Cs2403-Digital Signal Processing

CS2403 DIGITAL SIGNAL PROCESSING
A.R.ENGINEERING COLLEGE- VILLUPURAM DEPARTMENRT OF COMPUTER SCIENCE ENGINEERING QUESTION BANK SUB.NAME: DIGITAL SIGNAL PROCESSING SUB.CODE: CS2403 YEAR : IV BRANCH SEM : CSE : VII
UNIT-I SIGNAL AND SYSTEMS PART-A 1. Define Signal. 2. Define a system. 3. What are the advantages of DSP? 4. Give some applications of DSP 5. What are the various methods of representing discrete time signal? 6. Define the impulse and unit step signal. 7. How will you classify the discrete time signals? 8. When a discrete time signal is called periodic? 9. What are even and odd signals? 10. What is discrete time system? 11. What is impulse response and what is its and its significance? 12. Define the transfer function of an LTI system. 13. What are the various methods available to determine the response of LTI system? 14. Write few properties of discrete convolution. 15. List the various methods of classifying discrete system. 16. What is a static and dynamic system? Give examples. 17. Define time invariant system. 18. What is linear and nonlinear system? 19. What is casual and non-casual system? 20. What is importance of causality? 21. What is BIBO stability? What is the condition to be satisfied for stability?
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22. List any four properties of Fourier transform. 23. What is relation between Fourier transform and z- transform? 24. What is frequency response of LTI system? 25. What are the steps involved in digital signal processing? Or draw the general block diagram to show the schematic representation of a DSP system. 26. Define linear convolution and correlation. 27. Distinguish between linear convolution and circular convolution. 28. Define z-transform pair. 29. Define region of convergence. 30. How the stability is determined for a system in terms of z-transform? 31. State persavels relation. 32. State initial and final value theorem. 33. What are the different methods of evaluating inverse z-transform? 34. What are the standard discrete-time signals? 35. What are the operations involved in convolution? PART-B 1. Determine whether the following system are linear, time- invariant. i) y(n) = A x(n) +B ii) y(n) =x (2n) iii) y(n) =n x2 (n) iv) y(n) = a x(n) 2. Check for following systems are linear, causal, time in variant, stable, static. i) y(n) =x (2n) ii) y(n) = cos (x(n)) iii) y(n) = x(n) cos (x(n) iv) y(n) =x (-n+2) v) y(n) =x(n) +n x (n+1) 3.a) For each impulse response determine the system is i) stable ii) causal. i) h(n)= sin ( n / 2) ii) h(n) = (n) + sin n
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iii) h(n) = 2 n u(-n) b) Find the periodicity of the signal x (n) =sin (2n / 3) + cos ( n / 2) 4. Explain in detail about A to D conversion with suitable block diagram and to reconstruct the signal. 5. (i) State and proof of sampling theorem. (ii)What are the advantages of DSP over analog signal processing? 6. (i) Explain successive approximation techniques. (ii)Explain the sample and hold circuit. 7. (i) State and proof the properties of Z transform. (ii)) Find the Z transform of a) x(n) = [(1/2)n (1/4)n] u(n) b) x(n) = n (-1)n u(n) c) x(n) (-1)n cos (n/3) u(n) d) x (n) = () n-5 u(n-2) +8(n-5) 8. (i) Find the Z transform of the following sequence and ROC and sketch the pole zero diagram a) x(n) = an u(n) +b n u(n) + c n u(-n-1) , |a| <|b| <| c| b) x(n) =n2 an u(n) (ii) Find the convolution of using z transform (a) x1(n) ={ (1/3) n, n>=0 (b) (1/2) - n n<0} (c) x2(n) = (1/2) n 9. Find the inverse z transform X(z) = log (1-2z) z < |1/2 | X(z) = log (1+az-1) |z| > |a| X(z) =1/1+az-1 where a is a constant X(z)=z2/(z-1)(z-2)
