— \Y Sampling Theory 8.1 INTRODUCTION Most of the signals that are encountered in nature are continuous (analog). Analog signal processing, representation, transmission and recovery fall under the category of analog communications which have certain drawbacks. In digital communications, which is mor advantageous, it is required to transform an analog (continuous-time) signal into a discrete-time signal. This conversion process is called sampling. ‘Sampling is the first process that the analog signal encounters prior to its digital conversion A sufficient number of samples of the signal has to be taken so that the original signal is represented in its samples completely. Also, it should be possible to recover or reconstruct the original signal completely from its samples. The sampling theorem is certainly one of the most useful theorems since it applies to digit communication systems, which are proliferating constantly. The sampling theorem is another application of an orthogonal series expansion. In many applications, it is useful to representa signal in terms of sample values taken at appropriately spaced intervals. Such sampled-data systems find application in feedback control, digital computers, and pulse-modulstl communication systems. 8.2 SAMPLING THEOREM Sampling theorem gives a complete idea about the sampling of signals which is the fundamen operation in signal processing. Different types of samples like ideal samples, natural samples flat-top samples are to be taken into consideration. A sufficient number of samples of th #1 must be taken so as to represent the original signal in its samples completely. Also, it should Possible to recover or reconstruct the original signal completely from its samples. THe "™" of samples to be taken depends on the maximum signal frequency present in the signal. 340Sampling Theory * 941 Sampling Theory +341 pling theorent is stated in two parts: co SAUNT, rhe 8 contains no frequene ginal (A con Apa sil Y Components for tre npletely described by instantay conipletely Ke ‘quencies above f,, Hz, then it NOUS Sample Values uniformly spaced in time with period rel hes, lencies above fy, Hz, the signal rom the knowledge Of its samples taken at the rate of 2fy Samples per second, The first part represents the signal represent, vite {0 Fepresent a continuous-time ; oe frequency or Nyquist rate, an second! part of the coples gives the samples. i Combining both the parts, the Sampling theorem May be stated a “A continuous-time signal may be Completely repr ithe sampling frequency f; > 2p, Here ‘ented in its samples Jr's the sampling frequency and fqueney present in the signal.” tion in its samples ignal into samp and minimum sampling rate cs. The frequency 2f, is called the theorem represents pling rate Tequires Nuetion of the original signal from its Slactory reconstruction of the signal from its and recovered back Jn is the maximum Proof! To prove the lied 00 f4, Mz can be arate f,> Yf,, He, Let us consider a that the Sampling theorem, we sl hall show that reconstructed, exactly wi 4 signal whose spectrum is band: ithout any ero + From its samples taken uniformly N analog signal (0) whose spectrum is ban Signal x(0 has no frequen Therefore, Xf) = 0 for MAL > Sa Figure 8.1(a) shows this an LeUNCA) represent Let 8.0) be ani * impulses repeating ©) shows this impulse Figure 81(@), Wf. Figure 8,1( ‘ampled signal y(¢) as shown in 10 =a) 6,9) (8.2) AS the impulse train 57,0 is a periodic Signal with period Ds it may be expressed as a ‘OUrier series aS under: On q U4 2 cos 4 +208 2041+ 2.cos 3040-4. z i (8.3)342 + Signals and Systems 2a =F = 2h, where : A a(t) 4X(f) A DL LIN oo t Sh OEE Sige