DHANALAKSHMI COLLEGE OF ENGINEERING DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING QUESTION BANK DIGITAL SIGNAL PROCESSING SUBJECT CODE:

CS2403 UNIT I SIGNALS AND SYSTEMS PART A Z Transform 1. What is meant by region of convergence? 2. What are the properties of z-transform? 3. State Parsevals relation in z-transform. 4. What is the relationship between z-transform and DTFT? Discrete time signals 1. State the classification of signal. 2. What are energy and power signals? 3. Define Signal Processing 4. Distinguish between continuous time signal and discrete time signal. 5. Define Digital Signal 6. What is deterministic signal? State some examples. 7. What is random signal? 8. Define (a) Periodic Signal (b) Non Periodic Signal 9. Define Anti Symmetric Signals 10. What are the types of signal representation? 11. What are the types of operations performed on discrete time signal? Discrete time systems 1. What is Linear Time Invariant system? 2. Define Discrete System 3. State the classification of discrete time system. 4. What is meant by static system? 5. What is meant by causal system? 6. Define Stable System 1 (Nov/Dec 2010) (Nov/Dec 2010) (Nov/Dec 2012) (April/May 2008) (Nov/Dec 2008) (April/May 2011) (Nov/Dec 2010) YEAR: III IT/ IV CSE

7. What is meant by linear system? 8. What is unit sample response (impulse response) of a system and state its significance? 9. What is the causality condition for an LTI system? Sampling Theorem 1. State Sampling theorem. 2. Define Nyquist Rate 3. What is meant by aliasing effect? 4. What is an anti-aliasing filter? 5. Define Resolution or Quantization step size 6. What is the condition for the impulse response to be stable? 7. What is meant by discrete linear convolution? 8. What are the steps involved in the convolution process? 9. What is meant by sampling process? 10. What is meant by quantization process? 11. How can aliasing be avoided? 12. What is meant by critical sampling? 13. What are the steps involved in the A/D conversion? 14. What is meant by quantization error? 15. What is quantization level? 16. What is SQNR? 17. What is the use of a sample and hold circuit? 18. Define Conversion Time 19. Define Resolution in terms of Voltage 20. Define Resolution in terms of Percentage 21. What are the advantages and disadvantages of counter - ramp type ADCs? 22. What are the advantages and disadvantages of SAADC? (April/May 2011) (Nov/Dec 2008) (April/May 2008) (Nov/Dec 2008) (Nov/Dec 2006)

UNIT II FREQUENCY TRANSFORMATIONS PART A Discrete Fourier Transform 1. Define Fourier Transform of a discrete time signal 2. State applications of Fourier transform. 4. Write the equation for Inverse Fourier Transform. Properties of DFT 1. State any four properties of DFT. 2. What is zero padding? 3. Define Circular Convolution Filtering methods based on DFT 1. What is the frequency response of an LTI system? 2. What are the properties of an LTI system? 3. Write short notes on the frequency response of first order system. 4. Write short notes on the frequency response of second order system. 5. Find the DFT of the sequence x(n) = { 1,1,0,0 } 6. When is DFT {X(k)} of a sequence x(n) imaginary? 7. When is DFT {X(k)} of a sequence x(n) real? 8. State circular frequency shifting property of DFT. 9. What is meant by convolution? Fast Fourier Transform 1. How many multiplications and additions are required to compute N-point DFT using radix-2 FFT? (Nov/Dec 2004) (April/May 2008) 3. Calculate the number of multiplications needed in the computation of DFT and FFT with 64point sequence. 4. Distinguish between DIT and DIF algorithms. 5. What are the applications of FFT algorithms? 6. What is FFT? 3 (May/June 2009) (May/June 2006) (Nov/Dec 2006) 2. How many multiplications and additions for direct computation of N-point DFT? (Nov/Dec 2008) (May/June 2009) (May/June 2009) (April/May 2008) (Nov/Dec 2010)

3. Distinguish between Fourier transform of discrete time signal and analog signal.

7. What is the speed improvement factor in calculating 64-point DFT of a sequence using direct computation and FFT algorithms? 8. What is meant by radix-2 FFT? 9. Why is FFT of a discrete time signal called signal spectrum?

