R. V.

College of Engineering; Bengaluru-560059

(Autonomous Institution; Affiliated to VTU, Belgaum)
Department of Telecommunication Engineering
14ECS23 Modern Digital Signal Processing (MDSP)
Question Bank

Sl. No. Chapter-1 Assignment-I Marks

1. Design a digital FIR lowpass filter for the following specifications: 08
Passband: 4Khz
Stopband: 5KHz with atleast 40 dB attenuation
Sampling: 20KHz
2. Using the bilinear transformation, determine the order N and the cut-off 06
frequency c of analog prototype filter for the following discrete time design
Passband: 8 KHz
Stopband: 9 KHz
Passband ripple=0.5 dB
Stop band attenuation= -40 dB
Sampling frequency= 44 KHz
T = 1Second
3. Design a linear phase FIR filter with 7 taps and a cut-off frequency c=0.3 07
radians by frequency sampling method.
4. Design a digital lowpass Butterworth filter with the following specifications: 10
i) 3 dB attenuation at the passband frequency of 1.5 Hz.
ii) 10 dB stopband attenuation at the frequency of 3 kHz.
iii) Sampling frequency of 8000 Hz (Choose prototype Ha(s) for =1).
Use bilinear transformations.
5. What are the limitations of impulse invariant method of transformation? How 10
is mapping improved with bilinear transformation?
6. Design a digital FIR lowpass filter for the following specifications: 06
Passband: 4 KHz
Stopband: 5 KHz with atleast 40dB attenuation

1
 Sampling: 20K Hz
7. What is the drawback of Euler approximation method of IIR filter design? 06
Show how this drawback can be overcome in Bilinear transformation method?
8. Using the bilinear transformation, determine the order N and the cut off 08
frequency c of the analog prototype filter for the following discrete time
design.
Passband = 8KHz
Stopband = 9 KHz
Passband ripple = 0.5 dB
Stopband attenuation = 40dB
Sampling frequency = 44 kHz.
9. With relevant equation and diagrams, explain the lattice implementation of 10
FIR filter.
10. Differentiate between FIR & IIR system with suitable examples 08
11. Explain the design of FIR filter using windowing technique with appropriate 12
expression and sketches.
12. Explain the elementary operation of the digital filter implementation 10
13. Explain a general state space response of an IIR filter. 10
14. Given the system shown in figure below, determine the state space equation 04
and transfer function.

15. Find the reflection coefficient and show the lattice implementation of the 06
following filter.
X(z) = 1=1.4z-1+0.26 z-2+1.544 z-3 0.576 z-4- 0.414 z-5
16. Consider an IIR filter with transfer function 08

2
 2z 1 z 1.5
H( z )
z 2 1.6z 0.8
Draw the type -1 block diagram realization of H(z) & obtain the state space
equation of the output equation in matrix form.
17. Find the reflection coefficient and the lattice structure of an IIR filter given by 12
1
H( z ) when
A( z )

H(z) 1 0.09z 1 1.4299z 2 0.2789z 3 0.404z 4 0.1823z 5 0.0316z 6

Chapter -2 Assignment-II & III
18. For the system shown below, determine the equation for the spectrum Y() of 10
the output y(n).

19. A system is shown in Fig below, with an input x(n) & input spectrum X(. Determine 04
the DTFT Y() of the output signal for each of the following
i) B= /5
ii) B= /2
iii) B= 3/4
iv) B=

3
20. Explain with an example, how the filters with very sharp transition region are 06
designed, using multirate systems.
21. Prove that filter banks can be implemented very efficiently in terms of polyphase 10
decomposition DFT operation.
22. Explain how transmultiplexers are used to implement IDMA & FDMA. 10
23. Explain the design of perfect reconstruction filter banks, with real coefficients. 10
24. Discuss the lattice implementation of orthogonal filter banks. 10
25. What is the limitation of Fourier transformation? With relevant equation, explain how 06
Short Time Fourier Transformation (STFT) solves this problem.
26. Explain Gabor transform. 07
27. What is the limitation of STFT and Gabor transform? How CWT circumvent this 04
problem.
28. Write a short note on the Haar wavelet. 03
29. Explain the analysis of Downsampling and Upsampling. 10
30. Write a short note on the applications of multirate DSP. 10
31. Explain the analysis and synthesis of maximally decimated DFT filter banks 10
32. Explain the Time Division Multiple Access (TDMA) and frequency division multiple 10
technique in transmultiplexer.
33. ~ ~ 20
Determine the conditions on the four impulse responses g[n],h[n], g[n ] , h[ n ] in the
time domain as well as in the transform domain.
34. Explain the lattice implementation of orthogonal filter banks. 12
35. Write short note on Wavelet transform. 06
36. Explain the TDMA & FDMA technique in the context of multirate DSP theory. 10

4
37. Explain the design of perfect reconstruction filter banks with real coefficients. 10
38. With the help of an example, explain the multiresoltuion decomposition by 10
maximally decimated filter banks as applied to an audio signal.
39. Define Short Time Fourier Transform. Explain how STFT overcomes the limitations 07
of the Fourier transform.
40. Write short note on Daulcchies transform. 06
41. Explain the polyphase decomposition of a linear filter of Down sampling & Up 10
sampling.
42. Given that H(z) 1 z 1 2z 2 z 3 z 5 z 6 . Show a more efficient 10
realization in terms of polyphase decomposition.
43. Write the efficient re-diagram of decimator and interpolator, justify your answer with 10
FIR of the structure.
44. A FIR filter on Hamming window with following specification is to be implemented 10
with (i) Single stage (ii) Multiplestage (3 stages)
Passband is 0-450 Hz.
Stopband > 500 Hz
Sampling frequency Fs=96 KHz
Compare single stage and 3 stage implementation in terms of computation.
45. Explain analysis and synthesis structure of UDFT filter bank with efficient realization 10
structure.
46. With a block diagram, explain application of multirate DSP in subband coding of 10
speech signals.
47. Write a short note on
i) Adaptive DF filters and their applications 10
ii) Transmultiplexers for FDM to TDM and TDM to FDM. 10
48. Explain the sampling rate conversion by a factor D. 6
49. Consider an arbitrary digital filter transfer function 04

H ( z) h( n) Z
n
n
Perform a two component polyphase decomposition of H(Z).

1
Thus show that H(Z) can be expressed as H ( z ) H 0 (Z ) Z H1 (Z )
2 2

50. Obtain the direct form realization of sampling rate conversion by a factor I/D. 10
Mention advantages and disadvantages of it.

5
51. Explain Linear predictive coding of speech signal. 10
52. Explain the RLS algorithm and mention their properties and advantages over LMS 10
algorithm.