ir, (a) Analog signal (b) Spectrum of analog signal 5r,() , + t a) of a har Slo) tultiplier ~— F-—» yi 0 t (©) Impulse train (@) Multiplier xO re A 0 (©) Sampled signal (f) Spectrum of sampled signal Figure 8.1 Fourier transform series. ; i ; 4 Therefore, = [xO +2x(0) cos @t + 2x(t) cos 2@,t ++], To obtain the Fourier transform Y(/) of y(t), take the Fourier traisform of right hand si of Eq. (8.4). Fourier transform of x(t) is X(f). Fourier transform of 2x(t) cos @,t is [Xf -f) + Xf +f] Fourier transform of 2x(1) cos at is [X(f - 2f,) + X(f+ 2f)] and so on. 1 Therefore VDF IMD XS -§)4XE + H+ XS =26) + XE +A—_Sanping Theory + 343 X XU=nf) oe Foe (8.6) mn it is evident ths x5) and (8.6), i is evident that the spectrum Y(f) consi eat pot EO With period. f= 177, Wa. a8 shown in Figure 30. = 80 reeling | at Fam w(0. XP) should be recovered from ¥(/), This is possible if there joreon Figure 8.1(0) shows that this requires f, > 2, (8.1) ence im Ge x as the sampling frequency f, is greater than wi i fe, 8 ong as Js 18 greater than twice the maximum signal ae f, (signal bandwidth, f,,), Y(f) contains non-overlapping repetitions of X(). If this fa igre 8.1(0 show that x(?) Can be recovered from its samples )() by passing the sampled ver y() through an ideal low-pass filler of bandwidth fj, Hz. ‘This proves the sampling theorem. Tre following points are noteworthy: + From the spectrum of sampled signal, it is clear that as Jong as the signal is sampled at arate f, 2 2fy, the spectrum ¥(f) repeats periodically without overlapping. Thus, the original spectrum X(f) can be recovered easily. The spectrum of sampled signal extends up to infinity and the ideal bandwidth of sampled signal is infinite. The original or desired spectrum X(f) is centred f= 0 and has a bandwidth or maximum frequency of f, The desired spectrum may be recovered by passing the sampled signal with spectrum ¥(f) through an LPF with cut-off frequency f,,. This original spectrum X(f) can now be converted into a time-domain signal x(t). IEF, < 2fy, the successive cycles of the sampled spectrum overlap with each other and hence the original spectrum cannot be recovered and distortion occurs. Thus, for Feconstruction without distortion, the condition f, 2 2f,, must be satisfied. ‘3 NYQUIST RATE AND NYQUIST INTERVAL te sampling rate f, becomes exactly equal to 2fq samples per second, then iti called the Sctt also called the minimum sampling rate. It is given by f the maximum sampling interval is called the Nyquist in |. When ao the sampled eaage Me COntinuous-time band-limited signal is sampled at Nyquiserate { ) contains non-overlapping successive cycles of Y() repeating pe terval. Itis given by344 + Signals and Systems But they touch each other as shown in Figure 8.2. Therefore, the original spe, 4 be recovered from the sampled spectrum by using an LPF with a sharp cutofp ial ex Sn He. Ty op a) (VNIVVIVE\ BAD Og LS, % 7 Figure 8.2 Sampled spectrum at Nyquist rate. 8.4 RECONSTRUCTION OF SIGNAL It is possible to reconstruct the signal x(1) from the sequence of samples. The Process of reconstructing a continuous-time signal from its samples is called interpolation, A signal x(1) band-limited to f,, Hz can be reconstructed (interpolated) completely from ity | samples when the sampled signal is passed through an ideal LPF of cut-off frequency fH, t The sampled signal is expressed as vt) = x) 57,0) (89) ie., MO=F [a + 2xte0s 1 + 2x(1) cos 20,1 +++] (8.10) From Eqs. (8.9) and (8.10), it can be observed that the sampled signal contains a component (0 - WT,, To recover the original signal x(1) or the original spectrum X(f), the sampled signal is passed through an ideal LPF of bandwidth f,, Hz