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X(z) =1/ (1- z-1) (1-z-1)2 X(z)= Z+0.2/(Z+0.5)(Z-1) Z>1 using long division method. X(z) =1- 11/4 z-1 / 1-1/9 z-2 using residue method. X(z) =1- 11/4 z-1 / 1-1/9 z-2 using convolution method. 10. A causal LTI system has impulse response h(n) for which Z transform is given by H(z)= 1+ z -1 / (1-1/2 z -1 ) (1+1/4 z -1 ) i) What is the ROC of H (z)? Is the system stable? ii) Find THE Z transform X(z) of an input x(n) that will produce the output y(n) = - 1/3(-1/4)n u(n)- 4/3 iii) Find the impulse response h(n) of the system. 11. (i)The impulse response of LTI system is h(n)=(1,2,1,-1).Find the response of the system to the input x(n)=(2,1,0,2) (ii). Determine the response of the causal system y(n) y(n-1) =x(n) + x(n-1) to inputs x(n)=u(n) and x(n) =2 n u(n).Test its stability 12. Determine the magnitude and phase response of the given equation y(n) =x(n)+x(n-2) 13. (i)Determine the frequency response for the system given by y(n)-y3/4y(n-1)+1/8 y(n-2) = x(n)- x(n-1) (ii)Determine the pole and zero plot for the system described difference equations y(n)=x(n)+2x(n-1)-4x(n-2)+x(n-3) 14. Find the output of the system whose input- output is related by the difference equation y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the step input. 15. Find the output of the system whose input- output is related by the difference equation y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the x(n) =4 n u(n). 16. Find the output of an LTI system if the input is x(n) =(n+2) for 0 n 3 and h(n) =an u(n) for all n 17. Find the convolution sum of x(n) =1 n = -2,0,1
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= 2 n= -1 = 0 elsewhere and h(n) = (n) (n-1) + ( n-2) - (n-3) 18. (i) perform circular convolution of the two sequences x1 (n) ={2,1,2,1) and x2 (n) ={1, 2 ,3,4} (ii)Find the linear and circular convolution of the sequences x (n) = {1, 0.5} and h (n) = {0.5, 1} 19. (i) The input x(n) and impulse response h(n) of a LTI system are given by, x(n) = {-1,1,2,-2} ; h(n) = {0.5,1,2,0.75} Determine the response of the system a) using linear convolution b) using circular convolution. (ii) Find correlation of the following sequences. Cross correlation x (n) = {1, 1, 2, 3, 4}, y (n) = {1, 1, 2, 1} (8) (8) (16) (8) (8)
Circular correlation x (n) = {1, 2, 3, 4}, h (n) = {1, 2, 2, 1} Auto correlation y (n) = {1, 2, 3, 4} UNIT II FREQENCY TRANSFORMATIONS PART-A 1. Define DFT of a discrete time sequence. 2. Define IDFT. 3. What is the relation between DTFT and DFT? 4. What is the draw back in Fourier transform and how it is overcome? 5. Give any two applications of DFT (or mention the importance of DFT). 6. State the properties of DFT. 7. When an N-point periodic sequence is said to be even or odd sequence? 8. What is relation between Z-transform and DFT? 9. What is zero padding? Why it is needed? 10. Why circular convolution is important in DSP? 11. How will yo u pe r for m linear co nvo lutio n via c irc ular convolution? 12. What is sectioned convolution? 13. What are the two methods used for the sectional convolution? 14. What is overlap-add method? 15. What is overlap-save method?
