ayy Discrete-time Signals and Linear Systems 1.7391 Questions and Answers - p al | What do you understand by the terms: signal and signal processing. A signal is defined as any physical quantity that varies with time, space, or any other independent variable. Signal processing is an operation that changes the characteristics of a signal ‘These characteristics include the amplitude,shape,phase and frequency content of a signal What is a Deterministic signal? Give an example. A Deterministic signal is a signal exhibiting no‘ urteertainty of value at any given instant of time. Its instantaneous value can be accurately predicted by specifying a formula, algorithm or simply by its describing statement in words. Example: v(t) = Ap sinw(t) . What is random signal? A random signal is a signal characterized by uncer- tainty before its actual occurrence. Example: Noise Define (a) Periodic signal (b) Non-periodic signal. A signal x(n) is periodic with period N if and only if 2(n + N) = 2 n. ) for all If there is no value of N that satisfies the above equation the signal is called nonperiodic or aperiodic. . Define symmetric and antisymmetric signals. (AU ECE*07, AU EEE"07) A real valued signal «(n) is called symmetric if 2(~n) = 2(n). On the other hand, a signal x(n) is called antisymmetric if x(—n) = —a(n), . What are energy and power signals? (MU Oct’ 96) ‘The energy of a discrete time signal «(n) is defined as co Energy E= by le(n)P 00 A signal x(n) is called an energy signal if and onl i: ; relation 0 < E < 00. Y if the energy obeys the For an energy signal P = 0 ‘The average power ofa discrete time Signal 2(n) is defined as D 1 x Pe im a DL keegital Signa 1.240 Dit , 4, Define the following (wAnalog signal signal (1) Anani (B) Discrete-time signal (c) Digizgy amplitude varying continuous), ‘al is continuous in both time inusoidal function, the on having an 2 _ Hence, an analog sign fa a Examples of analog signals are the si step function, output of @ microphone. / ical (2) A discrete-time signal is a function defined only at 7 i jar time in. stants. It is discrete in time but continuous in amplitude. An example jg temperature recorded at regular intervals of time in a day. of discrete-time signal which is discrete (3) A digital signal is a special form si in both time and amplitude, obtained by quantizing each value of the discrete-time signal. These signals are called digital because their sam. ples are represented by numbers or digits. Examples of digital signals include the dot-dash Morse code, the output from 2 digital computer ete alog signal is a funct 8. What are the different types of signal representation? Signals are represented in various ways @ Graphical representation (ii) Functional representation (iii) Tabular representation (iv) Sequence representation (i) Graphical representation ii) Functional representation z(n)=1 for n=0,1,4 =0.5 for n=2,3 (iii) Tabular representation 01 4 2(n)|1 105 05 41 (iv) Sequence representation a(n) = {1,1,0.5,0.5, 1}Discrete-time Signals and Linear Systems 1.241 9. Define the following signals (i) Unit sample sequence (ii) unit step signal (i) eee i) Unit sample sequence (ii) unit step signal (iti) (i) The unit sample sequence is denoted as 5(n) and is defined as 6(n)=1 for n=0 =0 for n#0 , the unit sample sequence is a signal that is zero everywhere, exgept atn (ii) A unit step signal is denoted as u(7) an¢ is defined as 1 forn>0 u(n) = 0 forn<0 i.e., the unit step signal is a signal having unit magnitude for all » > 0 TT 4324 (ii) A unit ramp signal is denoted as u-(n) and is defined as u,(n)=n for n>0 =0 for n<01.242 Digital Signal Processing 5 bitrary sec 10. Give the analyticai and graphical represersado” oe merce (MU Oct’ 96) Graphical represe ice is given by jon of an arbitrary sequence IS £1 tation of 3 x(a) * | ‘ 4 2 into a sum of unit sample sequences If ed unit impulse 6(n — k), the result k, where its value is We can write any arbitrary sequence «-(n) wwe multiply two sequences 2(n) and delay is another sequence that is zero everywhere except at n= 2(k). Thus a(n)é(n - k) = 2(k)4(n - k) If we repeat this multiplication over all possible delays, -oo < k < oo, and sum all the product sequences, the result will be a sequence equal to the se- quence x(n), that is x(n) = > 2(h)5(n- k) 11. Define the following (a)System (b)Discrete-time system (a) A system is i ) - es a defined as a physical device that performs an operation on a (6) A discrete-time system is a device or algorithm that operates on a discrete time input signal 2(n), accordin i 2(n), ig to some well-defined rule, to produce another discrete-time signal y(n) called the output signal. ™ x) Disa tide ), a Sea Signal 12. What are the classification of discrete-time systems? (a) Static and dynamic systems (b) Time-variant and time-invariant systems (c) Linear and non-linear (d) Stable and unstable systemsa Discrete-time Signals and Linear Systems 1.243 (e) Causal and non-causal systems (f) MR and FIR systems 13. What are the different types of operations performed on discrete-time signals? The different types of operations performed on discrete-time signals are (1) Delay of a signal (2) Advance of a signal(3)Folding or reflection of a sig- nal(4)Time scaling(5) Amplitude scaling(6) Addition of signals (7) Multipli- cation of signals Dhasa 14, Represent the following finite duration sequence x(n) = {2,4, 1 sum of weighted impulse sequences Given x(n) {24 -1,-2} ie., 2(—1) = 2;2(0) = 4;2(1) = 1;2(2) = —2 we can write 2 a(n) = >> 2(k)6(n - k) kena = 26(n + 1) + 46(n) — 6(n — 1) — 26(m - 2) 15. What is the property of shift-invariant system? (Oct 95) (or) What is a time-invariant system? (MU Apr'96), (AU May 2006) (or) What is a shift-invariant system? Give an example. (MU Oct’ 98) If the input-output relation of a system does not vary with time, the system is said to be time-invariant or shift-invariant. If the output signal of a system shifts k units of time upon delaying the input signal by k units, the system under consideration is a time-invariant system. Example: y(n) = 2(n) + 2(n.— 1) 16. What is a causal system? Give an example. (AU May 2006), (EIE AU '03) (or) ; (APR’ 98) What is a causal system? (MU Apr 2000) A system is said to be causal if the output of the system at any time n depends only on present and past input, but does not depend on future inputs. This can be represented mathematically as ¥(n) = Fla(n),2(n - 1), 2(n-2).1.244 Digital Signal Processing Example: y(n) = 2(n) + 2(n — 1) y(n) (MU Oct 2000) 17. Define linear system and give example. es the superposition principle according to ighted sum of inputs is equal to the A linear system is one thet satisfi to each of the individual inputs which the output of the system to a wei; corresponding weighted sum of the outputs If ys(n) = Ther(n)] and yo(n) = Tla2(n) Tlarzi(n) + a2ma(n)] = 2 T[2a(n) are any arbitrarily chosen scalar coefficients. ] then for a linear system | + a2T ix2(n)] where a) and a; Example: y(n) = n2(n). 