and gain 7, The reconstruction or interpolation filter transfer function is expressed as . i H(f) = T, rect (£) Bl) Applying inverse Fourier transform to H(f) gives the impulse response f(t) of the LPF, ie. hi) = Pacer lr REE (4) aati 2, im Hi) = By T, sine (Cgt) 2) If the sampling is done at Nyquist rate, then 1 T= be = yl, =1 meme Nov, 0 A(t) = sine (pt) gH {Samping Theory + 345 hows the graph of A(Z). It is observed th; at h(a) = 0 at all Nyquist sampling Figure 8.3. Reconstruction of signal, Xow: when the sampled signal »(Q is applicd atthe input of th ( is filter, the i + sample in y(t), being an impulse, produces a when a Pulse of height equal éacl -qual to the strengt! re sample. Adding the sine pulses produced by all samples gives (1). For instance the " sole of the input y(0) is the impulse x(AT,) 6(t ~ kT). The filter ‘output of the impulse will be pate - kT) 7 2A) eon, the filler output t0 0) is (1) and is expressed as a sum given by MO) = LK) hk) x (8.15) = Exar, )sine [@,,(t ~ kT, )] (8.16) x)= Y x(kT,) sine (@,, - kx) (8.17) Equi (8.17) is called the interpolation formula, It provides values of x(t) between the samples a a weighted sum of all the samples. 85 EFFECTS OF UNDER SAMPLING—ALIASING When a continuous-time bs Ay, other as «and-limited signal is sampled at a rate lower than the Nyquist rate, then successive cycles of the spectrum ¥(f) of the sampled signal y(t) overlap with each shown in Figure 8.4, Figure 84 Spectrum for sampled signal at fy < 2aSignals and Systoms Hence, the signal is under sampled when f, < 2f, and some In fact, aliasing is the phenomenon in which a hi ignal takes identity of a lower fre AMount of i Alias h frequency Componeng ; 8's Prog uency component in ne Spectr Seg Spectrum ause of the over! al x(t) from the the over! signal, From Figure 8.4, it is clear that by possible to recover the original sig Because the spectral components in As the information signal cont frequency is 3 5 Hp duc 0 aliasing ph “me mpled signal y¢ ap regions add and hen ns a large number of frequency Momenon sit 1) by low.nage me ee, the signay jie : WS di a eS, decid: it 8 problem. So, the signal is fist paced through an Lop nee sang all the frequencies above fy, Hz. This process ie Called band- Limiting o sin EEF bl | LPF is -alias filter of the origi as itis used t0 prevent aliasing effec, Aiet basa mpling frequency since the im becomes easy to decide the san ‘ANd-limiting : aximum frequency ig fre ig In short, to avoid atiesing 8 Fred aj i * Pre-alias filter must be used to kin Sampling frequency *," 8.6 SAMPLING OF BAND-PASS SIGNALS In the previous section given signal is The ns, sampling theorem for low- 4 bandpass signal, then a different he sampling theorem for b The bandpass signal x() whose m ito and recovered from its samples if Here f,, is the m; and hence the minin ass sigmals is discussed. However, When he criteria must be used to sam andpass signals may be expt narimum bandwidth is 1 is sampled at the minimum rate ¢ aximum frequency component mum sampling rate for bandpass signals must be ure 8.5 shows the spectrum of an arbitrary by ntred around frequen bandpass signal completely represenes Pf twice the bandwidik” and the bandwidthis 4, Mn Samples per second ail. The spectrum in Figure 85 Thus, the frequencies present in he Present in the signal nandpass sign: Je. The bandwidth is 2p, are from fi ~ fy tO f+ fry xy) ote Ce i fags f i 2, — i 4%, Figure 85 Spectrum of an arbitrary bandpass signal. The centre frequency f, > f,. : ude This bandpass signal (0) is first represented in terms of its inphase and 4 components x1) and xp(t) respectively. oi) ie. XU) = x40) COs @.t ~ xolt) sin atSampling