16. What are differences between overlap-save and overlap-add methods. 17. What way zero padding is implemented in overlap save method? 18. What is FFT? 19. Why FFT is needed? 20. What is radix-2 FFT? 21. How ma ny multip licatio ns a nd add itio ns are r eq uired to co mp ute N po int DFT using radix-2 FFT? 22. What is DIT algorithm? 23. What DIF algorithm? 24. What are the applications of FFT algorithm? 25. Why the computations in FFT algorithm is said to be in place? 26. What are the differences and similarities between DIF and DIT algorithms? 27. What is phase factor or twiddle factor? 28. Draw and explain the basic butterfly diagram or flow graph of DIT radix-2 FFT. 29. What is DIT radix-2 FFT? 30. Draw and explain the basic butterfly diagram or flow graph of DIF radix-2 FFT. 31. Compare the DIT and DIF radix-2 FFT. 32. What is direct or slow convolution and fast convolution? PART-B 1. (i) compute 4-point DFT of casual three sample sequence given by x (n) = 1/3 ; =0 0n2 (8)
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(ii) Compute the DFT of the sequence x (n) = {0, 1, 2, 3}.sketch the magnitude and phase spectrum. (8) 2.Compute the DFT for the sequence.(1,1,1,1,1,1,0,0) 3. Find the DFT of the sequence x (n) 1 x (n)= for 0n2 for the total of 8 number of sequences. (16) (16) (16)
0 otherwise 4.Find the DFT of a sequence x(n)=(1,1,0,0) and find IDFT of Y(k) =(1,0,1,0) 5. Perform the linear convolution of the following sequences by (i) overlap add method and (ii) overlap save method. x (n)= {1,-1,2,-2,3,-3,4,-4}; h(n) = {1,-1} Also sketch the output sequence. 6. Find the output y (n) of a filter whose impulse response is h (n) = {1, 1, 1} and input
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signal x(n) = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1} using Overlap add and Overlap save method. 7. By means of DFT and IDFT, determine the response of an FIR filter with impulse response h(n)= {1,2,3} to the input sequence x(n) = {1,2,2,1}. 8. Derive and draw the 8 point FFT-DIT butterfly structure. 9. Derive and draw the 8 point FFT-DIF butterfly structure. 10.Find the 8 point DFT of the given sequence x(n) =(0,1,2,3,4,5,6,7,) using DIF,radix-2,FFT algorithm. 11. From the first principles obtain the signal flow graph for computing 8 point DFT using radix-2 DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute its 8point DFT of x(n) by radix-2 DIF-FFT. 12.Compute the eight point DFT for the sequence x(n)= {0.5,0.5,0.5,0.5,0,0,0,0} using the inplace Radix-2 DIT algorithm. 13. Find the 8-point DFT of the sequence 1, x (n)= 0, otherwise 0n7 using decimation in-time FFT algorithm.
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14. x (n) = sin (n/2) at N=8. Find X(k) using DIT FFT algorithm. 15. From the first principles obtain the signal flow graph for computing 8 point DFT using radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x (n) = 2 sin n / 4 for 0 n 7 16. Given x (n) =2n and N=8, find X (k) using DIT FFT algorithm. 17. Compute the FFT for the sequence x (n) = n+1 where N=8 using the inplace radix 2 decimation In frequency algorithm. 18. Determine the response of LTI system when the input sequence is x (n) = {-1, 1, 2, 1,-1} by radix 2 DIT FFT. The impulse response of the system is h (n) = {-1, 1,-1, 1}. 19. Find the DFT of a sequence x(n)={1,2,3,4,4,3,2,1} using DIT algorithm. 20. (i) Discuss the properties of DFT. (ii) Discuss the use of FFT algorithm in linear filtering. 21. An LTI system has the input x (n) = {1, 1, 1} and the impulse response h (n) = {-1, -1} Determine the response of LTI system by radix -2 DIT FFT. UNIT- III IIR FILTER DESIGN PART-A
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1. Define an IIR filter. 2. Distinguish between IIR and FIR filters. 3. Compare IIR and FIR filters. 4. Classify the filters based on frequency response. 5. What are the properties that are maintained same in the transformation of analog to digital filter? 6. What are the requirements for an analog filter to be stable and causal? 7. What are the requirements for a digital filter to be stable and causal? 8. Define ripples in a filter. 9. Write a brief note on the design of IIR filter. (Or how a digital IIRfilter is designed?) 10. Mention any two techniques for digitizing the transfer function of an analog filter. 11. Compare the digital and analog filter. 