( compensation.- W/W 3. ‘The dequantizing device cannot tell the difference between signals > for which @ is 3/q + € and signals for which a is 1/q- €, where € is a small number. These two signals will therefore be confused. Note that this theorem applies only to the density distribution and not to the function itself, Whereas the Sampling Theorem guarantees that a continuous function can be Feconstructed from a sampled function, provided only that the sampling frequency is sufficiently high, the Quantization Theorem merely states that a continuous function having same density distribution function as the original function can be recovered from a quantized function, provided that the quantization interval is sufficiently small. ‘The parallel between quantizing and sampling can be carried a step further. The sampling process described in Section 7.2 can be regarded as a modulation phenomenon, in which the analog signal f(t) modulates a series of rectangular pulses. Quantizer action can be regarded in a similar manner. Consider the effect of the quantizer shown in Figure 7.7 upon the density distribution function w(f). Since the values of the quantized signal are zero for all ordinate values except x = 0, +-q, + -2q,..., the density distribution function w! (f) of the quantized signal is likewise zero except at the quantization points, A plot of w'(f) vs. f therefore has the form of a series of impulses \ occurring at f= +-ng. The area se is equal to the are! r fie the curve w(f) from nq - Sto nq +4. This is illustrated in Figure 7.9. From 4. 4 this point of view, quantizing involves actually a sampling of the density distri- fi bution, The nature of these samples are different from those encountered in Section 7.2, however. In the sampling process illustrated in Figure 7.2, the data at the sampling instants are represented by the amplitudes of the pulses. In quantizing, on the other hand, it is the x the pulses which epee Tepresents the continuous data. It can be demonstrated, however, that the fy", + eee 2 area sampling illustrated in Figure 7.9 can be regarded as ordinary sam- ie pling of a related function, More specifically, w'(f) may be obtained from w(f) by first convolving w(f) with a rectangle having a unity height and a width, q and then subjecting the resulting function to ordinary sampliFIGURE 7.8 that is atn = 0, #1, £2, #8,... . Denoting this new function as W(f) and letting y be a dummy variable, wi(f) = qW(f) = af w(y)z(f-y)dy sampled at nq (7.14) where the function 2(f) is defined as = 4 2(f) =| 0 at t< -2 q att>a oatr>d _The function 2{f) ia known ae a fuzz-funetion for the following reason. Suppose as part of the data reconstruction mechanis} an instrument is random as specified by the distribution w(f). Assume now that this device some value between f -q/2 and f +g/2 is generated with equal probability. In essence the desampler can be considered to be adding @ random error to the b — Or ere Sriginal clan 7 ‘Original signal making it "fuzzy" This fuzzy signal corresponds to ) H(t) as { Gefined by Equation (7.14), as is iMlustrated in Figure ure 7.10, | vero igure 7 ya It appears therefore that the quantizing process can be considered Fay fw as a combined sampling and fuzzing operation, An a domain analysis of the quantized signal then demands the calculation of the a transforms of z(f) 7-18¥— q2 (29-2) hk w(t) FIGURE 7.10 as well as of w(t). Applying Equation (7.12) to 2(f) yields Zila) = (7.15) o The spect tion ig found by multiplying Z(a) by (a). Accordis 2a) Wa) = Wla) (7,16) Thia multiplication is illustrated in Figure 7,11, Note that the fuzzy function e has a somewhat narrower a spectrum than the original function. al wl og 2 st qa old FIGURE 7.11 7-19‘The sampling process associated with quantizing results in the forma- tion of a periodic spectrum. That is, the spectrui is repeated at intervals spaced 2n/q in the a domain. The repeated spectrum —of the fuzzy function Ts gare 7.12. The phenomenon illustrated in way the fuzny Fanction 16 showece Freure Figure 7.12 may be considered analogous to that illustrated in Figure 7.3 for e” ordinary sampling FIGURE 7.12 The recovery of continuous data from quantized data necessarily involves two steps{VFIFst, 1 ce {o-eliminate the complementar. signals by multiplying the quantized signal by a function of a which ig zero everywhere except In This corresponds then to the filtering 0 fd function using an ideal titer HW) is then necessary to divide the resulting signal by Z(a) as defined by Equation (7.15), In the final atep of dequantizing, the density distribution w(f) of the continuous time function {(t) is obtained from W(a) by inverse transformation in accordance wit the theoretical point of view, the statistics of a quantized signal can be recov- ered, provided only that the quantizing theorem is satisfied. The electronic implementation of these analytical techniques is another matter. 