Theory + 347 a quadrature components are obtained by mut i essing the sum fi nd then suppressing t requencies by ws gets) a) Cot only ow ne LPF. Ths, the ips ad se Hens i Hirited between fn, 10 +f, a shown in Pecarponen. The spectrum of ee x x(t) by cos wt and Sn 0 thm f Figure 8.6 Spectrum of x(t) and xg(t) components, simplifying and applying some mathematical manipulations to Eq. (8.18), the reconstruction ta is obtained as under: < n . a= = est} si (2t0-2} |e (1-32-] An ia with that of the low-pass signals given in the interpolation ced by x(/4f,). Jed version of bandpass signal and 1 yim taken, then the bandpass signal of bandwidth 2f, can be formu Conaring this reconstruction formul formula, we observe that x(f) 18 re] Here, lM.) = x(nT,), the samp! T, Thus, if 4/, samples per second are completely recovered from its samples. Hence, for bandpass signals of bandwidth 2fm. thu of the bandwidth, te f,=2x Bandwidth = 4f,, samples per second. the minimum sampling rate is equal to twice 47 SAMPLING TECHNIQUES | Sep : . a of a signal is done in several ways. Basically there are three types of sampling , ‘ ; (ssntneous sampling or Impulse Sampling oF Heal Sampling < fuusl Sampling x(t) «) x) — ' 0 0 f a) cy) Figure 8.10 Natural sampling. The sampling is termed natural as the profile of each sample retains the shape of segment during the pulse interval. eg © For natural sampling of a band-limited signal x(2), the signal is sampled intervals at the rate of 2f,. where f,, is the maximum frequency of x(0- if © The signal (1) can be reconstructed from these samples with no distortion by P the received samples through an LPF. = 4- Sampling Theory» 351 rd 300 to 34 KHz voice channel ‘Js which corresponds to the natural 1/8000 or 125 ps ag stat * sqao sample aren overt ISON OF VARIOUS SAMPLING METHODS COMPAR 1 comparison of various sampling methods, » the standard sampling rate is sampling. It means that a sample is rag sus te pte ‘Table 8.1 Various Sampling Methods Meal sampling Natural sampling Flat-top sampling Ituses multiplication by Tt uses choppin aE an impulse function 1B Process It uses sample and hold circuit rest Practically not possible Used practically Most popularly used Practical method sampling rate Tends to infinity Nyquist rate Hywila cle ie interference __ Nominal Maximum Minimum Bi i IMPORTANT FORMULAE » Jn is the maximum frequency component of the signal. Yuu }, For reconstruction of the signal from its samples without distortion, 1 Maquist rate, fx 1 Niquist interval, Ty 1 2 Yn TS Sn fi or zh 4. The low pass filter (LPF) used to recover original signal from its samples is also called interpolation filter. The process of reconstructing a continuous-time signal from its samples is called interpolation. Interpolation formula: a(t) = x(AT,) sinc (@,, — kit) k 5 Alsing occurs when the signal is under sampled, ie. when f, < 2fy ‘Minimum sampling rate for bandpass signals = Twice of bandwidth = 4f,, samples/s. ie t Tk 4p “onstruction formuta for bandpass signals: ws ¥ {ge} (2 = name A4f, a cos. E (: - Fal * Speen . Wm of ideally sampled signal: Yin=s, E XU-mh)352_+ signals and Systems Spectrum of flat-top sampled sign: YN=HNL Y XF-kf) where, H(f) = T sine (ft) e?