UNIT III IIR FILTER DESIGN PART A Infinite Impulse Response 1. Distinguish between FIR and IIR filter. (Nov/Dec 2008) (Nov/Dec 2008) 3. What are the types of structures to realize IIR systems? 4. Distinguish between recursive realization and non-recursive realization. 5. How many number of additions, multiplications and memory locations are required to realize a system H(z) having M zeros and N poles in (a) Direct form I realization (b) Direct form II realization. Analog Filter Design 1. Compare Butterworth filter with Chebyshev Type -1 filter. 2. What is transposed structure? 3. What is canonic form structure? 4. What are the disadvantages of direct form realizations? 5. What is the advantage of cascade realization? 6. What are the types of filters based on impulse response? 7. What is the general form of IIR filter? 8. Write the no correction magnitude of Butterworth filter. What is the effect of order (N) on magnitude and phase response? 9. State any two properties of Butterworth low pass filter. 10. What is Butterworth approximation? 11. What is Chebyshev approximation? 12. What is Type-1 Chebyshev approximation? 13. What is Type-2 Chebyshev approximation? 14. Write the magnitude function of Chebyshev low pass filter? 15. How does the order of the filter affect the frequency response of Chebyshev filter? 16. How will you determine the order N of Chebyshev filter? 17. What are the types of filters based on the frequency response? (May/June 2009) 2. What are the advantages of Direct form- II realization over Direct form I realization?

18. How are the digital filters designed from the analog filters? 19. Write any two procedures to digitize an analog filter. Bilinear Transformation 1. What is bilinear transformation? 2. What is frequency warping? 3. What is prewarping? What is the use of prewarping? 4. What is meant by impulse invariant transformation? 5. What is the relation between digital and analog frequency in impulse invariant transformation. 6. What is the relation between digital and analog frequency in bilinear transformation? 7. Define Prewarping in IIR filter High Pass Filter 1. Define Signal Flow Graph 2. What are the design techniques for designing HPF filters? 3. What is meant by linear phase response? Band Pass Filter 1. What are the design techniques for designing BPF filters? Band Reject Filter 1. What are the design techniques for designing BRF filters? (April/May 2008) (Nov/Dec 2008) (April/May 2008)

UNIT IV FIR FILTER DESGIN PART A Finite Impulse Response 1. Write the procedure for designing FIR filter by Fourier series method. 2. What are the properties of FIR filter? 3. What are the steps involved in the FIR filter design? 4. How is the constant group delay and phase delay achieved in linear phase FIR filters? Linear Phase FIR Filter 1. What is the necessary condition for the linear phase characteristic of a FIR filter? (May/June 2006) 2. State the design techniques for linear phase FIR filter. Windowing Technique 1. What are the desirable characteristics of the window? 2. Write the equation for Hanning window function. 3. Write the equation for Hamming window function. 4. Write the equation for Blackman window function. 5. Write the equation for Bartlett window function. 6. Write the expression for frequency response of rectangular window. 7. Write the characteristic features of rectangular window. 8. State the features of FIR filter designed using rectangular window. 9. Write the equation for Kaiser Windows. 10. Write the characteristic features of Triangular window. 11. State the features of Hanning window spectrum. 12. State the features of hamming window spectrum. 13. Compare the Rectangular with Hanning window. 14. Compare the Hamming with Blackman window. 15. State the features of Blackman window spectrum. 16. State the features of Kaiser window spectrum. Frequency Sampling Technique (May/June 2009) (Nov/Dec 2010) (Nov/Dec 2004) (Nov/Dec 2010) (Nov/Dec 2004) (May/June 2006) 3. What are the possible types of impulse response for linear phase FIR filters? (April/May 2008)

