Dr H S Prashanth, PESIT

Assignment questions ADSP (Microelectronics & Control systems)

Course Instructor: Dr Prashantha H S
DISCRETE TIME SIGNALS
(Sequences, representation of signals on orthogonal basis, sampling and reconstruction of
signals) (STUDENT: N SUKUMAR, ARUNA)

1 The signal g(t)=10 cos (20πt) cos (200πt) is sampled at the rate of 250 samples/sec.
a) Determine the spectrum of the resulting samples signal.
b) Specify the cutoff frequency of the ideal reconstruction filter so as to recover g(t) from its
Sampled version.
c) What is the Nyquist rate for g(t)?

2 A low pass signal x(t) has a spectrum X(f) given by

1  ||

   200  ; ||  200
0 ; 

a) Sketch the spectrum Xδ(t) for |f|<200 Hz if x(t) is ideally samples at fs=300 Hz.
b) Repeat part 1 for fs=400 Hz

3 State and prove sampling theorem for low pass signals

4 A discrete time signal x[n] is defined as

 n
1 + 3 ;−3 ≤ n ≤ −1

x[n] = 1;0 ≤ n ≤ 3
0; elsewhere


a) Determine its values and sketch the signal x [n].
b) Sketch the signal that result if we
1) First fold x[n] and then delay the resulting signal by four samples.
2) First delay x[n] by four samples and then fold the resulting sequence
c) Sketch the signal x [-n + 4].
d) Express the signal x [n] in terms of δ [n] and u [n]

5 Find the frequency response and step response of the following LTI system represented by h[n] =
(0.5)n u[n]

6 DIGITAL SIGNAL PROCESSING, 4TH EDITION, PROAKIS PROBLEMS

1.5-1.9, 2.2
 Dr H S Prashanth, PESIT
DISCRETE SYSTEMS

(Z-Transform, analysis of LSI system, frequency analysis, inverse systems, DFT, FFT,
Implementation of discrete time systems)

Z-TRANSFORMS (STUDENTS: RAKESHA & RAZIK P)

1 Find the Z Transform of the signal x(n)=3 (2)n u(-n)

2 Convolve x1(n) and x2(n) using Z Transforms: x1(n)=(1/3)n u(n) and x2(n)=(1/5)n u(n)

3 A causal discrete time LTI system is implemented using the difference equation
5 1
      1    1    2
6 6
1. What is the transfer function  of this system?
2. Sketch the pole-zero diagram of the system.
3. Find the impulse response 

4 The discrete time signal  is shown in the figure.

1. What is the Z-transform
 of the signal 
2. Define    
, sketch the signal 
3. Define ! 
, sketch the signal "
4. Define # 
1/, sketch the signal 

Figure
5 List the properties of ROC

% %
6 Find the Z-transform of    ' (   ' (  1
&

%
7 Find the inverse Z-transform of
   )1   % * 1   % 1  2  % 

8 Find the inverse Z-transform of


+

 
+ +%+%,
for |Z| > 2

9 Find the Z-Transform of an u(-n-1) and find ROC

10 + %
+ , &+-%
Find the inverse Z-Transform of X(Z)= for |Z|<

11 Obtain the convolution of αn u(n) and βn u(n) using Z-Transforms where α<1 and β<1
 Dr H S Prashanth, PESIT
12 Find the Z-transform and the associated ROC for the sequence ./  0 ' (

+
13 Using the power series expansion method, find the inverse Z-transform of
 
+ , &+-%
|Z|
<½

& %
14 A causal discrete-time LTI system is described by ./  .  1/  .  2/  ./
1 2
Where x[n] and y[n] is the input output of the system, respectively.
1) Determine the system function H(Z) 2) Find the impulse response h[n] of the system

15 DIGITAL SIGNAL PROCESSING, 4TH EDITION, PROAKIS PROBLEMS

2.33, 2.34, 2.35,

DFT & FFT (STUDENTS: VIMALPRASAD, ARAVINDAN, VIGNESH)

1 Perform the circular convolution for 3  4 and linear convolution for the two sequences given by
%   52, 1, 2, 17 and    51, 2, 3, 47. Perform the circular convolution of the above sequences
using frequency domain approach.

2 Discuss the procedure to obtain the filtering of long duration sequences by considering the length
of input sequence as 15 and length of the impulse response equal to 3 by using overlap add and
overlap save method.