*", the spectrum of Tectangular pulse, 10. Spectrum of naturally sampled signal: vip=raf, ¥ sine (nf.t) X(f = nf.) 11. Equalizer is used to compensate aperture effect. Transfer function of equalizer is given by ke RSE TT - Fis a constant H(f) sine (fr) m 12. Impulse train O,()= Y St-nT) SOLVED PROBLEMS Problem 8.1 Calculate the Nyquist rate and N (a) x1) (b) at) = ‘yquist interval for the following signals: 3 cos S00z1 + 15 sin 20021 - 5 cos 100zt cos 400z1 + 6 cos 6401 (©) x0 = sin (1000 x1) mt a. @) a= (#24) at Solution (a) Given x(1) The given sign: cos 5001 + 15 sin 20021 - 5 cos 100nt al has the following frequencies: 500z1 = wt = 2nfit = fy = 250 Hz 2nft = fe = 100 Hz Inf => fy = 50 Hz component present in x(t) is Jn = fy = 250 Hz Maximum frequency Therefore, Nyquist rate, Su = Ym = 2 x 250 = 500 He ‘Therefore, Nyquist interval, 1 — =2 fy 500 ~“™*Sampling Theory _* 353 4oont + 6 cos 640z1 5 the following frequencies 4oont = Ot = 2nft = f 640nt = Oot = Wht = fy = 320 Hz cos ws frequency component present in x(t) is 7 Sn = fy = 320 He fr = Yn = 2% 320 = 640 Hz tod eal Ty a ¥* 7 B0 1.5625 ms. yguist 1 __ yas ia sin (1000 jg Gnen a0 = sane a mt ‘ie 10001 = cor = 2aft => f = 500 He You = 2 = 2x 500 = 1 KH Nyquist rate. «Nyquist interval fw ) Given at) = ( 7 _ 1 =cos 100m dat 100mt = r= 2aft = f= 50 Hz s Nyquist rate, fy = fn = f= 2x50 = 100 Hz cs Nyquist interval, ‘os 10mt. This signal i yy an impulse 7 Hz and 14 Hz. Draw the spectra of Hz and 14 Hz. [JNTU: Nov 2004) ren ie ie A signal x(¢) is given as x(f) = 7¢ s sampled b el pling frequencies ‘of the impulse train are ignal with sampling frequency of 7 Mein ™ Given x(2) = 7 cos 1Oxt X= Frey -9+5+9) pera 235 [6-948 49) (/) is given in Figure 8.11.F 354_+ Signals and Systems ~5 0 5 Figure 8.11 Problem 8.2. () When sampling frequency, f, = 7 Ha: Here fy, = 5 Hz, fy = 10 Hz Se 7 He Clearly. f, < Ym. i.e. overlapping occurs and aliasin, “The output signal consists of freque The spectra of the sampled signal is giv S(t) (ii) When f, = 14 He: Here f, = 5 Hz, 2f,, = 10 He f= 14 Hz Clearly, f, > 2f, and no over! apping between successive samples takes place. xX Original signal fw Shifted version of Xf) —— —_——————_~ 3 ‘| I I’ 3 > s(t) -5 0 45 9 19 Figure 8.13 Problem 8.2. Problem 8.3 Given a 6 kee band-limited signal in the frequency range of 22.6 kHz to 30: What is the minimum s ‘ampling rate required to completely specify the signal? 290! [INTE Nov Solution: Giv ; ¢n frequency range of band-limited signal = 22.6 kHz to 30.6 kHz 4Sampling Theoy * 55 band me kHz carmpling Fates fr = Ym = 2 8 KHz = 16 kHe pe spectral 1a ing rae anc maxi inge of a function ext - us wm sampling i from 4.4 MHz to 5.5 MHz, Find the nee) Fa = 4.4 MHz to 5.5 MHz yt fn = (5:5 ~ 4.4) MHz = 1.1 MHz fe= Yn = 2% 1d MHz = 2.2 MHz ad fr 2.2x10° jgths oi ao aa 5 wii sampling Fa ium sarpli = 0.45 ls. : 1g time, Ts = at 5 A signal x(0) = cos 2m 1001 + cos 27 2201 is sar a ig i impled at f, = 300 Hz and the sgnal (0) is passed through an ideal LPF with bandwidth, B = 150 ing i ed sig pectrum of y(0). Hz. Assuming ideal prot on sketch the frequency Sl x(t) = cos 2% 100 + cos 2m ae sion: eH refre,the frequency components are fi = 100 Hz, fy = 220 He jaf; = 300 He and bandwidth, B = 150 Hz. pled signal would have the following frequency components: The spectrum of the samy fohthy th : 100, (300 + 100), (600 + 100), ..- = 100, 200, 400, 500, 700, «+ = 220, (300 + 220), (600 + 220), ..+ = 220, 80, 520, 380, 820, ... es only 80 Hz and 100 Hz fre fhth th - quency ‘components 4st bandwidth, B = 150 Hz, the LPF pass 0e2) te ouput Te sgeora Hf) of the sampled signal y(9 is shown it Figure 8.14. re) , oe < ‘ ssid =f ay A t t “S ’ i ' ' 150 Ga ae OE LPF ouput Figure 8.14 problem 85-386 _+ signals and Syatoms Vroblem 8.6 ‘The signal x(¢) = sinc? 51 is ideally 'Y sampled at = : , Or Reonsiructed by an ideal LPF with bandwidth B= 5 I, unity gain al 40, Determine the reconstructed signal y(t), 2 Solution: Given X(t) = sinc? 5, Sampled at ¢ = 0, 0.1, +0.2, ... Bandwidth, B= 5 Hz The reconstructed signal is given by: yt) =2Bk D *(nT,) sine 2B — 1g ~nT,) where, k = gain of LPF f = 0 