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Question 7 ( Exam June-July 2009) Define DFT. Establish a relation between the Fourier series coefficients of a continuous time signal and DFT Solution The DTFT representation for a finite duration sequence is X (j) = x (n) -jn n= - X (n) =1/2 X (j) e jn d , Where 2k/n 2 Where x(n) is a finite duration sequence, X(j) is periodic with period 2.It is convenient sample X(j) with a sampling frequency equal an integer multiple of its period =m that is taking N uniformly spaced samples between 0 and 2. Let k= 2k/n, 0kN Therefore X(j) = x(n) -j2kn/N n= Since X(j) is sampled for one period and there are N samples X(j) can be expressed as N-1 SJBIT/ECE Page 4
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Question 2 ( Exam Dec 08-Jan 09, Dec 07-Jan 08, ) State and Prove the: (i) Circular convolution property of DFT; (ii) DFT of Real and even sequence. Solution (i) Convolution theorem Circular convolution in time domain corresponds to multiplication of the DFTs If y(n) = x(n) h(n) then Y(k) = X(k) H(k) Ex let x(n) = 1,2,2,1 and h(n) = 1,2,2,1 Then y (n) = x(n) h(n) Y(n) = 9,10,9,8 N pt DFTs of 2 real sequences can be found using a single DFT If g(n) & h(n) are two sequences then let x(n) = g(n) +j h(n) G(k) = (X(k) + X*(k)) H(k) = 1/2j (X(K) +X*(k)) 2N pt DFT of a real sequence using a single N pt DFT Let x(n) be a real sequence of length 2N with y(n) and g(n) denoting its N pt DFT Let y(n) = x(2n) and g(2n+1) k SJBIT/ECE Page 9
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Question 3 ( Exam Dec 08-Jan 09) Distinguish between circular and linear convolution Solution 1) Circular convolution is used for periodic and finite signals while linear convolution is used for aperiodic and infinite signals. 2) In linear convolution we convolved one signal with another signal where as in circular convolution the same convolution is done but in circular pattern depending upon the samples of the signal 3) Shifts are linear in linear in linear convolution, whereas it is circular in circular convolution. Question 4 ( Exam June july 2009) State and prove the following properties of the DFT i) Symmetric property for real valued sequence x(n) ii) Parsevals theorem Solution
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Question 17 ( Exam Jan Feb 08) Using overlap-save method, compute yen), of a FIR filter with impulse response H(n) = (3, 2, 1) and input x(n) = (2, 1, -1, -2, -3, 5, 6, -1, 2, 0, 2, 1J. Use only 8-point circular convolution in your approach.
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Question 15( Exam June July 08) Show that the impulse invariant transformation is not one-to-one form s-plane to Z-plane
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Here {pk} are the poles of the analog filter and {Ck } are the coefficients of partial fraction expansion.
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Tabulate the number of complex multiplication and complex additions required for the direct computation of DFT and FFT algorithm for N = 16, 32, 128.
Question 17( Exam Jan/Feb 08) Compute circular - convolution using DFT and IDFT formulae for the following sequences, xl(n) = n and x2(n) = Cos(n/2) for 0<n<3.
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Hence the circular convolution of x1(n) and x2(n) can be obtained by taking the inverse IDFT of the product X1(k) . X2(k)
i) To obtain the DFT of X1(k): The signal flow graph for calculation of DFT is shown in Fig (ii) Calculations at intermediate steps:From above signal flow graph, the DFT is,
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Question 19( Exam Jan/Feb 08) Derive the signal flow graph for 8-point Radix-2 DIT-FFT algorithm. Sol:Let f1 (n) contain even numbered samples of x(n) and f2(n) contain ordered samples of x(n). Thus we can write,
Here time domain sequence x (n) is splitted into two sequences. This splitting action is called decimation. Since it is done on time domain sequence it is called decimation in Time (DIT). We know that N-point DFT is given as,
Since the sequence x(n) is splitted into even numbered and odd numbered samples, above equation can be written as,
From equation (1) we know that f1 (m) = x(2m) and f2(m) = x(2m + 1), hence above equation becomes,
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In the above equation we have rearranged WN factors. We know that (WN)2 = WN/2 Hence above equation can be written as,
Comparing above equation with the definition of DFT of equation (2), we find that first summation represent N/2 point DFT of f1(m) and second summation representsN/2 point DFT of f2(m)
Thus F1(k) is N - point DFT of f1(m) and F2(k) is N point-DFT of f2(m). Since F1(k) and F2(k) are then N/2 - point DFTs, they are periodic with period N/2. i.e.,
From equation (5) we know that WNK+N/2 = - WNK, and from equation (5) we can write above equation is,
Here observe that X (k) is N-point DFT. By taking k = 0 to N/2 -1 we can calculate F1(k) and F2 (k) since they are N/2 - point OPTs. The N-point DFT X (k) can be combinely obtained from equation (4) and equation (6) above by taking k = 0 to N/2-1.