1. What is Gibbs phenomenon? 2. Write the procedure for FIR filter design by frequency sampling method. UNIT V APPLICATIONS PART A Multi rate signal processing 1. Differentiate fixed point arithmetic from floating point arithmetic. 2. What is product quantization error? 3. What is truncation? 4. What is the need for multi rate signal processing? 5. Distinguish between down sampling and up sampling. 6. Distinguish between decimator and interpolator Speech processing 1. What is short time Fourier analysis? 2. What is linear prediction analysis? 3. What is channel vocoder? Musical processing 1. 1. 2. 1. What is the need for musical processing? State any two image enhancement methods. 2011) What is anti imaging filter What is adaptive equalization? Adaptive Filter Image enhancement

(Nov/Dec 2004)

(Nov/Dec 2006) (Nov/Dec 2010) (Nov/Dec 2006) (Nov/Dec 2012)

(April/May

(Nov/Dec 2012)

UNIT-I SIGNALS AND SYSTEMS PART B Z Transform 1. Find inverse Z transform of X(Z) = if (12) (Nov/Dec 2010)

a) ROC: |Z| > 1, b) ROC: |Z| < 0.5, c) ROC: 0.5 < |Z| <1. 2. Determine the transfer function, and impulse response of the system

y(n)

y(n 1) + y(n 2) = x(n) + x(n 1).

(8) (May/June 2009)

3. Find the Z-transform of y(n)-y(n-1)= x(n ) + x(n-1) Compute the system function H(Z) and the unit sample response of the system in analytical form. (8) (May/June 2009)

4. Consider a system y(n) + y(n 1) = x(n) + x(n 1). Find transfer function, and impulse response the system. 5. Derive expressions to relate Z-transform and DFT. 6. State and explain the scaling and time delay properties of Z transform. Discrete time signals 1. Find the convolution of the signals x(n) = 2. Explain the classification of signals with examples. Discrete time systems 1. Explain any four properties of LTI system with examples for each. (16) (Nov/Dec 2008) 2. Explain the classification of systems with examples. Convolution (8) and h(n) = u(n). (8) (8) (8) (8) (8)

1. Find the convolution sum of

and h(n) = (n) (n 1) + (n 2) (n 3). 2. Distinguish between linear and circular convolution with examples. UNIT II FREQUENCY TRANSFORMATIONS PART B Discrete Fourier Transform

(8) (May/June 2009) (16)

1. Using DFT and IDFT method, perform circular convolution of the sequences x(n) = {1, 2, 2, 1} h(n) = {1, 2, 3}. algorithms. 3. Compute the 8 point DFT of the sequence x(n) = [1,2,3,4,4,3,2,1]. 4. Compute DFT of the following sequences (1) x(n) = {1, 0, -1, 0} (2) x(n) = {j, 0, j, 1} 5. Find DFT of the sequence x(n) = { 1, 1, 1, 1, 1, 1, 0, 0} using radix-2 DIF FFT algorithm. (16) 6. Compute the eight point DFT of the given sequence x(n) = { , , , , 0, 0, 0, 0} using radix-2 DIT DFT algorithm. Properties of DFT 1. State the properties of DFT. 2. Illustrate the circular convolution property of DFT. Filtering methods based on DFT 1. Explain in detail the decimation in time algorithm with suitable examples. 2. Explain in detail the decimation in frequency algorithm with suitable examples. Fast Fourier Transform (16) (16) (8)(May/June 2009) (10)(April/May 2008) (16) (10)(April/May 2011) (16)(May/June 2009) (10)(April/May 2008) (8) 2. Compute the 8 point DFT of the sequence x(n) = [1, 2, 3, 4, 4, 3, 2, 1] using radix-2 DIT and DIF

10

1. Derive and draw the flow graph of the radix-2 DIF FFT algorithm for the computation of 8-point DFT. (10)(April/May 2008)

UNIT III IIR FILTER DESIGN PART B Infinite Impulse Response 1. Explain in detail the concepts of impulse invariance method of IIR filter design and summarize the design steps. (16)(May/June 2009)

2. Obtain the cascade and parallel form realization structures for the system given by the difference equation y ( ) = 0 .1 ( 1) + 0.2 ( 2) + 3 ( ) + 3.6 (n-1)+0.6 ( 2). (8)(Apr/May 2008) 3. Design IIR filter using impulse invariance technique. Given that and

implement the resulting digital filter by adder, multipliers and delays. Assume sampling period =1 sec. (8)