seconds, O1s xnT,) = sinc? (5nT,) = sinc? (0.5n) y= 10 > sine (0.5n) sine 10(1 ~ 0.1ny when, n = 0, y(t) = 10 sinc?) sine 101 = 10 sine 10¢ (sine (= When, n = 1, y(0) = 10 sinc? (0.5) sine 10(1 — 0.1) = 10 x 0.405 sine 10¢~ 0.1) <1, YO = 10 sinc? (0.5) sine 10(¢ + 0.1) = 10 x 0.405 sine 10(¢ +0.1) *2 or greater, sinc* (0.5) is negligible and other terms are neglected, Therefore, y(t) = 4.05 sine 1000 — 0.1) + 4.05 sinc 10(t + 0.1) + 10 sine 10r Problem 8.7 A rectangular pulse with duration r= 2 is sampled and reconstructed using = ideal LPF with B = f,/2. Determine the Output signal when T, = 0.8, Solution: A rectangular pulse is given by © ‘) l otst x(t) = rf t\= T 0, t>1 Therefore, the sampled signal is given by: 1, nT, <1 vor =a( =f i Given t= 2 0, nT, >1 i fi Bois Given 5 Therefore, Output, YO= LY xH7,) sinc 2B = nT,)—_—— Sampling Theory * 357 we first find the minimum value of y, 2 a8 we It a he it bi wre tyes ln 8 1 => 0.80 ee n¢ 1.25% (1.251 = 1 (assuming imeger) iy we aia isine ft -+8ine Fi 0.8) + sine +0.) J qaceauses MONT) = 1 for nt, $1), yal N* vio = sine it sine ft ~ O38) + sine f+ 0.8) i ne For the analog signal x) = 3 cos 100z¢ a Se ini sping ate to avi abaig 5 ter ae signal is sampled at the rate, f, = 200 Hz, what is the discrete-time 0 Se signal 0 suppose tl 5 supose pune alte iy Wt isthe fais it alter sampling? sined i" al is sampled at the rat ( J; = 75 Hz, what is the discrete-time signal i nipling? frequency 0 y{_)sinzag,t—m : gz] (2 fgt =m)‘Sampling Theory ° 361 =D cedun fmt =n), 20 2 Sry ; b#-———_ LPF. f. = 400 Ha, a Figure 816 Problem §.13, Wh the ideal LPF with f. = 400 He, the frequency Components. Present at the. Output will 160 Hz, 200 Hz, 300 Hz and 320 Hy. ty 160 Hz, 2 5 7 lem 8.14 Determine the minimum Sampling rate of en of Ia + jo). Assume that the Magnitude of snoiaed by sampling is at least 10 dB below the large: a signal x(0) = 6° (1) with Fourier f the largest frequency component ‘St spectral Component of x(1), Sion: Given x) se “ X(@) = - a+jo 1 @)| = MO 1 tea IF = 201og| = -3 dB - 20 log a Nosg (Foy = Va, which is the maximum value. -d as follows: “ale of w for which \F(@)| is 10 dB below the -20 log a can be obtaine FF@| = ~20 tog a 20 log V10 = 20 log (<5| : 364 + Signals and Systems a which [F(o)l will be 10.8 below the maxi num sampling rales a = 2(30) = 6, where «represents oan ic. @ the min Mn 3 q My ' Mreaege Mey Problem 8.15 Given a continuous-time signal x) with Nyquist Tate, i), Nyquist rate for the continuous time signals given below: ° ro 2(1) cos Wot. om, Det thy Solution: Given x(1) has Nyquist rate of a, X(@) = 0 for Jol > alr Therefore, (a) Given wt) = 2) Therefore, Yo) = 1x10) * Xa) 2n Itis clear that we can guarantee that YO) = 0° for lol > wy Therefore, the Nyquist rate for y(1) is 2ay. (b) Given M1) = x(0) cos Wot Therefore, Yo) = FM oa) X(@-ay)} It is clear that we can guarantee that Y() = 0 for jo > wy + 2 ie. ¥(@) = 0 for o| > wy + oe Therefore, the Nyquist rate for y(t) is 2(3@/2) = 3a. OBJECTIVE TYPE QUESTIONS 1. In communication, the sampling technique leads to (a) High speed (b) More efficiency (c) Low cost (d) All the above, 2. The minimum sampling frequency is called (a) Nyquist sampling rate (b) Pulse rate (©) Carlson frequency (d) None 3. According to the sampling theorem, the signal should be sampled at least (a) Twice each cycle of its lowest frequency (b) Twice each cycle of its highest frequency (©) Guard time should be as large as possibie. (d) Noney Sampling Theory _¢ 365 ata signa ai Ty ne N eof / yuist wal ie (b) 1p, tah & 9 (hen aximum frequency of 19 4-45. ° a naive aren Kitz is Sampled at Nyquist rate The time interval 5 aren wtp as (b) 50 us e 358 