12. What are the advantages and disadvantages of digital filters? 13. Mention the important features of IIR filters. 14. What is impulse invariant transformation? 15. How analog poles are mapped to digital poles in impulse invariant transformation (or in bilinear transformation)? 16. What is the relation between digital and analog frequency in impulse invariant transformation? 17. What is aliasing? 18. What is aliasing problem in impulse invariant method of designing digital filters? Why it is absent in bilinear transformation? 19. What is bilinear transformation? 20. What is the relation between digital and analog frequency in bilinear transformation? 21. How analog fr eq ue nc y is mapped to d igita l freq ue nc y in b iline ar tra ns fo r ma tio n? 22. Wha t is freq ue nc y warp ing? 23. What are the advantages & disadvantages of bilinear transformation? 24. What is prewarping? Why it is employed? 25. Explain the technique of prewarping. 26. Compare impulse invariant and bilinear transformations. 27. What is butterworth approximation? 28. Write the properties of Butterworth filter. 29. What is Chebyshev approximation? 30. Give the properties of Chebyshev filter. 31. State the structure of IIR filter
32. State the advantage of direct form structure over direct form structure. 33. What do you understand by backward difference? 34. What are the properties of bilinear transformation? PART-B 1. Obtain the cascade and parallel form realizations for the following systems Y (n) = -0.1(n-1) + 0.2 y (n-2) + 3x (n) +3.6 x (n-1) +0.6 x (n-2) 2. (i) Obtain the Direct form II y (n) = -0.1y(n-1) + 0.72 y(n-2) + 0.7x(n) -0.252 x(n-2) (ii) Find the direct form II H (z) =8z-2+5z-1+1 / 7z-3+8z-2+1 3. Obtain the i) Direct forms ii) cascade iii) parallel form realizations for the following systems y (n) = 3/4(n-1) 1/8 y(n-2) + x(n) +1/3 x(n-1) 4. Find the direct form I, cascade and parallel form for H(Z) = z -1 -1 / 1 0.5 z-1+0.06 z-2 5. Explain the method of design of IIR filters using bilinear transform method. 6. (i) For the analog transfer function Ha(s) = 2 / (s+1) (s+3), determine H (z) if (a) T=1 sec and (b) T=0.1 sec (ii) ) Convert the analog filter with system transfer function Ha(s) = (s+0.1) / (s+0.1)2 + 9 Into a digital IIR filer means of the impulse invariant method. 7. (i) apply the bilinear transformation to Ha(s) = 2 / (s+1) (s+2) With T=1 sec and find H (z). (ii) Convert the analog filter with system function Ha(s) = into digital filter using bilinear Transformation, Ha (s) = (s+3)/(s+0.3)2 + 16 (8) (8) (8) (8) (16) (16) (16) (8) (8) (16)
8. (i)The normalized transfer function of an analog filter is given by H a (sn ) = 1/ sn 2 +1.414 s n +1. Convert analog filter to digital filter with cut off frequency of 0.4 using bilinear transformation. (ii) Ha (s) = 1/(s+0.1)2 + 9 convert the analog BPF into digital IIR filter using backward difference for the derivate. 9. (i) Design a single pole low pass digital IIR filter with -3db bandwidth of 0.2 by using bilinear transformation. (ii) Design a Chebyshev filter with a maximum pass band attenuation of 2.5 dB; at p = 20 rad/sec and the stop band attenuation of 30 dB at s = 20 rad/sec.
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10. For the constraints 0.8 |H (e j )| 1.0 |H (e j )| 0.2 ; 0 0.2 ;0.6 with T= 1 sec . (16)
Design a Butterworth digital filter using impulse invariant transformation. 11. The specification of the desired lowpass filter is 0.8 |H (e j )| 1.0 |H (e j )| 0.2 ; 0 0.2 ; 0.6
Design a Chebyshev digital filter using bilinear transformation. 12. The specification of the desired lowpass filter is 1/2 |H ()| 1.0 |H ()| 0.08 ; 0 0.2 ; 0.4
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Design a Butterworth digital filter using impulse invariant transformation. 13. The specification of the desired lowpass filter is 0.9 |H ()| 1.0 |H ()| 0.24 ; 0 0.25 ; 0.5
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Design a Chebyshev digital filter using bilinear transformation. 14. Design a digital Chebyshev low pass filter satisfying the following specifications 0.707 |H (ej )| 1, |H (e )| 0.1
j
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0 0.2 0.5 with T= 1 sec (16)
using impulse invariant transformation. 15. Design a digital Butterworth filter satisfying the following specifications 0.9 |H (e j )| 1, |H (e j )| 0.2, 0 /2 3/4 with T= 1 sec.