3 A point sequence  is given by   52, 1, 2, 1, 2, 1, 2, 17. Compute 8-point DFT of  using
radix 2 DIT-FFT.

4 Determine the 8 point DFT of x(n)=cos (2πkon / N), 0 ≤ n ≤ N-1

5 What are the properties of phase factor WN that are exploited in FFT algorithm? Prove them

6 A long sequence  is filtered through a filter of impulse response  to give output .
Given  and  as follows, compute  using overlap add technique.
  .1 1 1 1 1 1 3 1 1 4 2 1 1 3 1 1 1/
  .1  1/
Compare overlap add technique with overlap save method for filtering long duration sequences

7 Develop radix 2 decimation in frequency (DIF) FFT algorithm with all necessary steps and neat
signal flow diagram used in computing N-point DFT
9 of an N-point sequence. Using the
same, compute the 4 point DFT of a sequence   .44 22 33 22/using FFT algorithm.

8 Consider the finite length sequence   :  2:  5

1. Find the 10-point discrete Fourier transform of 
,=
2. Find the sequence that has a discrete Fourier transform 9   ;< >?
9
Where
9 is the 10-point DFT of 
3. Find the 10-point sequence that has a discrete Fourier transform 9 
9@9
Where
9 is the 10-point DFT of , and @9 is the 10-point DFT of the sequence
 Dr H S Prashanth, PESIT
1 0 B  B 6
  A
0 CDE

9 Let  be the sequence  2:  :  1  :  3. The 5-point DFT of sequence 
is computed and the resulting sequence is squared: 9 
9
A 5-point inverse DFT is then computed to produce the sequence . Find the sequence 

10 Assume that a complex multiply takes 1µs and that the amount of time required to compute a DFT
is determined by the amount of time it takes to perform all of the multiplications.
1. How much time does it take to compute a 1024 point DFT directly?
2. How much time is required if an FFT is used?

11 Determine the 8 point DFT of = {3, 1, 5, 4, 2, 1, 0, 1} using radix 2 DIF FFT algorithm. Show
clearly all the intermediate results. Plot both magnitude and phase spectra.

12 Determine the response of an LTI system with   51, 27 for an input   51, 2, 1, 2,
3, 2, 3, 1, 1, 1, 2, 17. Employ overlap save method with block length N=4.

I' I'
13 Find the 4-point circular convolution of 1 with 2 if 1  cos   and 2  sin  
J J

14 We would like to linearly convolve a 3000 point sequence with a linear shift invariant filter whose
unit sample response is 60 points long. To utilize the computational efficiency of the FFT
algorithm, the filter is to be implemented using 128 point discrete Fourier transform and inverse
discrete Fourier transforms. If the overlap add method is used, how many DFT’s are needed to
complete the Filtering operation?

15 A signal M D that is band limited to 10 KHz is sampled with a frequency of 20 KHZ. The DFT of
N=1000 samples of  is then computed, that is,
OP

9  ∑J%
'RS 
; I
Q with N=1000

1. To what analog frequency does the index K=150 correspond?

What is the spacing between the spectral samples?

16 Let
9 denote the N-point DFT of an N-point sequence .
9 itself is an N-point sequence,
if the DFT of
9 is computed to obtain a sequence % . Determine %  in terms of  

17 Consider the finite length sequence ;   51, 3U4 , 1U2 , 1U4 7. The 4-point DFT of  is

9, find the sequence  whose DFT is 9  @1 &<
9

18 Compute the circular convolution of the two sequences using DFT and IDFT approach where the
two sequences are given by
%   2 :  3 :  1  :  2  :  3
   :  3 :  1  5 :  2  3 :  3

19 Derive radix 2 DITFFT algorithm for N=8 and using the resulting signal flow graph compute the 8-
point DFT of an 8-point sequence   51, 1, 0, 0, 1, 1, 0, 07. Show the results at intermediate
stages
 Dr H S Prashanth, PESIT

20 Determine the response of an LTI system with   51, 1, 27 for an input signal  
51, 0, 1, 2, 1, 2, 3, 1, 0, 27. Employ overlap add method with block length N=4 and use DIFFFT
algorithm to compute the response of each block.