18. Define (a) Static system —(b) Dynamic system (a) A static or memoryless discrete-time system is a system whose output at any instant depends on the input values at that instant but neither on the past nor on the future values of the input Example: y(n) = ax(n) (b) A dynamic or a system with memory is one in which the past inputs or outputs are stored to calculate the present output. Example: y(n) = 2(n) + 32(n — 1) 19. Define a stable and causal system. Stable system Any relaxed s,stem is said to be bounded input-bounded output (BIBO) stable if and only if every bounded input yields a bounded output. Mathematically, their exists some finite numbers, M, and My such that, (AU IT Dec’03) (AU IT Dec’ 03) (MU Oct’97} |2(n)| h(k)2(n — k) k==00 kano) 1.246 Digital Signal Processing The above equation that gives the response y(n) of an LTI system as a function of the input signal x(m) and the impulse response h(n) is called a convolution sum, 2S. What are the properties of convolution? (AU IT Dec’03) (a) Commutative property (n) « h(n) = h(n) + 2(n) (b) Associative property (x(n) « hy(n)] * ha(n) = a(n) * [ha(m) * ho(n)] (©) Distributive property x(n) * [ha(n) + ho(n)] = a(n) + ha(n) + a(n) ho(n)} 26. What are FIR and IIR systems? FIR system: This type of system has an impulse response which is zero outside a finite time interval. Example: h(n) =0, for nN IIR system: An IIR system exhibits an impulse response of infinite duration. 27. What is the property of recursive and non recursive systems? Recursive system: This type of system has the property that output y(n) at time nis a function of any number of past outputs u(n—1),y(n — 2),...y(n — N) as well as present and past inputs 2(n),2(n— 1),2(n ~2)...2(n— N). bea y(n) = Tle(n),2(n—1),-..2(n—N),y(n—1),y(n 2)... y(n —N)] Non recursive system: In this kind of system, the output y(; ‘n) depends only on the present and past input signal values, ie, u(n) = Tle(n),2(n — 1),2(n ~2),...0(n — N)) | 28. A causal system is one whose impulse ! True/False response h(n) = 0 for n < 0. True 29, A recursive system described by a linear con stant differenc neon and time-in variant. True / False ference equation is lin True 30. A linear system is stable if its impulse response is absolutely summable. True / False TrueDiscrete-time Signals and Linear Systems 1.247 31. How can you find step response of a system if the impulse response h(n) is known? We have y(n) = x(n) * h(n) For input x(n) = u(n) y(n) = u(n) « h(n) = YS un- hk) ‘ru(n—k)=0 for k>n ene Se 32. Determine the unit step response of the LTI system with impulse response h(n) =a"u(n) al <1. Unit step response 33, Define Fourier transform of a sequence. (MU Apr 2000) The Fourier transform of a finite energy discrete-time signal x(n) is defined as SS anyon 00 34. What is the sufficient condition for the existence of DTFT? The sufficient condition for the existence of DTFT for a sequence 2(n) is > |2(n)| < 00 é 35. State Parseval’s energy theorem for discrete-time aperiodic signals ‘The Parseval’s energy theorem for discrete-time BS) lene aperiodic signals is given by at1.248 Digital Signal Processing (MU Apr'99) 36. Define DTFT pair. j is The Fourier transform patt of. discrete-time signals 1 wy em ain) = 5 f XO” a xiemy= DY aime" 37. What are the properties of Fourier spectrum of a discrete-time aperiodic sequence? “The Fourier spectrum of an ap period 27. eriodic sequence is continuous and periodic with tansforms ofa discrete- 38. Whatare the two basic differences between the Fourier .s time signal? time signal and the Fourier transform of a continuow a. Foracontinuous signal, the frequency range extends from —o0 to co. On the other hand, the frequency range of a discrete-time signal extends from torn. b. The Fourier transform of a continuous signal involves integration, whereas, the Fourier transform of a discrete-time signal involves a summation of terms. 39. Find the Fourier transform of a sequence 7") =1 for —2n <2 =0 otherwise X(e*)= YO a(nje en yew 2 Pi 4 4 1p eniY 4 ediw = 14+ 2cosw + 2cos2w 40. The discrete-time Fourier transform of real part of a sequence a(n) is : Xe(ei) 7 41. The discrete-time Fourier transform of 5X (n)is Xo(e™) 42. The discrete-time Fourier transfor sform of even com ponent of signal is Xp(ei)Discrete-time Signals and Linear Systems 1.249 43. Define Fourier transform of a sequence and give its symmetry property. ‘The Fourier transform of a sequence is given by X(e#”) = > x(n)e7" ‘The symmetry properties are shown in page 1.120. - 44. Obtain the impulse response of the LPF described by H(e =1 for |u| < We =0 for we <|w| <2 (MU Oct’97) h(n) = sf H(e*) "dw ad jun 1 jn |Weo Oday = emit, 1 sin Weon aoe” = eter] = Se 45. Define the frequency response of a discrete-time system. (MU Apr’99, Oct’98) Consider a class of input sequence 1(n) = e? Then y(n) = > A(k)a(n —k) = > h(keseer-®) rad co = ef Se h( ke I" = fH (e*) ko where H(e) is called the frequency response of LTI system. 46. Obtain the frequency response of discrete-time system with impulse, function (a) h(n) = b"u(n) for |b] <1 (Mu Oct’98), (AU 2006) (b) h(n) = (0.3)" for n>0 (Mu Apr’98) (a) The frequency response of a system is given by oo . H(e#) = YS Ane" 00 = Score edn = Ne —jwyn n=0 (b) For b = 0.3 1 H(e) = 0.3e-341.250 Digital Signal Processing : (eo) of an LTI system? 47. What are the properties of frequency response H(e) of H(e2) isa eontinvous function of i) ig periodic with period 27 sponse Hl c with respect tow = The frequency re fe function is even symmetti 1 3, |H(e)|, the magnitud! wet z i i¢ wi ect to 4. ZH(e™), the phase function is anti-symmetri¢ with resp n= 0,41, £2,-.++ ‘ 7 ier transform of x(n) = (0.8) 48. Find the Fourie 5 not absolutely summable. So For negative values of n, the sequence 2(n) i Fourier transform does not exist. 49. What is aliasing effect? (AU EEE’07) (ECE AU'03) Let us consider a band limited signal 2(#) having no frequency component for iO] > Su If we sample the signal x(t) with a sampling frequency F< 2fm, the periodic continuation of X(j®2) results in spectral overlap. In this case, the spectrum X (j) cannot be recovered using a lowpass filter. This effect is known as aliasing effect. 50. State sampling theorem. (AU IT Dec’03) A band limited continuous time signal, with higher frequency f; Hertz, can be uniquely recovered from its samples provided the sampling rate F > 2fm samples per second. 