using bilinear transformation. 16. Design a realize a digital filter using bilinear transformation for the following specifications i) Monotonic pass band and stop band ii) -3.01 db cut off at 0.5 rad iii) Magnitude down at least 15 db at = 0.75 rad.
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UNIT IV FIR FILTER DESIGN PART-A

1. What are the different types of filters based on impulse response? 2. How phase distortion and delay distortion are introduced? 3. What are FIR filters? 4. Write the steps involved in FIR filter design. 5. What are the advantages and disadvantages of FIR filters? 6. What are the design techniques of designing FIR filters? 7. What is the necessary and sufficient condition for the linear phase characteristic of a FIR filter? 8. What are the conditions to be satisfied for constant phase delay and constant group delay in linear phase FIR filters? 9. What is Gibbs phenomenon (or Gibbs oscillation)? 10. Write the steps involved in the design of FIR filters using windows. 11. What are the desirable characteristics of the window function? 1 2 . W ha t is t he p r in c ip le o f d e s ig n in g F I R f i lte r u s in g f r e q ue nc y s a mp lin g method? 12. Write the procedure for FIR filter design by fr e q ue nc y s a mp lin g method. 13. What is the draw back in FIR filter design using windows and frequency sampling method? How it is overcome? 14. Distinguish between FIR filters and IIR filters. 15. Write the characteristic features of rectangular and triangular window. 16. Why triangular window is not a good choice for designing FIR filters? 17. List the characteristics of FIR filters designed using windows. 18. List the features of hanning and hamming window spectrum. 19. List the features of Blackman and Kaiser Window spectrum. 20. What are the advantages of Kaiser Window? 21. Draw the direct form realization of FIR system. 22. Draw the direct form realization of a linear Phase FIR system for N even. 23. Draw the direct form realization of a linear Phase FIR system for N odd. 24. When cascade form realization is preferred in FIR filters? 25. What is transposition theorem & transposed structure? 26. What are the different types of arithmetic in digital systems? 27. What is meant by fixed point number? Write the types of fixed point arithmetic. 28. What is meant by sign magnitude and floating point representation? 29. What is meant by 1s and 2s complement form?
30. What are the quantization errors due to finite word length registers in digital filters? 31. What is input quantization error? 32. What is product quantization error? 33. What is the different quantization methods employed in digital system? 34. What is truncation? 35. What is rounding? 36. What are the two types of limit cycle behavior of DSP?. 37. What is dead band? 38. What is zero input and over flow limit cycle? 39. What are the methods to prevent overflow (or how overflow limit cycles can be eliminated)? PART-B 1. (i) Prove that an FIR filter has linear phase if the unit sample response satisfies the condition h(n)= h(M-1-n), n =0,1,.. M-1.Also discuss symmetric and anti symmetric cases of FIR filter. (ii) Explain the need for the use of window sequence in the design of FIR filter and Describe the window sequence generally used. 2. Design a lowpass filter using rectangular window by taking 9 samples of w(n) and with a cutoff frequency of 1.2 radians/sec. 3. Design a high pass filter using hamming window, with a cut off frequency of 1.2 rad/sec and N=9. 4. Design a bandpass filter to pass frequencies in the range 1 to 2 rad/sec using hanning window, with N=5. 5. Design a bandstop filter to reject frequencies in the range 1 to 2 rad/sec using rectangular window, with N=7. 6. Determine the coefficient of a linear phase FIR filter of length N= 15 which has a symmetric unit sample response and a frequency response that satisfies the conditation 1 Hr 2k/15 = 0.4 0 k=0, 1, 2, 3 k=4 k= 5, 6, 7 (16) (16) (16) (16) (16) (16) (8) (8)
7. Design a linear phase lowpass FIR filter with a cut-off frequency of /2 rad/sec.Take N=17. 8. The desired response of a low pass filter is e - 3 j H d (e j ) = 0 -3 /4 3/4 3 /4
