4thSEM DSP QUESTION BANK

VINAYAKA MISSIONS UNIVERSITY

V.M.K.V ENGINEERING COLLEGE, SALEM

 

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

Semester: IV                                                                                    Year: II

DIGITAL SIGNAL PROCESSING

QUESTION BANK

UNIT I

PART A

1.                  What is Digital Signal Processing?

2.                  List any four applications of DSP?

3.                  Define DFT.

4.                  What is circular convolution?

5.                  State shifting property of DFT.

6.                  What is zero padding?

7.                  What is DIT radix-2 FFT?

8.                  Why FFT is needed?

9.                  Compare the DIT and DIF radix-2 FFT.

10.              Draw the basic flow graph of DIT radix-2 FFT.

11.              Draw the basic flow graph of DIF radix-2 FFT.

12.              Give any two applications of DFT.

13.              List any four properties of DFT.

14.              What is the relation between Z-transform and DFT.

15.              Find the DFT of the sequence x(n)={1,1,0,0}

16.              Find the IDFT of Y(k)={1,0,1,0}

17.              How many multiplications & additions are involved in radix-2 FFT.

18.              What is twiddle factor?

19.              What are the phase factors involved in the third stage of computation in the 8 point DIT radix-2 FFT?

20.              Arrange the 8-point sequence x(n)={1,2,3,4,-1,-2,-3,-4} in bit reversed order.

21.              What is bin spacing?

22.              What are the phase factors involved in the first stage of computation in the 8 point DIF radix-2 FFT?

23.              What are the 2 methods of sectioned convolution?

24.              Why circular convolution is important in DSP?

25.              List the differences between linear and circular convolution.

PART B

1. i) Find the circular convolution of the two sequences x₁(n) = (1, 2, 2, 1) and

       x₂ (n) =(1,2,3,1) using concentric circle method. 

   ii) Find the circular convolution of the two sequences x₁(n) = (1, 2, 2, 1) and

         x₂ (n) =(1,2,3,1) using matrices method

2 .Find the circular convolution of the two sequences x₁(n) = (2, 1, 2, 1) and

          x₂ (n) =(1,2,3,4) using DFT and IDFT method 

3. An 8 point sequence is given by x(n)=(2,1,1,2,1,1,1,1) compute 8 point DFT of x(n) by radix-2 DIT-FFT        

4. An 8 point sequence is given by x(n)=(2,1,1,2,1,1,1,1) compute 8 point DFT of x(n)   by radix-2 DIF-FFT

5.An 8 point sequence is given by x(n)=(2,2,0,2,1,1,0,1) compute 8 point DFT of x(n) by radix-4 DIT-FFT.

6. An 8 point sequence is given by x(n)=(2,2,2,2,1,1,1,1) compute 8 point DFT of x(n) by radix-2 DIT-FFT .

7.Find the linear and circular convolution of the sequences x(n) & h(n),x(n)={1,0.5} & h(n)={0.5,1}.

8. An 8 point sequence is given by x(n)=(1,2,3,4,4,3,2,1) compute 8 point DFT of x(n)   by radix-2 DIF-FFT.

9.In an LTI system 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.

10.An 8 point sequence is given by x(n)=(1,2,3,4,4,3,2,1) compute 8 point DFT of x(n)   by radix-2 DIT-FFT

UNIT II

PART A

1. Compare IIR and FIR filter.
2. What is main objective of impulse invariant method?
3. What is bilinear transformation?
4. Compare the digital and analog filter.
5. What is impulse invariant transformation?
6. Mention the important features of IIR filters.
7. Write the impulse invariant transform used to transform complex conjugate poles.
8. How bilinear transformation is performed?
9. Write the transfer function of unnormalized Butterworth low pass filter.
10. Write the transfer function of normalized chebyshev low pass filter.
11. Write the transfer function of normalized Butterworth low pass filter.
12. Mention any two techniques for digitizing the transfer function of an analog filter.
13. Classify the filters based on frequency response.
14. What is frequency warping?
15. What are the advantages & disadvantages of digital filters?
16. What are the requirements for a digital filter to be stable & causal?
17. How the poles of Butterworth transfer function are located in s-plane?
18. What is chebyshev approximation?
19. Sketch the magnitude response of type-1 chebyshev filters.
20. Write the properties of Butterworth filter.
21. What is pre-warping? Why it is employed?
22. What are the requirements for a analog filter to be stable & causal?
23. What is the relation between digital & analog frequency in bilinear transformation?
24. Distinguish between recursive realization and non recursive realization.
25. What are the properties of chebyshev filter?

PART B

1. Convert the analog filter with transfer function H(s) into digital filter using bilinear    transformation. 

i) H(s) =  ii)         H(s) =     

2. i) Convert the analog filter with system transfer function                                     

                                                                  (S+0.1)

H(s)    =   -------------         

                                                                (S+0.1)²+9

     into a digital IIR filter by means of the impulse invariance technique.

    ii) Write the design procedure for lowpass digital Butterworth IIR filter.

3. Convert the analog filter with system transfer function

            i) H(s) =       ii)  H(s) =

   into a digital filter by means of the impulse invariance technique. If a) T= 1 sec and b) T=     0.1 sec.