51. What is an anti-aliasing filter? (or) How do you prevent aliasing while sampling a CT signal ? (AU ECE May’07) The frequency spectra of real signals do not confined to a band limit mn. There we almost always frequency components outside Q,,. If we select sampling cquency F > 2fm using the sampling theorem, the frequency components a fn wll appear as low-frequency signals of frequencies between 0 and °F due to the aliasing effect and lead to loss of informati id ali Ing, we use an analog low pass filter before apie Fb veshege he ieee spectrum of the signal so that the frequency spectrun a This filter is known as anti-aliasing filter 52. Explain the frequency shifting property of Discrete-tir j If Flz(n)| = X(e™) of Discrete-time Fourier transform. then Fee a(n)} = X[eslo-wn) 53. Explain the time shifting property of discrete If Flx(n)) = X(e™) then time Fourier transform. Fla(n— k)| = @ Wh X (eh)Discrete-time Signals and Linear System 4, If X (02) is the Fourier transform of (1), find the Fourier transform r(-n) The Fourier transform of .r(n) is given by (e) = YS alnje Founer transform of c*(—n) is given by Fietin)}= Do *(-nyeo , = [= ane] = | y aime = X*(e"*) n=—c0 35, If X(e*) is the Fourier transform of x(n), what is the Fourier transform of x(n)? Ans: Fla*(n)) = Ye a"(nje" = ¥ al cme] =x") n=—00 what is the Fourier transform of 56. If Xe!) is the Fourier transform of zn), nz(n)? Fast) = 7X) 57. Explain the linear property of DIFT. ip Flex(n)|=2a(e) and Flea(n)) = ale") then Ffarei(n) + 422(n)] = a, X;(e™) + a2X2(e) 58, Find the transfer function ofthe system given by y(n) - dy(n-1) = a(n) + 3: Az(n - 1) (Annamalai University Apr’03) ‘The transfer function of the system is defined as Y(e) He”) = (*) = ¥ (ga) Applying Fourier transform for the difference equation we get ¥(e*)- jemy(e) = X(e) + gexte™) veh ie |x (efi )} 252, Digital Signal Processing 9, The Fourier transform of @ discrete and aperiodic sequence is . 59. Continuous and periodic a(n)=A MensM,, 60. The Fourier transform of ) _ herwise sin (™ + ) w ts sin $ 61. Find the transfer function of the 3-sample averager. Mn)=} for -1 a(nyen where 2 is a complex variable. In Polar form.z can be expressed as 2 — re, where r is the radius of the circle 2. What is meant by Region of convergence? The region of convergence (ROC) of X(2) X(2) attains a finite value, (EIE AU’ 03) is the set of all values of z for which 3. ROC of a causal signal is the of a circle of same radius r. Exterior 4. ROC of an anticausal signal is the interior of a circle of same radius r. True/False True §-Explain the linearity proper of the z-transform. Hf Z{2(n)} = X4(z) and Z{29(n)) = Xo(z) then Z{ayx1(n) + aze2(n)} = a1 Xy(2) + ayXo(z) (Beplain the ime shifting property of the z-transform. U2{x(n)} = X(z), then Z{v(m—k)} = 2-*X(z) 7. Explain the scaling property of the z-transform. H 2{a(n)} = X(z) ROC: ry < |z| < ra, then Zla2(n)) = X(an!2) ROC: Jal ry < |2| < Jal rp 4 Explain the time reversal property of the z-transform. #2{a(n)} = X(2) ROC: ry < |e| < ro then Z{2(-n)} = X(21) ROC: x << i 4 Explain the convolution property of the z-transform. "4ailn)} = Xy(2) and Zx2(n)} = Xo(z), then 4{ a(n) # x9(n)} = Xy(z)Xo(z) re—— 2.78 Digital Signal Processing sltiplication property of the setransform. 38. Explain the mi ielgernysl if Z{2a(n)} = Xa(2) and Z2{z2! 2) yd afeatndeate)) = 543 f xaos (7) 2° 2 relation in z-transform. 11. State Parseval’s valued sequences. then "f-24(n) and 29(n) are complex = 1 (2), FP emsiin) = gy f HOM (4) c n==20 itial value theorem and the final value theorem. ¥2. State the int Initial value theorem: If «(n) is causal, then 2(0) = Jim X(2) Final value theorem: If 2(n) is causal, Z[z(n)] = X(z), where the ROC for 'X(2) includes, but is not necessarily confined to |2| > 1 and (z ~ 1)X(z) has no poles on or outside the unit circle, then 2(c0) = lim(z ~ 4) X(2) 13, An LTT system with the system function H(z) is BIBO stable if and only if the ROC for H(z) contains the unit circle. True/False True 14. What are the properties of Region of convergence? (AU EEE’07) 1. The ROC is aring or disk in the 2-plain centred at the origin. _ 2. The ROC cannot contain any poles. 3. The ROC of an LTI stable system contains the unit circle. 4. The ROC must be a connected region, 15. Determine z-transform and ROC of the finite-duration signal. duration sign a(n) = {2, 4,5,7,0,1} X(z)= SO a(nThe Z-Transform 2.79 We have 2(0) 1) = 4;2(-2) =2:2(1) =7 2(2) =0;2(3) =1 substituting the sequence values we get X(z) = 22? 4424547274277 ROC: entire z-plane except z = 0 and z = 2 16. Find the z-transform and ROC of the signal x(n) = —b"u(—n — 1) 47. Find the z-transform of (a) A digital impulse (b) a digital step. (a) Since x(n) is zero except for n = 0, where x(n) is 1, we find X(z) (b) Since «r(n) is zero except for n > 0, where x(n) is 1, we find 18. The z-transform of a sequence x(n) is X(z), what is the z-transform of na(n)? If Z{z(n)} = X(2) Z{n2(n)} = 19, What is the relationship between 2-transform and DTFT? (MU Oct’ 96) ‘The z-transform of 2(n) is given by X(2z) a)2.80 Digital Signal Processing ( where = = re” 2) Substituting Bq, (2) in Bg. (1) we get Xx(re) = SS aln)r 8) ‘The Fourier transform of «(n) i8 given by X(e*)= YO ale" “4 1. In the z-plane this corresponds (3) and Eq. (4) are identical, when 7 u ia | Hence X (ei) is equal to X(2) to the locus of points on the unit circle |2} evaluated along the unit circle, or Xe") = X(2)|,_0 For X(e) to exist, the ROC of X(2) must include the unit circle. 20. What are the different methods of evaluating inverse z-transform? The inverse z-transform can be evaluated using several methods 1. Long division Method 2. Partial fraction expansion Method 3. Residue Method 4. Convolution Method ==00 Bi Express the z-transform of y(n) = 3° 2(k) in terms of X( y nd = ¥ 2k) +2(n) k==00 un—1)= S* a(k) B00 uln) = y(n ~1) + 2(n) Taking z-transform on both sides ¥@) = 27Y(2) 4 X(2 y(z) - X@)The 7 Transform 2.81 22, Find the convolution of the following using >-transform. r(m) =A 7-20): h(n) = (1,151) We know 1 1 2(n)2- = 14227) 4.5272 4.7273 + 2-5 26. Draw the direct form I structure for the second order system function 27} + bya H(z) = ny r y(n) >) ] ¢ J , 1 z | [e') 4, | " ¥ t‘The Fast Fourier Transfo, questions and Answers we |, Why FFT is needed? m1 4.53 (MU Oct 95, Apr’og) The direct evaluation of DFT usin, requires N? complex multiplicat for reasonably large values of N requires an inordinate amount g the formula X (k) ions and N(N (in order of 1000) of computation. Bi: number of computations can be reduced. For ex: the number of complex multiplications required N = 16, the number of complex multiplications re of DFT is 256, whereas by using DFT only 32 m SON! a(n jensamnkin 1) complex additions. Thus direct evaluation of the DFT y using FFT algorithms the ample, for an N-point DFT, using FFT is ¥ log, N. If ‘equired for direct evaluation tultiplications are required. What is the speed improvement factor in calculating 64-point DFT of a se- quence using direct computation and FF: T algorithms? or Calculate the number of multiplications needed in the calculation of DFT and FFT with 64-point sequence. (MU Oct’97, Oct’98) The number of complex multiplications required using direct computation is N? = 64? = 4096. ‘The number of complex multiplications required using FFT is N 64 1 Flee = Floss64 = 192. 