7.5 Filtering 7 ‘The conversion of digital information into analog form involves two tasksVa. ‘The translation of the numbers in binary code, occurring at discretely-spaced intervals of time, into voltage pulses, such that the amplitude of each pulse is proportional to the corresponding number. [hh 2. The smoothing or the filling-in of data between sampling instants. ‘The former task is handled by the digital-analog converter and presents few theoretical problems. The second task, on the other hand, requires con- siderable mathematical insight. It is the function of the smoothing or filtering device to eliminate or compensate all erroneous signals introduced into the data in the analog- digital conversion process. It is therefore necessary to remove the comple- mentary signals generated in the sampling as well as in the quantizing operations. As yet no satisfactory electronic device has been developed for the removal of the complementary signals in the a domain. For this reason it is necessary to assume, in designing a hybrid system, that the quantization interval q is sufficiently small so that quantization errors are negligible, Attention can then be focused upon the removal of the complemen- tary signals in thew domain , mentary signals in the w domain, while leaving the desired or pure signal untouched, requires a filter having the gain versus frequency characteristics shown in Figure 7.13(a) and the impulse response shown in Figure 7. 13(b) Not only must there be no attenuation within the pass-band, but the signals fo be filtered should also undergo no phase shifts. As already pointed out, a response of the type shown in Figure 7, 13(b) implies a predictor mechanism one which produces a response before an excitation is applied. A filter of the type shown in Figure 7,13 is therefore not physically realizable, and it is necessary to seek approximations to the desired characteristics As described in more detail by Ragazzini, polynomial extrapolation methods can be applied to the derivation of physically realizable filter transfer functions. The derivation begins, nition that the Laplace transfer functions. The derivation begir transform characterization for a predictor eof producing an outpu 7-21|Flo)| _ ) }AO 0 /T a Pe = 20 time (a) (>) FIGURE 7.13 7 seconds before an input is applied, is F(e) = ® Laplace Trent form a pf (ap ' Peed cee € SS where s is the complex frequency parameter. Equation (7.17) can be rearranged to read ri) =[1-(1- -/T 7) (7.18) ‘The binomial theorem is now used to expand this expression, so that 2. -Ts -Ts 1 - die wf ). rear, (7.19) T ‘The greater the number of the terms of Equation (7.18) which are realized using the practical filter, the closer is the correspondence between that filfer and the "ideal" filter characterized in Figure 7.13. If an input transient r(t) is applied to such a filter, the Laplace transform of the output may be found by multiplying each term in Equation (7.19) by Ris), the Laplace transform of the input, The resulting expression can then be inverted to obtain the time domain response, which takes the form Hebe) = n(t) EET), x H(t) an(t-T)$ wt 27) (r+ r)n Be eo) v 7-22In sampling data systems, data are only available at discretely spaced incre- ments of time t = nT. Equation (7. 20) can therefore be re-expressed as ——— Veer) Vent) (T+ ae r(n'T +7) = r(nt) + 5 5 — we ——qewmr where the terms V "r(nT) are nth-order backward differences. For example, (7.21) the first-order difference is equal to the response at t=nT, minus the ny response at t=(n-1)T, V r(nT) = r(nT) - r(n-1)T (7. 22) Oe The second backward difference is found by taking the first backward difference at nT and at (n- 1)T and by taking the difference of these two differences, 9? raT)= 7 rat) - 9 ra YT HoT) -2in- yt | + [pin- arr - xin ayr | = r(nT) - 2 r(n-1)T + r(n-2)T (7. 23) O ‘The first term on the right-hand side of Equation (7.21) can be regarded as v° r(nT). Note that the higher the order of the difference, the greater the order difference no terms must be remembered; the term r(n - 1)T must be remembered for the first-order difference; and the terms r(n - 1)T and r(n- 2)T must be remembered for the second-order difference. Therefore the higher the order the difference expression employed, the greater are the memory requirements imposed upon the desampling system. On the other hand, the greater the number of differences employed, the more closely does the resulting filter correspond to the "ideal" Equation (7.21) is used as a classification for desampling filters. If only the first term on the right-hand side of Equation (7. 