21 Derive radix 2 DITFFT algorithm for N=8 and using the resulting signal flow graph compute the 8-
point DFT of an 8-point sequence   51, 1, 0, 0, 1, 1, 0, 07. Show the results at intermediate
stages

22 Determine the response of an LTI system with   51, 1, 27 for an input signal  
51, 0, 1, 2, 1, 2, 3, 1, 0, 27. Employ overlap add method with block length N=4 and use DIFFFT
algorithm to compute the response of each block.
 Dr H S Prashanth, PESIT
FIR FILTERS

(Window method, Park-Mcclennan’s method) (STUDENTS: VIPINKUMAR , YASHWANT REDDY)

1 Design and realize LPF using rectangular window by taking 9 samples of  and with a cutoff
frequency of 1.2 radian/sec

2 List the desirable features of Kaiser Window spectrum.

3 Design a ideal differentiator with frequency response

 ;V ) W X Y B  B Y
Using Hamming window with N=7, plot the frequency response.

4 Discuss the steps involved in the design of FIR filters using window based method. Also write
explicitly the following window functions mathematically
1. Rectangular 2. Bartlett 3. Hamming 4. Hanning

5 Design a ideal differentiator with frequency response

 ;V ) W X Y B  B Y
Using Hamming window with N=5, plot the frequency response.

6 Determine the fir filter coefficients h(n) which is symmetric low pass filter with linear phase. The
desired frequency response is
Hd(w) = { e-j [(M-1)w] / 2 ; 0 ≤ |w| ≤ π / 4
{0 ; otherwise
Employ rectangular window and Hanning window with M=7

7 A low pass FIR causal filter is to be designed with the following desired frequency response
 ; V ; YU4 B  B YU4
Z  ;V   [ 
0 ; YU4 B || B Y
Determine the filter coefficients  if a rectangular window of width 5 samples is used.
Also find the frequency response.

8 Discuss the frequency sampling method of FIR filter design. Use the frequency sampling method to
design an FIR filter with  51, 2, 17. Also indicate the signal flow graph.
9 Discuss the design steps with equations involved in the design of FIR filter using window method.

10 Let  be the unit sample response of FIR filter so that  is zero for   0,  \ 3. Assume
 is real and the frequency response of the filter can be represented in the form
 ;]    ^  ;]  ;_]
1. Find `@ for 0 B  B Y when  satisfies the condition   3  1  
2. If N is even, show that   3  1   implies that a3U2b  0 where 9 is the N-
point DFT of 

11 Discuss the frequency sampling method of FIR filter design. Use the frequency sampling method to
design an FIR filter with  51, 2, 17. Also indicate the signal flow graph.
 Dr H S Prashanth, PESIT
12 Let  be the unit sample response of FIR filter so that  is zero for   0,  \ 3. Assume
 is real and the frequency response of the filter can be represented in the form
 ;]    ^  ;]  ;_]
1. Find `@ for 0 B  B Y when  satisfies the condition   3  1  
2. If N is even, show that   3  1   implies that a3U2b  0 where 9 is the N-
point DFT of 
 Dr H S Prashanth, PESIT
IIR DIGITAL FILTERS

(Butterworth, Chebyshev, Elliptic approximations, Lowpass, bandpass bandstop and high pass
filters) (STUDENTS: PRATAP C PATIL, VANITHA , DIVYAJYOTHI)

1 Given the specifications cd  3 ef, cg  16 ef, d  1 9h and i  2 9h. Determine the order of
the filter using Chebyshev approximation. Also find .

2 Design an analog Butterworth filter that has a -2 dB pass band attenuation at a frequency of 20
radians/sec and at least -10 dB stop band attenuation at 30 radians/sec.

3 Discuss the following frequency transformation in analog domain

1. Low pass to high pass 2. Low pass to band pass

4 Convert the analog filter with system function M j in to a digital filter using bilinear
transformation
j  0.3
M j 
j  0.3  16

5 Obtain the impulse response of a digital filter to correspond to an analog filter with impulse
response M D  0.5   k and with a sampling rate of 1 KHz using impulse invariant method.

6 With respect to bilinear transformation, what is warping effect? Discuss the relation between
analog and digital frequency. What is the effect on the magnitude and phase response due to
warping effect?

7 Design low pass Butterworth filter using impulse invariant method for satisfying the following
constraints:
• Pass band wp : 0.162 radians
• Stop band ws : 1.63 radians
• Pass band ripple : 3 dB
• Stop band attenuation : 30 dB
• Sampling frequency : 8 KHz
Choose the cut-off frequency to meet the requirements of stop band only.