b @ 10 ps «7 gimited 8A FAS M0 SPectal gonna imi Pome above the f mean be uniquely determined by ig" foo teduency of 100 kHz. The o Mla be uni y ‘One "Mform intervals of duration es thes “5s Ss a om " @) 50 ns On ass an LPF with ¢ anal 8 £0 pass an LPF With cute zavoice 8 samples/s, Fequency of 4 KHz, The sampling rate is +0 (b) 2000 fy sn0 (@) 100 « the sampling rate is always between * (4) 0nd 2B (©) 2B and 4p jo) Band 2B () Oand B 4 The basic restriction on the reconstruc tion of a sampled signal (The signal must be passed through an LpR mae (9 The sampling must be in impulse train form (6) Both (a) and (b) ( None 11 Which of the following sampling techniques is Preferred? (@) Uniform (b) Non-uniform (¢) Both (a) and (b) (@) None \l Iessence, practical sampling is (@ PCM (b) PPM () PAM (@) PDM 1 Forstandard 300 to 3.4 kHz voice Channel, the standard sampling rate is — samples, (@) 8000 (b) 400 (© 500 (@) 100 in 6 7 rely distortion less received Signal in pulse modulation, itis required that the speed ld be (4 More marco the Signal frequency, ‘ ‘re than twice (b) Less than twice ( ss than 7 (@) None Ss than the Nyquist rate, then e used to recover the original signal ian Sampling rate is je ry i, filters can b rs Ch wid increases (Noe “@PACity increases ~f-B8 (d) None ‘erlaps the base bands, the distortion jg called (a) cross-talk (b) cross-over distortion (©) Aliasing (d) None aime for aliasing is (a) cross-talk (b) cross-over distortion (c) Frequency fold-over distortion (d) None fal from samples is. due 10 the los frequeney components, (6) 1000 @ 10 eat— Sampling Theory _* 367 i * Sample-and-hold (S/H) circuit is I a itself (6) Constan ne sigm ) None nos amine Me Sampling page batymu Frequency Wiee the wi 4 ©) Ma imum frequene ganda (4) Nong. "eauency A oxygut te FF the Sigal 0 = 3g, 100% + 10 sin 400g j 3 oo () 109 fs AOD er is meas @ W 50 re Nysuist interval for the signal x) = 7 a Si seconds, ja) 0.002 () 0.004 2 (d@) 4 re low pass filler used 10 eliminate al 8) Matched filter jo Prealias filter iasing is called (b) Interpolation fitter (d) Wiener fitter avers 10 Objective Type Questions 2. @) 3. (b) 4) Li 5. (b) 6. 8 10. (b) I. (©) @ 43, ic © 2) 3.0 (0) 25. (6) 26.) ee : yb 30. (c) +(e) 28. (a) REVIEW QUESTIONS | Sute and explain sampling theorem. {JNTU: Noy 2003) mpling theorem for low pass and bandpass signals. Prove the and analytically and mention various results used. UINTU: Nov. 2002, 2003, 2004, 2007; May 2004; Feb. 2007, 2008] graphical example, explain sampling theorem for bandlimited signals, [INTU: Feb, 2007, 2008; Nov 2007] Sampling theorem, Prove graphically with neat diagrams in time and fins and analytically proving the results used. 2 Sute and explain sa theorem graphically 3 With the help of + Saye \niform "eduency dom; [JNTU: Nov. 2004] 5 Bilin the : ‘tin the Signal recovery (reconstruction) from its sampled aaron Nor. 2004 6 pai "the filter characteristics of LTI systems. 5 May 2004, 20051 [JNTU: Nov. 20025y 368_+ Signals and Systems 1. Zz 14, 15. + A signal x(0) is given as x(0) = 6 cos 10 mt. This si . The signal x(t) = cos Sar + 0; . Define the following terms: What is aliasing effect? Draw in frequency domain 'gnal is samplea 4, The sampling frequencies of the impulse train are THz and 14 Hy, the sampling frequency of 7 Hz and 14 Hy. maximum interval of sampling, Y a train of pulses of wi @) Find the expression for the sampled signal, (b) Determine the spectrum of the sampled signal and sketch it, (a) Nyquist rate (b) Nyquist interval (c) Sampling rate (d) Aliasing (ec) Aperture effect Explain the flat-top sampling method, Explain the natural sampling method, Compare various sampling methods,