4096 _ Speed improvement factor = 4998 = 21.33 2007 What is the main advantage of FFT? (AU May 2007) ji ier trans- FFT reduces the computation time required to compute discrete Fourier form. ; ‘i rp Calculate the number of multiplications needed in the calculation of DI it i e. using FFT algorithm with 32-point sequence For x point ber the number of complex multiplications needed using FFT algorithm is YlogyN. i S2log)32 = For N = 32, the number of complex multiplications is equal to “logy N= 32, the 16 x5 = 80. © What is PET? 1 fa ri ot e the DET. It The fast F transform (FFT) is an algorithm used to compute | © fast Fourier trans! use of the symmetry and peri 'y properties of tw le factor Wx © y! y f twiddle fac x a dicity properties o! im : : ely reduce the DFT computat ie. It is based on the fundame! evel ce the DFT comp ee a. ile decomposing the computation of DET of a se ence of length4.52. Digital Signal Processing ely smaller discrete Fourier transforms. The FET — srease factors, when compared with direct computatigy ately 64 and 205 for 256-point and 1024-point tans s, \N into successiv' provides speed-in DFT, of approxim: respectively. How many multiplications and additions are required to compute low DFT using radix-2 FF’ av ‘The number of multiplications and additions required to compute M-pojny oy using radix-2 FFT are Nlog,N and ‘logy N respectively. N-poiny 2 7. What is meant by radix-2 FFT? The FFT algorithm is most efficent in calculating N-point DFT. If the noni, of ourput points NV can be expressed as a power of 2, that is 2! where M is an integer, then this algorithm is known as radix-2 FFT algorithm 8. What is a decimation-in-time algorithm? (MU Oct’ 95) Decimation-in-time algorithm is used to calculate the DFT of a N-point se. quence. The idea is to break the N-point sequence into two sequences, the DFTs of which can be combined to give the DFT of the original N-point se. quence. Initially the N-point sequence is divided into two 4-point sequences z¢(n) and zto(n), which have even and odd members of x(n) respectively. The -point DFTs of these two sequences are evaluated and combined to give the N-point DFT. Similarly the 4-point DFTs can be expressed as a combination of {-point DFTs. This process is continued until we are left with 2-point DFT This algorithm is called decimation-in-time because the sequence x(n) is often split into smaller'subsequences. 9. What is Decimation-in-frequency algorithm ? (MU Oct’95, Apr’98) It is a popular form of the FFT algorithm. In this the output sequence X(t) is divided into smaller and smaller subsequences, that is why the name dec- imation in frequency. Initially the input sequence (n) is divided into ‘vo Sequences 1(n) and zr9(n) consisting of the first samples of a(n) and the last ¥-samples of z(n) Tespectively. Then, we find the point sequences f(n) and g(n) as , fl) =ai(n) + 2a(n) and g(n) = [er(n) — a(n) N The point DFTs of the above two sequences give even membered and of membered output samples respectively. , : The above procedure can now be iterated to express each point DFT * combination of two % -poj ; “+ we are lel " 2 point DFTs. This process i d until we with 2 point DET. pI is continue‘The Fast Fourier Transform 4.53 30. What are the differences and similarities between DIF and DIT. algorithms? Differences 1. For DIT, the input is bit reversal wi as for DIF, the input is in nat reversal The DIF butterfly is slighth ference being that the com subtract operation in DIF hile the output is in natural order. whe! tural order while the output is bit ly different from the DIT butterfly, the dif- iplex multiplication takes place after the add- Similarities Both algorithms require same number of operations to compute the DFT. Both algorithms can be done in-place and both need to perform bit reversal at some place during the computation. Ll. What is the basic operation of DIT algorithm? The basic operation of DIT algorithm is the so called butterfly in which two inputs Xmm(p) and X,,(g) are combined to give the outputs X,,+1(p) and Xm+1(q) via. the operation Xm+i(P) = Xm(p) + WEXm(9) Xm+1(q) = Xm(p) — WEXm(a) where 1% is twiddle factor. 12, What is the basic operation of the DIF algorithms? The basic operation of the DIF algorithm is the so called butterfly in which two inputs X;m(p) and Xm(q) are combined to give the outputs Xm4i(p) and Xm4+i(q) via the operation Xm+1(P) = Xm(p) + Xm(9) Xm41(q) = [Xmn(p) ~Xm(Q)] Wi Where Wf, is twiddle factor. Draw the flow graph of a two-point DFT for a decimation-in-time decomposition. -poi for a decimation-in-time algorithm is The flow graph of a two-point DFT vation nce Xm(q)’ Xp ail@ = Xp (P= Xm (Q) (on,cessing 4.54 Digital Signal Pro ph Hg (0) Xm 4 Xe si 1 XW) I Keg gD = Xu (P= Xo (Q) Xd “7 - 4 ws 2 inputs to the butterfly, Xm4.1(p) and ore Xpu(p) and Xmn(q) are the input ha ieee mien of the butterfly. The nodes p and q represent memory loca’ 14. Draw the flow graph of a two-point radix-2 DIF-FFT. The flow graph of a two-point DFT for a decimation-in-time frequency up rithm is : XyuiB) Xe) + XQ) wt Yo ¢ vv ‘ x0) Xa -&o-x@) yee where Xma(p) and Xm(q) are inputs, Xmea(p) and Xm+1(Q) are the oupus of the butterfly. The nodes p and q represent memory locations. 15. Draw the basic butterfly diagram for DIT algorithm. (AU EEE’07) The basic butterfly diagram for DIT algorithm is x6) NS XyuP)= Xa (P) + We, Xa) X@ Xml) =X, (p)- WE X,(@) a Xrr(p) and Xp(q) are the inputs to the butterfly, Xim4.1(p) and Xm+i(9) “re outputs of the butterfly. The nodes P and q represent memory locations. 16. Draw the basic butterfly diagram for Dif algorithm. The basic buuerfly diagram for DIF algorithm is # Xa) =X, (0) X,(@) Xl eal = XB) XQCQ)] WE linge ma x, m() ate the inputs, Xmnta(p) and Nyy 1(q) ate the pu butterfly. The nodes p and “represent memory locations.‘The Fast Fourier Transform 4.55 17, What is meant by ‘in-place’ in DIT and DIF algorithms? ‘The basic butterfly diagrams used in DIT and DIF algorithms are shown in Fig. | and Fig. 2 respectively. Xm (P) + Xq(@) Xm (p)+ We p Xm (P)+ Wa, Xq(Q) Po 4 X qi) Ws 4 Xa a NN ra My at XP) ~ Wy XQ Xa(@) + Kul) = XQ Fig.t Fig2 In Fig. 1 two lines emerging from two nodes cross each other and connected to the two nodes on the right hand side, These nodes represents memory lo- cations.- At the input nodes Xm (p) and X;(q), the inputs are stored. After the outputs Xm+1(p) and Xm41(p) are calculated, the same memory location is used to store the new values in place of the input values. An algorithm that uses the, same location to store both the input and output sequences is called an ‘in-place’ algorithm. 