8 Design low pass Chebyshev filter using bilinear transformation method for satisfying the following
constraints:
• Pass band wp : 0-400 Hz
• Stop band ws : 2.1-4 KHz
• Pass band ripple : 2 dB
• Stop band attenuation : 20 dB
• Sampling frequency : 10 KHz
Determine the difference equation representation of the digital filter

9 Design an analog band pass filter to satisfy the following specifications:

• -3 dB upper and lower cutoff frequency of 100 Hz and 3.8 KHz
• Stop band attenuation of 20 dB at 20 Hz and 8 KHz
 Dr H S Prashanth, PESIT
• No ripple in the within both stop band and pass band
Check your design by plotting 20 log |WΩ| for different values of Ω

10 Derive the bilinear transformation for obtaining IIR filters from analog filters. Verify the stability
of the mapping. What is the relation between analog and discrete frequency variables?

11 Design a chebyshev type1 analog filter to meet the following specifications:

Passband attenuation of 2dB at 4rad/sec and stop band attenuation of 10dB at 7rad/sec

12 Design a maximally flat digitally LPF to meet the following specifications:

0.8 ≤ |H(e jw)| ≤ 1 ; 0 ≤ w ≤ π/4
|H(e jw) | ≤ 0.18 ; 0.75π ≤ w ≤ π
Use impulse variant transformation

p.q rp.stous 
Realize the system function no 
13
r-p.rour p.vsous
Using a) direct form I
b) cascade form and
c) parallel form

Given that !j  1U

  √2   1
14 represents a normalized second order low pass Butterworth
filter.
a) Plot 20 log |!WΩ| for different values of Ω. At what radian frequency Ω is the magnitude
down 3 dB? Down 20 dB?
b) Apply Low pass to high pass transformation, S→10/S, to the G(S) to obtain a new filter
H(S) and plot 20 log |WΩ| for different values of Ω. Does the new filter perform as
expected? At what radian frequency is the filter magnitude down 3 dB? Down 20 dB?
c) The transformation S→5 S/(S2 +50) is applied to the G(S) given resulting in new filter H(S).
What type of filter results and what are the critical frequencies?

15 Design a lowest order analog filter with maximally flat response in the pass band and an
acceptable attenuation of -2.5 dB at 15 radians per seconds. The attenuation in the stop band
should be more than -12 dB beyond 25 radians per seconds. Sketch pole zeros of the filter.

16 Show that the mapping function used in bilinear transformation satisfies all the requirements in
transforming analog filter to a digital filter effectively.

17 What is matched Z-Transform? Compare the matched Z-transform with impulse invariant
transformation

18 The square magnitude response of an analog Butterworth low pass filter is

%
|M j|  g >,
.%-a U b /
a. Determine the order of the filter.
b. Determine the cut-off frequency of the filter
c. Derive the transfer function of a normalized Butterworth filter and show the pole
locations in the S-plane

19 Design a Chebyshev type-1 analog low pass filter to meet the following specifications:
Pass band attenuation of 2 dB at 4 rad/sec
Stop band attenuation of 10 dB at 7 rad/sec.
Verify the design.
 Dr H S Prashanth, PESIT

20 Compare matched Z-Transform with impulse invariant transformation method for the transfer
g- 
function j  g-%g-&

21 Explain how an analog filter is mapped on to digital filter using impulse invariant transformation
method. What are the limitations of the method compared to bilinear transformation?

22 Design a digital band pass filter from 2nd order analog low pass Butterworth prototype filter using
bilinear transformation method. The lower and upper cut-off frequencies for band pass filter are
5YU and 7YU assuming T=2 sec. Obtain the difference equation representation of the filter.
12 12

23 Consider a difference equation   0.1   1  0.2   2  3   3.6   1 
0.6   2, obtain direct form I and cascade form realization.

24 The square magnitude response of an analog Butterworth low pass filter is

%
|M j|  g z
y%-a U b {
1. Determine the order of the filter.
2. Determine the cut-off frequency of the filter
3. Derive the transfer function of a normalized Butterworth filter and show the pole
locations in the S-plane

25 Design a Chebyshev type-1 analog low pass filter to meet the following specifications:
Pass band attenuation of 2 dB at 4 rad/sec
Stop band attenuation of 10 dB at 7 rad/sec.
Verify the design.

26 Compare matched Z-Transform with impulse invariant transformation method to convert analog
g- 
filter to a digital filter by considering the transfer function j  g-%g-&

27 Design a digital band pass filter from 2nd order analog low pass Butterworth prototype filter using
bilinear transformation method. The lower and upper cut-off frequencies for band pass filter are
5YU and 7YU assuming T=2 sec.
12 12