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    5 views12 pages

    Digital Signal Processing Dec 10

    Uploaded by

    Rajat Sharma
    fde

    Copyright:

    © All Rights Reserved
    Download as PDF or read online from Scribd
    Flag for inappropriate content
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    . B.TECH. SEVENTH SEMESTER EXAMINATION, 2010-2011 TEN-701 Time : 3 Hours Note : Auempt ALL Questions. -1.. Attempt any four parts ofthe follewing:~ (Sx4=20) (a) Determine the response ofthe following system to the UP signal : sve f-ssns| . [0 otherwise @ yar@-D 0 Yer ne ‘Ans.(i) "This.system simply detays the input by ‘one semple. ‘Thus its ofpis given by ato) ={-.0392,40.12.30, } Gi Inthisthe system advances the inputone sampic into the future, The value of the ‘output at time m= 0 is (0) = a(N). The response ofthis systems aa) = { 93.240120, | (b) For each of the following impulse response of LTI systems indicate ‘whether or not the system int eausal : @ hyn 2u (a+ 2)—wn—21 i) C4)" win =1) (The system is non causal since the ‘output depends on future values of + the input also. for n=O8(0)= u(2)—u -2) Total Marks : $00 vated [4], wal) ‘The system is causal since the ofp depends on present and past values. uct) (©) For the following impulse-response of LT system indicate whether or not the system in stable: Ha] =sin(nn/3) ula) ‘Ans. The system is casual, since the summation of sn [tote ree yeni stable. h(n) will decay exponentially owaeds zero as ‘n' approaches infinity. (@_ The given sigaal is periodic oF not, if periodic calculate period : x(n] 2 om) ‘Ans (n= 20089.” for every value of “ni, x(n) is periodic with fundamental frequency k A= § orpetiod = 3 (©) Find the DFT ofthe sequence : Hln)=1 for 0S nc2=Ootherwise Ais. x)= Mfor0sns? k= =O oerwvise ok 513072) po © Sincw 72) 2xk Replacing = = [ay = SORKIN [EES ‘sieety®). () State and explain “time reversal of a sequertce” property of DFT. Ans, Time reversal of DFT 16 adn) 9 x8), then Lileu)hy =a(N=n) LL X( (Ry = XNA) Reversing thx N Point sequence in time is equivalent to reversing the DFT values. x DET {AN —n)} = Sox =n) er Af we change the index frorim to m= n -n then ket DET (iW =m} = Shaq ere snee mt = Sun ete = Baan oP extn any four ofthe following : (6x4=20) sampling theorem. Draw trum of a sampled signal explain aliasing. Ans. Sampling Theorem : A bandlimited continuous tne signal, with highest frequency (bw) ‘" Ha| can be uniquely recovered from its samples provided thatthe sampling rate F, 2 2B samples/sec Bandlimited analog signal x40 Spectrum of I FiXa (F + Fi) =F, oR F +t Atiasing : the sampling frequency F_isselecied such that F, < 26, the periodic cont svation of X,(P results in spectral overtop, xn) ote” x) Ry “OF ‘Thus the spectrum of X(F/F,) of the discrete time signal contsins aliased frequency enmponeats of ‘anatog signal spectrum X,(). Theend result is the aliasing ©) Define all pass systems and minkmom phase systems. ‘Ans, All pass systems : It is defined as A system that has @ constant magnitude response for all frequencies [Hw =1 Oswsn ‘Minimura phase system : When all the er0s sins the wit cel cach term nthe rod of Hoe) =bo(1 =F) = 927) LImage™) ‘Comesponding ta teal valued zero, will undergo net phase change of zero betweea w= Oand w= Fundamental of Digital Signal Processing © Gi) 1, Also each pair of complex conjugate factess in ‘io wil undergo a nei hase change of zer0 iv LH(x)-£H(O)=0, The system is called ‘minimum phase system. | (©) Explain the need for multirate signal processing. ‘Ans. Multirate sign processing : In many practical application of DSP, one is faced with problem of changing the sampling rate ofa signal, either increasing it or decreasing it by some ‘amount. In telecom systems that transmit and receive different types of signals thers isa require: ‘ment 19 process the various signals at different rates commensurate with the corresponding bt of te signal. The prockss of converting a signal from a given rate to h different rate is called ‘Sampting rate convertion. Systems employing ‘mukiple sampling ratedin the processing of digital signals are called muttrate DSP systems. (@, Explain how sampling rate can be {increased by.on Integer factor. ‘A. An crease inthe smpling ae by a iteger factor of I can be accomplished by interpotating { = t new samples between successive values ofthe signal i Let vim) denote a sequence with a rate F, = JF, which is obtained from x40) by adding 1 1 2et0s bw successive galues of xn). Thus wimp fxd) m=O. 1, £2... Yo ottierwise and its sampling ral is identical to the rate of kor), This sequence has z—tansform D xem" exes wa= SS veme" “The corresponding spectrum of v(m) is obtained on the unit circle, ts 16) = X60.) Where fy, © frequehey variable relative «new sampling rate F, = P4F/F, Now F9=1 F, and hus bxows 0 Bx Se T © We TMT T “xo ‘The spectra (wand v(w,) are shown, the sampling rate increase, obta samples between successive values of x(n), resuli in a signal whose s repetition of ip signal spectrum XGv,). (©) Explain the use of oversampling to simplify the process of brie. by the addition of (L-1) Zero um x0, isan fold periodic g-tondigital conversion in ‘Ans. The over sampling in A/D ¢ snversion is done to increase the samplibg rate of the signal to the point where a low resolution quantizer suffices, By oversampling we