18. How can we calculate IDFT using FFT algorithm? The inverse DFT of an N-point sequence X(k); k= 0,1,....V — Lis de- fined as N= x(n) = x SS x(e)Wy"* k=0 If we take complex conjugate and multiply by N’, we get N-1 xeewst k=0 The right hand side of the above equation is DFT of the sequence X*(k) and may be computed using any FFT algorithm. The desired output sequence x(n) can then be obtained by complex conjugating the DFT of Eq. (2) and dividing by Vw give 1 [Mt . al) = [x owe koial ss 44.56 Digital Signal Processing he 4point radix-2 DIF-FET butterfly structure for DET 19, Draw the 4 Moa L ate X(0) x(1) De we ~ X02) a > aN _x(1) x0) SS XB) 20. Draw the 4-point radix-2 DI T-FFT butterfly structure for DFT. - x(0) (0) hee Sh xa) X(2) XO wi 2) 16) yp So, XG) 21. Find DFT of the sequence x(n) = {1,2,3,0} using DIF algorithm. x(0)=1 Wy wi 6=X(0) x(1)=2 V4 2=x02) xQ)=3 We 2-2)= x0) x@)=0 -2+2}=X@) The twiddle factors are WP =1; Wz} = e73?7/4 = —j X(k) = {6,-2— 27,2, -2 + 23} 22. What are the applications of FFT algorithms? The applications of FFT algorithm include (i) Linear filtering, (ii) Correlation, (ii) Spectrum analysisTngin Infinite Impulse Response Filters 5.11] ns and ANSWeTs esto w 1, Gi van} the magnitude function of Butterworth filter. Whi A iat is the effect ing order of N on magnitude and phase responce? est of qhe magnitude function of the Butterworth filter is given by 1 [#60 =——1 ny yay P+ (a)") where V is the order of the filter and 0 isthe cutoff frequency. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N increases. The phase response becomes more non-linear as Nv increases. 2. Give any two properties of Butterworth lowpass filters (AU ECE Nov'06) 1, The magnitude response of the Butterworth filter decreases monotoni- cally as the frequency © increases from 0 to co. 2. The magnitude response of the Butterworth filter closely approximates the ideal response as the order NV increases. 3. The poles of the Butterworth filter lie on a circle. 3. What are the properties of Chebyshev filter? - (AU ECE Nov'06) 1. The magnitude response of the Chebyshev filter exhibits ripple either in passband or in stopband according to type. 2. The poles of the Chebyshev filter lie on an ellipse. 4. Give the equation for the order of N and cut off frequency Q. of Butterworth Sitter, The order of the filter quency % “hete a, = stopband attenuation at stopband frequency Os frequency 2p = Passband attenuation at passband frequency °% Ny Qe = (loner — 1)5.110 Dist” function and its magnitude "esponse \ cave the cebsser ‘The transfer function 1 1 Le | . H(i2) i +jeCn (#) | filter related to the ripple in the Passbang . meter of the . ang where € iss ° mh coder Chebyshev polynomial defined as a iS Gales Cy(a) = 608 (Neos?) for |2| <1 filter transfer. Chebyshev filter is given by a C(x) = cosh (N cosh7!z) for |z|>1 The magnitude response 15 shown in Fig. 1. Fig. 1 6. Poles of Rutt.rworth filter lie on an ellipse True/False False. 7. Poles of Chebyshev filter lie on an ellipse True/False True. 8. Chebyshev filter poles are close to jQ-axis than those in Butterworth fil True/False True. 9. A causal and stable 11R Silter cannot have linear phase True/False True. y 10. Distinguish between the frequene ype filter Pr odd and N even, quency response of Chebyshev typ “ a st The frequency response curve stants from unity for odd values of Ns *" from Vint for even values of N.and zeros and le behaviour in Fig. 2 12. Give the Butterworth filter transfer function and its magnitude characteristics for different orders of the filter. ‘The transfer function of a Butterworth filter is given by HQ) = x 145 (8) The magnitude characteristics for different order of the filter are shown in Fig. 55, 13. Give the equation for the order N, major, minor and axis of an ellipse in case f Chebyshev filter. The order N’ of Chebyshev filter is given by 1 fo cos h~ JOO, — 1 N= o% -1 Ss cosh a is passband Where ag ig stopband attenuation at stopband frequency 5 and o> SP ellipse tetuation at passband frequency 2p. The major and minor ax Biven by aN n/N -1/N “ 6 0, [eee “| Se ee 25.112. Digital Signal Processing 14 f1te? and e = V10@1 = 1 where j: 14, What are the parameters that can be obtained from the Chebysp, ey specification? (ay May’07) sat yshev filter specifications we can obtain th From the given Chebyshev A b © Params like the order of the filter N €, transition ratio k, and the poles ofthe Pia ‘ Give the expression for the location of poles and zeros of a Ch, eBYshey II filter The zeros of the Chebyshev type II filter are Jie L a pe uN | The poles of the Chebyshev type II filter are | tet iy k=l. N an a= k= we = acoso: Q = bsingds, k=1,. oe = 4 oe k aN 16. Give the expression for location of poles of a Chebyshev type I filter. The poles of the Chebyshev filter can be found by using the formula Sk = acosO + jbsing, k=1,2,...N where N is the order of the filter a7, @k-1) ; m= ot tN REA 2. N JN _ = v=, en) 1/N — ea, [2 +H * Qp = passband frequency bweelt+Vi¢e2. € = V10%107 —] py = passband attenuationInfinite Impulse Response Filters $.113 sve the expression for location of poles of normalized Butterworth filter. (AU ECE May’07) (AU EEE‘07) se ptes ofthe Butterworth fier is given by y is order of the filter. 18, Distinguish between Butterworth and Chebyshev (Type-t) filter 1. The magnitude response of Butterworth filter decreases monotonically as the frequency 2 increases from 0 10 5c, whereas the magnitude response of the Chebyshev filter exhibits ripple in the passband and monotonically decreasing in the stopband. ‘The transition band is more in Butterworth filter compared to Chebyshev filter. 3, The poles of the Butterworth filter lie on a circle whereas the poles of the Chebyshev filter lie on an ellipse. 4. For the same specifications the number of poles in Butterworth filter are more when compared to Chebyshev filter, ie., the order of the Chebyshev filter is less than that of Butterworth. 19, How one can design digital filters from analog filters? (2) Map the desired digital filter specifications into those for an equivalent ana- log filter. (b) Derive the analog transfer function for the analog prototype. (©) Transform the transfer function of the analog prototype into an equivalent digital filter transfer function. 2. Men ; 20. aon any two procedures for digitizing the transfer function of an analog ilter pe ‘wo important procedures for digitizing the transfer function of an analog Iter are ( “) Impulse invariance method (by ? Bilinear transformation method Ly { fe the properties that are maintained same in the transfer of analog ler ir into a digital filter?5.114 Digital Signal Processing plane should map into the unit circle in the, jQ-axis in the s- . : Dla, (a) The jQ-axi direct relationship between the two frequency ve Thus there will be a d oles in the two domains. ) rhe left half plane of the s-plane should map into the inside of circle in the 2-plane. Thus a stabl stable digital filter. 22. Why IIR filters do not have linear phase. (AU May 2055 Linear phase filter must have a system function that satisfies the condition H(z) = 42°" H(z") the Je analog filter will be eonvereg ’ where z~ represents a delay of N units of time. But if this is the case, for every pole inside the unit circle there'must be a pole outside the unit circle. Hence the filter would be unstable. Consequently a causal and stable IR fier cannot have linear phase. 23, What do you understand by backward difference? One of the simplest method for converting an analog filter into a digital filteris to approximate the differential equation by an equivalent differential equation “_ ula?) - y(n? - 7) T = ¥(n) = y(n ~1) T ‘The above equation is known as backward difference equation. 