4. i)Obtain the Direct form–I realizations of the LTI system governed by the equation        

           y(n)= 0.5y(n-1)-0.25 y(n-2)+x(n)+3x(n-1)  

    ii) Determine the direct form II realizations for the following system

            y(n)= - 0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.252x(n-2)                

5. Design a Butterworth IIR filter with the following specifications                       

                     0.8 ≤ |H (ω)| ≤ 1.0       ;           0 ≤ ω ≤ 0.2p

                                |H (ω)| ≤ 0.2     ;           0.32p ≤ ω ≤ p

 Using impulse invariant method.

6. Find the digital network in direct & transposed form for the system described by the difference equation.

         y(n) =x(n)+0.5x(n-1)+0.4x(n-2)-0.6y(n-1)-0.7y(n-2).

7. Convert the analog filter with system transfer function

            i) H(s) = ­­     ii) H(s) =

  into a digital filter by means of the approximation of derivatives technique. If a) T= 1 sec and b) T=     0.2 sec.

8. i) Write the design procedure for low pass digital Chebyshev IIR filter.

   ii).compare the Butterworth & chebyshev type-1 filters.

9. Design a Butterworth IIR filter with the following specifications                       

                     0.89125 ≤ |H (ω)| ≤ 1.0           ;           0 ≤ ω ≤ 0.2p

                                |H (ω)| ≤ 0.17783          ;          0.3p ≤ ω ≤ p

Using impulse invariant method.

10. i. Apply bilinear transformation to H(s) =       with T=1sec and find H (Z).

      ii. A digital filter with a 3dB bandwidth of 0.25π is to be designed from the analog filter whose system response is

                                                         H(S) =

UNIT III

PART A

1.      What are the types of digital filter according to their impulse response?

2.      How phase distortion and delay distortion are introduced?

3.      Write the steps involved in FIR filter design.

4.      What are the advantages of FIR filter?

5.      What are the disadvantages of FIR filter?

6.      Write the procedure for designing FIR filter using windows.

7.      What is Gibb’s phenomenon?

8.      List the well known design technique for linear phase FIR filter design?

9.      Write the characteristic features of rectangular window.

10.  Write the expression for Kaiser Window function.

11.  What do you understand by linear phase response?

12.  What are the properties of FIR filter?

13.  List out the disadvantages of fourier series method.

14.  What is window? Why it is necessary?

15.  List the desirable features of Kaiser Window spectrum.

16.  What are the desirable characteristics of the window?

17.  Write the expression for frequency response of rectangular window & sketch the magnitude response.

18.  For what type of filters frequency sampling method is suitable?

19.  Write the magnitude & phase function of FIR filter when impulse response is symmetric and N is even.

20.  Write the magnitude & phase function of FIR filter when impulse response is antisymmetric and N is Odd.

21.List the well known design technique for linear phase FIR filter design?

22.What is the reason that FIR filter is always stable?

23.When cascade from realization  is preferred in FIR filters?

24.What is the principle of designing FIR filter using frequency sampling method?

25.For what type of filters frequency sampling method is suitable?

PART B

1. Design a FIR low pass filter with cutoff frequency 1 kHZ and sampling rate of 4 kHZ

with 11 samples using fourier series method.

2.Design a low pass filter using rectangular window by taking 9 samples of w (n) and with a

     cutoff frequency of 1.2 radians/sec.                                                                     

3. Determine the coefficients of a linear phase FIR filter of length M=15 has a symmetric

    unit sample response and a frequency response that satisfies the condition.        

           

                        H (2pk/15) = {1          ; for k=0, 1, 2, 3

                                                0          ; for k=4, 5, 6, 7}

4. A low pass filter is to be designed with the following desired frequency response

                      H( e)={ e,-p/4p/4

                                       0       , p/4p     }

Determine the filter coefficients h(n) if the window function is defined as

                          w(n)= {1, 04    

                                       0, otherwise

5. Design a band stop filter to reject frequencies in the 1 to 2 rad/sec using rectangular window, with N=7   

6. Determine the coefficients of a linear phase FIR filter of length N=15 which has a symmetric unit sample response that satisfies the conditions                           

                        H (2pk/15) = {1          ; for k=0, 1, 2, 3

                                                0.4       ; for k=4

                                                0          ; for k=5, 6, 7}           

7. (i) Draw the direct form structure of the FIR system described by the transfer function          

                H (Z) =1+1/2z^-1+3/4z^-2+1/4z^-3+1/2z^-4+1/8z^-5                                                                                 

    (ii) Realize the following system with minimum number of multipliers                            

                H (Z) =1/4+1/2z^-1+3/4z^-2+1/2z^-3+1/4z^-4

8. A filter is to be designed with the following desired frequency response

                      H( e)={0, -p/4p/4

                                       e            , p/4p     }

Determine the filter coefficients h(n) if the window function is defined as

                          w(n)= {1, 04    

                                       0, otherwise

9. i)Obtain direct form and cascade form realizations for the transfer function of an FIR system given by

H(Z)=( 1-1/4z^-1 +3/8z^-2  ) ( 1-1/8z^-1 - 1/2 z^-2  )

ii) Realize the following system with minimum number of multipliers                                 

                H (Z) =(1+1/2z^-1+z^-2)  (1+ 1/4z^-1+ z^-2  )

10. Design a FIR low pass filter with cutoff frequency 2 kHZ and sampling rate of 5 kHZ

with 9 samples using fourier series method.