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Determine frequency response h (ej ) for M=7 using a Hamming window. 9. Design a filter with Response Hd (ej ) = e-j2 0 for for - / 4 / 4 /4
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using Hanning window for M=11 10. Design an ideal differentiator with frequency response Hde j = j - (16) (8)
using rectangular and hamming window with N=7 11. (i) Determine the direct form of following system H (z) =1+2z-1 - 3z-2 + 4z-3 - 5z-4 (ii) Obtain the cascade form realizations of FIR systems H (z) = 1+5/2 z-1 + 2z-2 +2 z-3 12. (i) Explain the characteristics of a limit cycle oscillation with respect to the system described By the equation y(n) = 0.95 y(n-1) + x(n). Determine the dead band of the filter. (ii) Explain Gibbs phenomenon (or Gibbs oscillation). 13.(i) The output of A/D converter is applied to digital filter with the system function H (z) = 0.5 z/z-0.5. Find the output noise power from the digital filter when the input signal is quantized to have 8 bits. (ii) Compare hamming window and Kaiser Window. 14. Design and obtain the coefficients of a 15 tap linear phase FIR low pass filter using hamming window to meet the given frequency response. 1 Hd(w) = 0 for /6 w 15. Design an ideal Hilbert transformer having frequency response
H (e j) = j -j - 0 0
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Using rectangular window and Blackman window for N=11.

16. Design an ideal filter with
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1 H d (e ) =
j
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Using rectangular window with N=11.
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17. Design a FIR filter whose frequency response

H (e j) = 1 =0 /4 3/4 | | 3 /4.
Calculate the value of h (n) for N=11 and hence find H (z). UNIT- V APPLICATIONS PART-A 1. What is multirate signal processing? 2. Define down sampling. 3. What is meant by up sampling? 4. What is the need for anti-aliasing filter prior to down sampling? 5. What is the need for anti- imaging filter after up sampling? 6. Define sampling rate conversion. 7. Mention two applications of multirate signal processing. 8. Draw the block diagram of Quadrature mirror filter. 9. What is decimator? 10. What you mean by sub band coding? 11. What is an anti- image filter? 12. Explain the need for scaling in the digital filter realization. 13. Explain briefly the musical sound processing. 14. What are the basic operations of multirate signal processing? 15. What is meant by image enhancement? 16. Give some examples of image enhancement process. 17. Mention the basic approaches of image enhancement. 18. Mention some applications image sharpening. 19. What is bit-plane slicing? 20. What is a histogram? 21. What are the applications of histogram processing?
22. Mention some applications of image sharpening. 23. What are the various enhancement techniques in image processing? 24. List various voice compression coding techniques. 25. What is adaptive filter? 26. Mention some applications of adaptive filters. 27. What types of algorithms used in adaptive filters? PART-B 1. Explain the concept of decimation by a factor D and interpolation by factor I 2. With help of equation explain sampling rate conversion by a rational factor I/D 3. Explain the following application i) ii) speech compression sound processing (16) (16) (16)
4. With neat diagram explain any two applications of adaptive filter using LMS algorithm. 5. Explain speech vocoders and subband coding. 6. Explain how image enhancement restoration and coding can be done using signal processing. 7. Explain the methods of speech analysis and synthesis in detail. 8. Explain in detail the design and various implementations (structure) steps for filter using sampling Rate conversion system. 9. Describe how multirate dsp concepts are applied to basic music processing. (16) (16) (16) (16) (16) (16)
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CS2403 DIGITAL SIGNAL PROCESSING