The above two equations show that N-point DFT can be obtained by two N/2-point DFTs. SJBIT/ECE Page 50
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Fig. 2 shows that 8-point DFT can be computed directly and hence no reduction in computation. It is shown symbolically as a single block which computes 8-point DFT directly. Fig. 3 shows the symbolic diagram for this creation.
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Question 3 ( Exam Jan Feb 05) Starting from low pass butter worth prototype analog filter, design a fourth order Butterworth band pass analog filter with upper and lower band edge frequencies 10 rad/sec and 5 rad/sec.
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Question 5 ( Exam Jan Feb 05) Using the impulse response technique design a low pass digital filter that is equiripple in the passband, and monotone in the stopband. The filter passband edge is at 0.1 rad and with a ripple of 2.5 dB or less and the stopband edge at 0.2 rad with attenuation of 40 dB or more. Use T =1.
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Question 6( Exam Jan Feb 05) Transform the analog filter H(s) =1/s+a, a> 0 to a digital filter using the back-ward difference mapping. Comment on the stability of the digital filter.
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Question 7 ( Exam June July 08) An analog Chebyshev low pass filter with pass band attenuation of 2.5 dB at 200 Hz, stop band attenuation of 30 dB at 500 Hz is required. Find the order of filter. Obtain the system transfer function H(s). Sketch the magnitude response of the filter.
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iii) To obtain the magnitude response The magnitude response can be obtained by putting s = j, in Ha(s). i.e.
Question 8 ( Exam June July 08) Obtain the transfer function of the analog Butterworth low pass filter of third order with cut-off frequency of 10 Hz. Find the transfer function of 3rd order high pass Butterworth filter with 100 Hz cut-off frequency by frequency transformation of low pass filter. Ans : Given N = 3, Fe = 10 Hz (Low pass) N = 3, Fe = 100 Hz (High pass)
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Question 11 ( Exam Jan Feb 08) Design a digital IIR low pass Butterworth filter that has a 2 dB pass band attenuation at a frequency of 300n rad/sec and at least 60 dB stop band attenuation at 4500 rad/sec. Use backward difference transformation.
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This is the required system function of the digital filter. Since it is single pole function it can be realized using direct form.
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Question 2( Exam Jan Feb 09) b) What is Gibbs phenomenon? How it can be reduced?
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Question 3( Exam Jan Feb 09) c) Show that the roots of H (z) occur in reciprocal pair for a linear phase FIR filter.
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Putting for k = 0 in equation (4), we get H (0) = 1 and with M = 17, above equation becomes,
UNIT 7
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Show that the impulse invariant transformation is not one-to-one form s-plane to Question 2 ( Exam June July 09)
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UNIT 8 Implementation of Discrete-Time Systems Question 1 ( Exam Dec 10 Jan 11) Consider a FIR filter with system function: H(z) = 1+2.82 Z-1 +3.4048z-2 +1.74z- 3. Sketch the direct form and lattice realizations of the filter. SJBIT/ECE Page 88
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Question 2 ( Exam Jan Feb 09) For y(n) = -0.1y(n-1)+0.2y(n-2) +3x(n) +3.6x(n-1)+0.6x(n-2), obtain form I and II, cascade form and parallel form with single pole-zero subsystems.
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