24, What isthe mapping procedure between s-plane and zeplane in the method of mapping of differentials? What is it's characterietiog? ‘The mapping procedure between i 7 : Fits _ cof mapping Of differentials is given by oo H(z) = H(s) jedagt ‘The above mapping has the following characteristics | (a) The left half of s-py, see z 1/2 in the z-plange ”* M4P® inside a circle of radius 1/2 centered! a (b) The right half of 5. 1/2 in the z-plane, dius Plane maps in to the region outside the circle of |2. 2, Infinite Impulse Response Filters 5.115 What is the matched 2-transformation? It is a method of converting an analog filter into an equivalent digital filter. Suppose that the system function of the analog filter is expressed in the factored form M i (s~ 2) TI (s - px) fe where {2} are the zeros and {p,} are the poles of the filter. Then the system function of the digital filter is M T] (@.-e#7z-1) H() == . TQ — em? 2) k=l where T is the sampling interval. What is meant by impulse invariant method of designing IIR filter? In this method of digitizing an analog filter, the impulse response of resulting digital filter is a sampled versior. of the impulse response of the analog filter. First we express the transfer function of analog filter in partial fraction form, ie, N Ha(s) = So —* faa °~ Pk Then the transfer function of digital filter can be obtained using the transfor- mation 1 1 S— Pk erat 21 N i =y ie, Hla) = 3 ore By impulse invariance method obtain the digital filter transfer function and the differential equation of analog filter H(s) =. 1 For H(s) = 5 h(t) =e h(nT) =e" nt) = Shem5116 Digital Signal Processing Th differential equation can be obtained from , Y(s)_ 1 , X(s) s+1 s¥(s) + ¥(s) = X(s) ‘Taking inverse Laplace transform on both sides we get d : y(t) + y(t) = a(t ult) Fuld) = 2(0 28. Obtain the impulse response of digital filter corresponding to an Analog fi le with impulse response ha(t) = 0.5¢-* and with a sampling rate fin using impulse invariance method. k, Given h,(t) = 0.5e7?* hg(nT) = 0.5e72"7 T=1 sec HG@)= 08 05 lre?zt 7 0.1352-1 29. Why impulse invariant method is rr 1e design oj od | i i ; is not preferred in the desig hie a 006) In impulse invari o PI ; invariance method, the mapping from s-plane to z-plane is me one, i.e., all the . oh 1)F (2k + 1) Poles in the s-plane between the intervals om © (for k = : , 9,1,2.... map into the entire z-plane, Thus, there is infinite nu . aliasing efec, tore that map to the same location inthe z-plane, pros 0 spectrum aliasing the impulse invariance method inappropriate for desipniny hi, 7 method is not preferred meas filters, That is why the impulse it i lesign of IIR filter other than lowposs 30. Give the bilinea transform equation between seplane and 2-plane 2fi-2 31. — realizable and stale AIR filters cannot have line ayelt rar phase: TreInfinite Impulse Response Filters 5.117 42, Obtain the digital fiter transfor ified. II poles of analog fil i aii = Ce Me ecified the system func as H(s) = © —* where tp, as HG) sap Te Pe partial fraction coefficients, Now into digital as Sunction if att potes f analog filter are spec ion can be written } are poles of the analog filter and {cj} are the above system function can be transformed Nv _ Tey H(z) = x T— eT yai Where T = sampling interval 33, What is bilinear transformation? “AUT, BME, 2011) The bilinear transformation is amapping that transforms the left half of s-plane into the unit circle in the z-plane only once, thus avoiding aliasing of frequency components. The mapping from the s-plane to the 2-plane in bilinear transformation is offs T [1427 34. What are the properties of the bilinear transformation? 1. The mapping for the bilinear transformation is a one-to-one mapping; that, is for every point z, there is exactly one corresponding point s, and vice versa. 2. The j-axis maps on to the unit circle |z| = 1, the left half of the s-plane maps to the interior of the unit circle |z| = 1 and the right half of the s-plane maps on to the exterior of the unit circle |z| = 1. 38. What is warping effect? What is its effect on magnitude and phase response? The relation between the analog and digital frequencies in bilinear transforma- tion is piven by 2 w Q= Ftan> For smaller values of w there exist linear relationship between w and ©. But for large values of w the relationship is non-linear. This non-linearity introduces’ distortion in the frequency axis. ‘This is known as warping effect. This effect “ompresses the magnitude and phase response at high frequences. 36. Wes : ' Write a short note on prewarping. can be com- The effect of the non-linear compression at high frequer Pensated. When the desired magnitude response is piece-wise constant over5.118 Digital Signal Processing pression can be compensated by introducing . y, this comy mn s : cf oe or prewarping the critical frequencies by using the Form tle prescaling, w tan = Q= 2 37. What are the advantages and disadvantages of bilinear Wransformations (AV 2H, Advantages 1. The bilinear transformation provides one-to-one mapping. 2. Stable continuous systems can be mapped into tealizable, Stable igi systems. 3. There is no aliasing. Disadvantages 1. The mapping is highly non-linear producing frequency compression 2 high frequencies. 2. Neither the impulse response nor the phase response of the analog fier is preserved in a digital filter obtained by bilinear transformation, 38. Distinguish between recursive realization and non-recursive realization, (MU Apr’ 96) For recursive realization the present output y(n) is a function of Past outputs, Past and present inputs, This form corresponds to an Infinite-Impulse ‘Response UR) digital filter. For non-recursive realizations the current output y(n) is a function of| only past and present inputs. This form corresponds to an Finite Impulse Response (FIR) digital filter. 39. What are the different types of structures for realization of IIR systems? ‘The different types of structures for Tealization of IIR system are - (i) Direct-form I structure, (ii) Direct-form II structure (iii) Transposed direct- form II structure, (iv) Cascade form structure, (v) Parallel form structure (vi) Lattice-Jadder structure, 40. How many number of additions, multiplications and memory locations art required to realize a system H(z) having M zeros ond N poles in (a) direct Sorm I realization (b) direct-form 11 realization? (2) The direct-form I realization requires M + Né1 multiplications, M+N ad tions and M+ N+1 memory locations, (b) The direct-form If realization equies M4 Ne multiplications, MN ditions and the maximum of (M,N) memory locationsInfinite Impulse Response Filters. 