21) is realized, the “zero-order hold". If the first two terms on the right- etc. Since system is termed a hand side are realized, the device is called a "first-order hold" Ld “The mechanivations of higher-order holds are extremely complex, only zero- order and TIFst-order folds have found applications in hybrid computing systems 7-23By far the simplest and most widely used of these filter devices is the zero-order hold, often called the "data clamp". This device consists 1 “jnerely of an analog memory which holds or clamps an input voltage at a con- } ag shown in Figure 7.14. ‘In the hybrid computer application, of course, The 4 input function is not a continuous signal but rather a sequence of voltage it levels separated by the sampling interval T. The sample-hold device there~ J fore acts to maintain the voltage between sampling periods at the last received voltage level, In order fo evaluate the effectiveness of the zero-order hold circuit in eliminating complementary frequencies, it is useful to compare its impulse response and frequency-domain characteristics with those of the "ideal" filter of Figure 7.13. In Figure 7. 15(a) and (b) are shown respectively, the frequency response and the impulse response of a zero-order hold circuit. Note that the gain within the pass-bend is not uniform, but falls off monotoni- cally, Furthermore, the gain is not zero outside the pass-band, so that \ 4 certain complementary frequencies are permitted to pass through. In addition, b 7 \ signals within the pass-band undergo phase shifts proportional to -wT/2, which gg OT r(t) 5 yo adhe 4? sre 7? aT one introduces further errors into the solution, -at-T jor ar ar FIGURE 7.14ol x Tt mr } 9 time T T Ly (a) (b) FIGURE 7.15 ‘The next most complex filtering device is the first-order hold, which realizes the first two terms on the right-hand side of Equation (7.21), Recall Xe that the derivative of a curve f(x) is defined as A y/A x. The second term on AY the right-hand side of Equation (7 21) therefore represents the slope of the \ curve at t = nT while the first term represents the actual values. In place l x a of the staircase function generated by the zero-order hold, the first-order fa4 hold approximates the continuous function by a series of trapezoids. This is illustrated in Figure 7.16, Evidently a closer approximation to the contin- Evidently a closer approximation to the contin—_ b uous curve is obtained in this manner. In particular, if a given function is va actually a sequence of straight line segments, not necessarily horizontal with break points at the sampling instants, the first-order hold is able to represent the function with no error whatsoever. On the other hand, it is necessary to provide electronic circuitry to memorize the value of the func- tion r(t) at Mme t = (n-1)T,_In Figure 7.17(a) and (b) the frequency response and impulse response respectively of the first-order circuit are illustrated. Note that within the pass-band, the frequency response overshoots the speci- fied value by approximately 50%, resulting in excess amplification of some of the pure signal components. At the same time a certain amount of comple- mentary frequencies are not eliminated, Undesirable phase shifts also occur within the pass-band.r(t) x(t) St on or 7 -2T T ‘T 2T 3T FIGURE 7.16 rata 3A 5 6 F FIGURE 7.17 1-26It has been found practical on occasion to employ partial rather than full tiret-order Fold systema. That ise Second term in Equation (7.21) is multiplied by a constant in the range of 0.3 to 0.5. The resulting response curve is then a compromise" between that of Figure 7. 15(a) and that of Figure 7, 16(a), giving more or less flat response characteristics in the pass- band, ‘The mechanization of zero-order, first-order and higher-order systems using conventional analog computer ha cussed in Section 8.5 1-27REFERENCES Ragazzini, J.R. and’G. F. Franklin, "Sampled-Data Control Systems", MeGraw-Hill Book Co., Inc., New York, 1958. Susskind, A.K. (Editor), ''Notes on Analog-Digital Conversion Techniques", The Technology Press, Massachusetts Institute of Technology, 1957. Also published by John Wiley & Sons, New York, 1958. Jury, E.I., "Sampled-Data Control Systems", John Wiley & Sons, Inc., New York, 1958. 7-28