can signal values between successive samples and thus reduce the resotutio} for a given fixed SQNR, a redvciion inthe variance of the signal to thd the number of bits inthe quantizer. (© With the help of block diagram explain Discrete time p ‘Ans. We first solect the b.w of the signal to be processed, since the signa sampling rate. The continuous s-gnal is prefillered to remove the hi thus waste the channel b.w to transmit for e.g. speech signal, othervis will be meeded to represent thes: high frequency components. When selected we can specify the sampl ng ate, we proceed with design of converting. acontinuous signe! toa digital sequence by a A/D convers by a digital systemfprocessor. Ths digital processed signal is again con DIA ccaversion and finally post filter to remove any high frequency ‘oFiginal components of the continuous signal Analog the dynamic range of the requirement on the quantizer [quantiyed allows us wo reduce of continuous time signals. b.w determines the minimum frequency components and during digitization extra bits desired frequency band is Jopetation to be performed like |. The signal is then processed d back to analog signal by ponents and preserve the igri (0 [AID] 30) [Digna] 9) [BIA | 9 [Pow (coral EEE conenca] * fassal > feoneniod neh iO signal) 3. Attempt any two of the following : (a) Obtain the cascaded realization for the following systems : (43/2274 4/22) 03/22 @ Hw @ a=W22"! +¥4e*) BO Tae dat aOR Wa RED fer dle (i) H@)= f 7 2040 ©) Develop Cascade ond F ‘parallet realisation structures for: 1645/2449) 24 2 41/08 2? 1 vae eae? fore Sety Ans, Hig)= 2 Hae Hed=dH (0440 wont ete! -45e Ta1,t,2 prtede (©) What ig the effect of roundoff noise in ightal filters? Analyse the direct forma IR structure, ‘Ans. Round of noise in digital Miter : t0 the realisation of a digital fitter either in digital ‘hardware ori software on & digital computer the quantization inherent in the finite precision _aithmetic operations render the system nonlinear. In recursive system the nonlinearties due to the finite precision arithmetic operations often cause periodic oscillations to occurs in the output, even ‘when the input sequence is zero or some nonzero ‘constant value. Such oscillations in recursive system are called limit cycles and are directly atwibutable to round off Hye) Direct form HR Structure . (i) Direct form- 1 realization:- H(Z)= H (0) Hale) where (2) consists of 2er0 of H(2) H(z) const of poles of He) ow Hy = sae Hto= ms Dae co By staching the nlpole system in cascede with H;(2) we obiain ditetform this realization equces M+N-s1 multiplicaions, M+ N adding & M +N‘ | memory locations xo erste 5 Direct form -It ‘The difference equation for all pole 24a) ¥ finer Hen)= — raw(a—b) +202) ro Since w(n) is the input to all zero system, its ofpis 208) = Lam ‘The above equation involve delaved versions ofthe sequence w(n). Only a single delay line or a single set of "memory location is required fer storing the past values of w(n). The resulting sarueture is called direct form 1. This strectore requires M +N + 1 multiplication, M +N addition and ‘max. of MN) memory locations 4. Attempt any two of the following : {) Explain the procedure for designing an FIR fiter using Kaiser Window. ‘Ans. Kaiser window Furst the specifications must be established selecting the desired w, and w, and the maxi- mum tolerable approx. errors for window design the resulting filter will have the same peak errr (8) in bath pass and the sop band. i) The cor off freq. of underlying ides! LPF w+, aust be found. Set w, = “ES = 0.5 (iii) To determine the parameter of kaiser window compute Aw=¥,-¥, =02%,A=— 20 0g105= 60 ‘objain the required values of B & M using. [4102(A-B7) A>s0 B= |5842(A-208.4-+ 07886 A-2) 21S AS 50 loo ae Ans 3M = Tyisaw vj The impulse response of the filter is computed using rey S0LH=M egy rin—M72) on) = [Zale =len~ay/02y"4) wa) ra | Osasm sang a) sfoe—foe—ovet ay = SLA). : n=) iyi lo atbewise ) Discuss the Bitinear transformation design techniques for HR filters. ‘Ans. The bilinear transformation is a conformal ‘mapping that transforms the JO axis into the unit circle in the z-plane only once, thus avoiding aliasing of freqeuney components. All points ia the RHP of ‘s' are mapped into corresponding points outside the unit circle in the z-plane Let us consider an analog linear filter with b tem function His) = syst Here ‘This system is also characterized by differential equation - - ro We integrate the derivation and approximate the integral by the trapezoidal formula ye)= fytsr(be)+ yt0) ¥'() > derivative of ya) ‘The approximation of the integral by the teupezoidal formula at r= nt and ig = nT T gives so = Eytan yer Deyo = ‘The differential equation (i) evaluated st gives ¥ Ga) =-ay(ns) + balnT) co) Using id eqn. to substitute forahe derivative nega. (i) and thus obtain difference equation for the equivalent discrete time system with y(n) = you and a) xa) we get. an . 1+ Ym na [1+Zeo-(1-Z pean 4 (ny xin} The ztransform of this equation is at at) 1 122 ( z xo { > ya) =Zeex ‘The system function ofthe equivalent digital filter ery ies nee. 2 UE) Way 6 L (Clearly the mapping from the s-slane tothe 2 plane ‘This is calted bilinear transformation (©) A filter is to be designed with the following desired frequency response: 0 =r/4

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