5.119 41. Draw the direct-form I realization structure of a 3rd order syst system - - (AUT 2011), (MU Oct’ 95, Oct” 98) a pe to) ; 7] + a xo — by 4. Draw the general realization structure in direct-form I of IR system. x(n) by vi) MU Apr’ 96) 43. Give direct-form I and direct-form II structure of second order system real- ization, (MU Oct’ 99, Apr’ 2000, Oct’ 2000, Oct’ 96) Direct-form I realization of a second order system 6 x(n) —> yin) ‘ he ner) ua —— Power) j5.120 Digital Signal Processing Direct-form I realization of a second order system be x(n) 44. What is the main advantage of direct-form II realization when compared to direct-form I realization? In direct-form II realization, the number of memory locations required is less than that of direct-form I realization. 45. Draw the cascade structure direct-form II realization of 6th order filter. ya) 46. Define signal flow graph. _ A signal flow graph is a graphical representation of the relationship between the variables of a set of linear difference equations. 47. Draw the signal flow graph of first order digital filter. x(0) be yo) 48. What is transposition theorem and transposed structure? (Oct’ 95) The transpose of a structure is defined by the following operations. (i) Reverse the directions of all branches in the signal flowgraph. (ii) Interchange the input and outputs. (iii) Reverse the roles of all nodes in the flow graph. (iv) Summing points become branching points. (v) Branching points become summing points, 4Infinite Impulse Response Filters 5.121 ‘according t0 transposition theorem if we r. transmittance and interchange the input and fanction remains unchanged. verse the directions of all branch Output in the flowgraph, the system 49. Give the transposed direct-form II structure of ITR second order system. xin) be by 50, Draw the parallel form structure of IR filter. A parallel form realization of an IIR system can be obtained by performing a partial fraction expansion of H(z). N H@)=c+>> ps where {p,} are the poles =C+ (z)+Ho(z) +... Ay(2) ‘The above system can be realized in ~arallel form as shown in Figure. c x(n) | Hy(2) na POX” 51. What is the main disadvantage of direct-form realization? The direct-form realization is extremely sensitive to Parameter uantization When the order of the system 1V is large, a small change in a Ber 7 ae due to parameter quantization, results in a large change in the locatior Poles and zeros of the system. F s 1 in cas rm network. 2, Realize vn) + y(n +1) + gun -2)= a(n) in cascade for vv os) Given u(n) + y(n — 1) + Futon — 2) = a(n).5.122 Digital Signal Processing Taking z-transform on both sides ¥(z) [ tote i] ¥(2) _ 1 H@)=y@) "ieee! 1 =I a2) 2 (1452 JO 3 = Hite) ate) where Hy (z) = Ho(z) = xin), 53. What is the advantage of cascade realization? Quatization errors can be minimized if we realize an LTI system in cascade form. 54, Distinguish between first order and second order filters. The first order filter has a single pole, whereas the second order filter has two poles. The frequency response of second order filter contain ripples, whereas the fre~ quency response of first order filter does not contain ripples. ‘The frequency response of second order filter is better than the frequency of first order filter. 55. Find the digital transfer function H(z) by sing impulse invariant method for the analog transfer function H(s) = Assume T = 0.5 sec. #@)= 4, = —L_- 1 _ 1—eW2z-1 etz-T ~ 1—0.3682-1 56. Convert the analog bandpass filter Hels) 5 /(s + 0.1)? 4-9 in to a digital IIR filter by the use of the mappins 8= praInfinite Impulse Response Filters 5.123 H,(s)=—_1___ (s+0.1)? +9 H(iz)=—1__| (s+01? 49,4 i Ter 1 =—_+ __ [> (3) +0] +9 57. Transform the single pole lowpass Butterworth filter with system function H(s) = Qp/s + Op into a bandpass filter with upper and lower band edge frequencies Qu, and Q, respectively, H(s)= mt 1 5240. stm) t 1 (Mu = 2)s__ SE + (Oy — Ms + UMW- “ilters 6.13 Finite Impulse Redpdnse Pa uestions and Answers used 0 7 the different type: 1, What are Pes 1. IR filter 2. FIR filter The IIR filters are of recursive type, pends on the present input, past input The FIR filters are of nonrecursive depends on the present input sample whereby the Present output sample de- Samples and output samples. ype whereby the present output saruple and previous input samples, 2. What are the different types of filters based on frequency response? The filters ean be classified based on frequency response, ‘They are (i) lowpass filter (ii) highpass filter (iii) Bandpass filter (iv) Bandreject filter, 3, What is the most general form of IIR filter? The most general form of IIR filter can be written as M —k H(z) Zeca 1+ Shi apem* where at least one of the a; is nonzero and all the roots of the denominator are not cancelled by the roots of the numerator. 4, What are the desirable and undesirable features of FIR filters? (AU May 2006),(Oct’98) OR What are the advantages and disadvantages of FIR filters? Advantages 1. FIR filters have exact linear phase. ~ 2. FR filters are always stable. i 3. FIR filters can be realized in both recursive and nonrecursive emt 4. Filters with any arbitrary magnitude response can be tackled using sequency, ‘ Disadvantages AR fi ssign can be as 1. For the same filter specifications the onfer of FIR fiter desig high as 5 to 10 times that of an IIR design. 2. Large storage requirements needed. i implementation. 3. Powerful computational facilities required for the imp6.130 Digital Signal Processing Ss. 6. 8. 9. 10. Distinguish between FIR and IIR filters. FIR filter OR filter 1. | These filters can be casily de- | These filters do not have jin signed to have perfectly linear | phase. ‘al (Oer99) phase. 2, | FIR filters can be realized recur- | IIR filters are easily realized re. sively and non-recursively. cursively. 3. | Greater flexibility to control the | Less flexibility, usually limited shape of their magnitude re- | to specific kind of filters. sponse. 4. | Errors due to roundoff noise are | The round off noise in IIR filters less severe in FIR filters, mainly | are more. because feedback is not used. What are the techniges of designing FIR filters? (Oct’95, Apr’'98) There are three well-known methods for designing FIR filters with linear phase, ‘These are (1) windows method (2) frequency sampling method (3) optimal or minimax design. : What do you understand by linear phase response? (Apr’ 96) For a linear phase filter 6(w) «x w, the linear phase filter does not alter the shape of the original signal. If the phase response of the filter is nonlinear the ” output signal may be distorted one. In many cases a linear phase characteristic is required throughout the passband of the filter to preserve the shape of a given signal within the passband. An DR filter cannot produce a linear phase. The FIR filter can give linear phase, when the impulse response of the filters symmetric about its mid-point. Suppose the axis of symmetry of impulse response h(n) lies half way between two samples, for what kind of applications this type of impulse response used. If the axis of symmetry lies midway between two samples, such type of se- quences can be used to design Hilbert transformers and differentiators. For what kind of applications, the antisymmetrical impulse response 2" be used? : ‘The antisymmetrical impulse response can be used to desi