4. Obtain the direct form I, canonic form and parallel form realization structures for the system given by the difference equation y( ) = 0 .1 ( 1) + 0.72 ( 2) + 0.7 ( ) 0.252 ( 2). 5. If 5 samples/sec. Analog Filter Design , find (8)

using impulse invariant method for sampling frequency of (8)

11

1. Design Butterworth filter using bilinear transformation method for the following specifications 0.8 |H(ej)| 1; |H(ej)| 0.2; 0 0.2 0.6 (16)(Nov/Dec 2012) 2. Find H(s) for the third order low pass Butterworth filter. (6)(April/May 2008) 3. Design digital low pass filter using bilinear transformation. Given that

Assume sampling frequency of 100 rad/ sec. 4. Explain the round off effect in digital filters. 5. Explain in detail the designing methods of IIR filters from analog filters. 6. Compare FIR with IIR filters. Bilinear Transformation 1. Design a digital filter with H(s) = 1/(s2+7s+12) using T = 1sec. of 100 rad/sec and sampling time T = 1.2ms.

(8) (10) (16) (8)

(8)(April/May 2008) (8)(April/May 2008)

2. Design digital low pass filter using BLT for H(s) = 2/{(S+1)(S+2)} with cut-off frequency 3. Design an IIR digital low pass Butterworth filter to meet the following requirements: Pass band ripple (peak to peak) 0.5dB, Pass band edge: 1.2kHz, Stop band attenuation 40dB, Stop band edge: 2.0 kHz, Sampling rate: 8.0 kHz. Use bilinear transformation technique. (16)

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UNIT IV FIR FILTER DESGIN PART B Finite Impulse Response 1. Design a symmetric FIR low pass filter whose desired frequency response is given as ( )= The length of the filter should be 7 and = 1 rad/sample using rectangular window. (16) (May/June 2009) 2. Determine the first 15 coefficients of FIR filters with magnitude specification given below using frequency sampling method:

)=

(12) (10)

3. Explain the effect of finite word length on digital filter. Linear Phase FIR Filter 13

1. For an FIR linear phase digital filter approximating the ideal frequency response,

( )= Determine the coefficients of a 5 tap filter using rectangular window. (10)(April/May 2008) 2. Determine the coefficients h(n) of a linear phase FIR filter of length M = 15 which has a symmetric unit sample response and a frequency response

Hr

= (12)(May/June 2009)

3. Determine the unit sample response ( ) of a linear phase FIR filter of length M = 4 for which the frequency response at (8) Windowing Technique 1. Explain any three window techniques used in the design of FIR filters. (16)(April/May 2008) and is given as
r

(0) = 1 and Hr(

Frequency Sampling Technique 1. State the advantage of floating point representation over fixed point representation. 2. Explain in detail the type-I frequency sampling method of designing an FIR filter. (8) (8)

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UNIT V APPLICATIONS PART B Multi rate signal processing 1. Derive and explain in detail the frequency domain characteristics of the decimator by the factor M and interpolator by the factor L. (12)(April/May 2011)

2. With neat diagram and supportive derivation explain in detail multirate signal processing using two techniques. decimated signal. Speech processing 15 (16) (16) 3. Explain in detail decimation of sampling rate by an integer factor D and derive spectra for

1. Write short notes on a) Speech Processing b) Vocoder Musical processing 1. Write notes on: a) b) Sub band coding of speech signals. (8) Musical sound processing. (8) Adaptive Filter 1. With neat diagram explain in detail any two applications of adaptive filter using LMS algorithm. 2. Explain in detail : a) b) filter. 4. Discuss in detail various quantization effects in the design of digital filters. Adaptive noise cancellation with a neat block diagram Image enhancement techniques (12)(April/May 2011) (Nov/Dec 2012) (8) (8) (16) (12) (8)

3. What is adaptive filter? With neat block diagram explain any four applications of adaptive

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