UNIT-IV

PARTA

1. What is meant by finite word length effects in digital filters?
2. What are the different formats of fixed point representation?
3. Compare the fixed point and floating point number representations.
4. What are the advantages of floating point arithmetic?
5. What is truncation?
6. What is rounding?
7. What are the three quantization errors due to finite word length registers in digital filters?
8. What are limit cycles?
9. What is zero input limit cycle?
10. What is dead band?
11. What is quantization error?
12. How overflow limit cycles can be eliminated?
13. What do you mean by overflow oscillations?
14. What is A/D conversion noise?
15. What are the advantages of floating point arithmetic?
16. Identify the various factors which degrade the performance of the digital filter implementation when finite word length is used.
17. Express the fraction -7/8 in sign magnitude,1’s complement and 2’s complement.
18. What is product quantization error?
19. What is meant by quantization step size?
20. How would you relate the steady state noise power due to quantization to the b bits representing the binary sequence?
21. What are the methods used to prevent overflow?
22. What is meant by saturation arithmetic?
23. What are the two kinds of limit cycle behavior in DSP?
24. Explain briefly the need for scaling in the digital filter implementation.
25. Why rounding is preferred to truncation in realizing digital filter?

PART B

1. Briefly explain finite word length effect in digital filters?                            
2. For the digital network shown in figure find H(z) and scale factor S₀ to avoid overflow in register A₁                                                                              

  x(n)            So                   A1                  w(n)                                        0.245          y(n)

                                                           w(n-1)

                                       

3. For the recursive filter shown in figure the input x(n) has a peak value of 10V, represented by 6 bits. Compute the variance of output due to A/D conversion process?

 

4. (i) Discuss on fixed point representation?                                                     

      (ii) Explain signal scaling in finite word length effects?                               

5. Write short notes on

i)                    Channel vocoder                           ii)       Homomorphic vocoder

6. For the digital network shown in figure find H(z) and scale factor S₀ to avoid overflow in register A₁

  x(n)            So                   A1                  w(n)                                        0.345          y(n)

7. Explain the characteristics of a limit cycle oscillation with respect to the system described by the difference equation y(n)=0.95y(n-1)+x(n).Determine the dead band of the filter.

8. The output of an A/D converter is applied to a digital filter with the system function

a)   H (Z) =                       b)   H (Z) =

Find the output noise power from the digital filter, when the input signal is quantized to have eight bits.

9. Write short notes on the following

             

                                                              i.      Overflow error

                              ii. Truncation error

                                                            ii.      Coefficient Quantization Error.

10. For the recursive filter shown in figure the input x(n) has a peak value of 15V, represented by 8 bits. Compute the variance of output due to A/D conversion process?

UNIT V

PART A

1.      What are the classification digital signal processors?

2.      What are the factors that influence selection of DSPs?

3.      What is pipelining?

4.      What is pipeline depth?

5.      What are the different stages in pipelining?

6.      What are the different buses of TMS320C5x?

7.      What is the function of read bus?

8.      What are the arithmetic instructions of C5x?

9.      What are the logical instructions of C5x?

10.   What are the shift instructions of C5x?

11.  Define dual ported memory?

12.  Define RISC Processors?

13.  Define CISC Processors?

14.  What are Programmable Digital Signal Processors?

15.  What is meant by circular addressing mode?

16.  What is meant by bit reversed addressing mode?

17.  What is the use of host ports in P-DSPs?

18.  List the relative merits and demerits of RISC and CISC Processors?

19.  Explain the memory mapped addressing mode used in the P-DSPs.

20.  What are the different ways in which the operand for instructions can be specified using indirect addressing mode?

21.  Compare the multiplier units in 54X and 5X.

22.  Compare the address and data buses of 54X and 5X.

23.  What is the use of the guard bits in accumulators?

24.  Mention some applications of on-chip timer in P-DSPs.

25.  What are the different types of addressing modes available in TMS 320 C 54X .

PART B

1) i) With a suitable diagram describe the functions of ALU unit of TMS320C54x.

    ii) Explain the operation of multiplier/adder unit of TMS320C54x.

2) Draw and explain the architecture of TMS320C50?

3) Write short notes on

Multiported memory

MAC

VLIW architecture

4) Explain the different addressing modes of TMS320C50?

5) Explain the different applications of PDSPs?

6). Explain the difference between Von Neumann and Harvard architecture for the Computer. Which architecture is preferred for DSP applications and why?

7). Explain what is meant by instruction pipelining. Explain with an example, how pipelining increase the throughput efficiency.

8). Explain the various ON-_CHIP peripherals of P-DSPs.

9). Explain how convolution is performed using a single MAC unit. And also list out the merits and demerits of RISC and CISC processors.

10). Explain the different addressing modes in P-DSPs.