ers and differentiators. ign Hilbert transform sed? For what kind of application, the symmetrical impulse response can be ber of samp! bandpass 1g ca ‘The impulse response, which is symmetric having odd num is be used to design all types of filters, i. lowp highpass, bandreject.a se Finite Impulse Response Filters 6.131 metric impulse response having even number of s; The SYN pass and bandpass filter. ‘amples can be us to design i ;, the condition for the impulse response o, Satisfy for i the con Is of FIR filter to sati fy yl group and phase delay and for only constant group lelay? ee at is delay and. i) ” delay? * 98) For li delay inear phase FIR filter to have both constant group delay and constant phase 8(w)=-ow -a =0 otherwise 3. Find the transfer function of the realizable filter (N-1)/2 H(2)= 2-9? |rO)+ DY A(n)(e" +2) n=0 32. What are the desirable characteristics of the window? ‘The desirable characteristics of the window are 1. The central lobe of the frequency response of the window should contain most of the energy and should be narrow, The highest side lobe level of the frequency response should be small. x 3. The side lobes of the frequency response should decrease in energy rapidly as w tends to 7, 33. What is the principle of designing FIR filter using windows? (oer?) One possible way of obtainin atn= + (51) where Mg FIR filter is to truncate the infinite Fourier series ication She Fen nah ofthe desired sequence, But band. These oath niet Seties results in oscillation in the passband and sto? 1 ese oscillations are due to slow convergence of the Fourier series t reduce thes i ., © 05 tions the Fourier coefficients of the filter are modified % nite impluse responsi i ce w(") c se by a finite wei equence called a window, where P Y a finite weighing sequ w(n) = w(~n) £0 for |n| < x =0 0 for |n| >6 5. Draw the frequency response of N-point rectangular window. ‘The frequency response of the rectangular window is given by "te Impulse Response Filters 6.137 ater ltplying window sequence wn) by hu(n), we pet Eaquonce h(n) that satisfies the desired m y a finite duration agnitude response A(n) = ha(n)w(n) for all |nj < N=1 2 =0 for |nj > N=1 What is the need for employing window technique for FIR filter design? (Apr’ 99) OR What is window and why it is necessary? One possible way of finding an FIR filter that approximates (e%”) would be totruncate the infinite Fourier series atn = + (4). Abrupt truncation of the series will lead to oscillations in the passband and stopband. These oscillations can be reduced through the use of less abrupt truncation of the Fourier series, This can be achieved by multiplying the infinite impulse response by a finite weighing sequence w(n), called a window. : (Oct’95) in oN Wale) = ng where NV is number of samples. The frequency response is shown in Figure. we’) = ° * “2yiN 2x8 Give the equation specifying Hanning and Blackman windows. (Oct. 97) Hanning window: The equation for Hanning window is given by arn for -(N-1/2 > in FIR filter? “ao ent tcp ‘The necessary and sufficient Condition for jj y'07) filter is, the impulse response h(n) of the anes PM property, ie., yystem should =h(n) = AUN ~1~ ny where N is the duration of the sequence, Characteristic et 96, Oc'2900) characteristic in FIR have the symmetical 42. Give the equation specifying Kaiser window, The Kaiser window is given by z wp(n) = Ip _f- (sy N-1 Tela’ foros “= =0 otherwise where a is an independent parameter Jp(cz) is the zeroth order Bessel function of the first kind oe h(a) =1+ > +@)] k=l 43, What are the advantages of Kaiser window? 10 select the side lobe level and N. 1. It provides flexibility for the designer t ; 2. Ithas the attractive property that the side lobe level can be varied Cars uously from the low value in the Blackman window to the high val the rectangular window. ‘i ing method? 4 Whats the principle of designing FIRfer wing frequency PEN" ho is sampled and i jtude response is samp! In frequency sampling method the desired eee desired frequency response i i es i alinear phi is specified. The samp i" determined a nee noe = eee The filter coefficients are then 48 the IDFT of this set of samples.| ae Ps Signal Processing yy For what type of filters freq method is attractive for narrow ban ‘Frequency sampling and fy amples of the fi frequen filters where arly few of the SemP Tequency responses? nn ancy sampling method ig suitable 2 mn of the frequency sampling realization o of FIR Filter 46. Give the equatio’ 9) ‘The 2-Transform of FIR filter designed using frequency samp sf 5 on m givenby my i" et hog & Gaz) where G2) = ate Wa eitn/n ‘This can be realized as shown in Figure. a ae + Gaz) ¥(0) Ge G.@) 47. it Show that the filter with hn] = {—1,0,1} is a linear phase filter H(e*) = x (neni From Ea) we can find Which is the 8(w) = —w Proportiona ‘ow. Hence the filter h(n) is a linear phase filetFinite Word Length Effects in Digital Filters’ 7.59 gions and Answers we . . 0 iat are the different types of arithmetic in digital systems? re are three types of arithmetic used in i metic, 2+ Floating point arithmetic, 1. igital systems 1. Fixed point arith- 3. Block Floating arithmetic. what do you understand by a fixed-point number? (MU Oct’ 95) In fixed-point arithmetic the position of the binary point is fixed, The bit to the sight represent the fractional part of the number and those to the left represent the integer part. For example, the binary number 01.1100 has the value 1.75 in decimal. What are the different types in fixed-point number representation? Depending on the way negative numbers are represented, there are three differ- ent forms of fixed-point arithmetic, ‘They are 1) sign-magnitude, 2) 1’s com- plement, 3) 2’s complement, - |. What do you understand by sign-magnitude representation? For sign-magnitude representation the leading binary digit is used to represent the sign. If it is equal to 1 the number is negative, otherwise the number is positive. For example, the decimal number ~1.75 is represented as 14.110000 and 1.75 is represented as 01.110000. In sign-magnitude form the number 0 has two representations, i.e., 00.000000 or 10.000000. With b bits only 2° — 1 numbers can be represented. Write a short note on 1’s complement representation? In one’s complement form the positive number is represented as in the sign- magnitude system, To obtain the negative of a positive number, one simply complements all the bits of the positive number. For example, the negative of 01.116000 would be represented as ~(01.110000)2 = (10.001111)2 ‘The number 0 has two representations i.e., 00.000000 and 11.111111in a 1's complement representation. . What do you understand by 2’s complement representation? Intwo’s complement representation positive numbers are represented as in sign magnitude and one’s complement. The negative number is obtained by com- Plementing all the bits of the positive number and adding one to the least sig- nificant bit. For example (0.5625)10 = (0-100100)2 (—0.5625)i0 = 1.011011 0.000001