School of Engineering

Department of Computer and Communication Engineering

Fall 2016
Course: EENG527 – Digital Signal Processing

Instructors: Dr. Ammar El Falou, Dr. Ayman Khalil, Dr. Hussein Hijazi, Dr. Mohamad Husseini,
Dr. Mohamad Raad

Final Examination
Date: 1 February 2017
Time: 120 minutes

Student Name: _________________________ Section:_________________

There are three Questions in the booklet each has several parts, please answer all parts of the 4 questions to
the best of your ability.

Marking Scheme:

Questions Weight Mark

Question 1 40 points
Question 2 28 points
Question 3 32 points
Total 100 points

1. This booklet contains 20 pages including this one. Make sure you have all these pages.
2. Closed Book Examination.
3. Do not take the staple out. The exam booklet must remain intact
4. The cheating penalty will be “F” in the exam

Good Luck

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Question 1: (40pts)

A- An LTI system H (z) is characterized by the system function given below.

( )
( )
( )( )

1) Indicate whether the system is FIR or IIR. Justify your answer. (2pts)

the system has poles ==> IIR system

2) Plot the pole-zero pattern. (3pts)

0.5

2
Im(z)

-0.5

-1

-1 -0.5 0 0.5 1 1.5 2 2.5 3 3.5 4

Re(z)

3) Specify the different possible ROC of H (z) and indicate the causality and the stability in each
case. Justify your answer. (5pts)

Poles: p1 = -0.5 (order = 1) ; p2 = 4 (order = 1) (1pts)

If system non-causal or two sided  ROC1: 0.5 < |z| < 4  system stable (unit circle included)

If system causal  ROC2: |z| > 4  system unstable (unit circle not included)

If system anti-causal  ROC3: |z| < 0.5  system unstable (unit circle not included)
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 4) Determine the gain K if ( ) . What can you say about causality and stability in this
case? (2pts)

H(2) = -K/5 = -1/5 ==> K= 1 and

Since H(z) is defined at z = 2  ROC: : 0.5 < |z| < 4  system non-causal and unstable

z 2  2z  1
B- Consider a system defined by G (z)  .
7
z - z 2
2

2
1- Find the difference equation that characterizes the system. (3pts)

z 2  2z  1 1  2z 1  z 2 Y (z )
G (z)   
7 7
z 2 - z  2 1 - z 1  2z 2 X (z )
2 2 (1pt)

y(n) = 7/2 y(n-1) +2y(n-2)+x(n) -2x(n-1) +x(n-2) (2pts)

2- Specify the ROC of G(z) and compute the inverse z-transform g(n) for the following cases:

a- The system is stable. (3pts)

z 2  2z  1 z 2 1 2z 1
G (z)     
 1 1 z4 1 1 1  4z 1
 z +   z  4 z + 1 + z
 2 2 2

Poles: p1 = -1/2 and p2 = 4

Stable ==> unit circle is included in ROC ==> ½ <|z| < 4 (1pt)
1 2z 1
G (z)   1
1 + z 1 1  4 z
1
2
g(n) = (-1/2)nu(n) -4n-1 u(-n) (2pts)

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 b- The system is causal (3pts)

Causal ==> |z| > 4 (1pt)

g(n) = (-1/2)nu(n) +4n-1 u(n-1) (2pts)

c- The system is anticausal (3pts)

anticausal ==> |z| < 1/2 (1pt)

g(n) = -(-1/2)nu(-n-1) -4n-1 u(-n) (2pts)

3- Assume that the system is causal and relaxed. Determine the response y(n) of the system to the
input x(n) = {3, 0, 1}. (3pts)

Y(z) =G(z)X(z) =3G(z) + z-2G(z) ==> y(n) = 3 g(n) + g(n-2)

= 3(-1/2)nu(n) +3*4n-1 u(n-1) + (-1/2)n-2u(n-2) +4n-3 u(n-3)

4- Assume now that the system is causal and nonrelaxed with the initial conditions ( ) ,
and ( ) . Determine the response y(n) of the system to the same input x(n) = {3, 0, 1}.
(4pts)

X+(z) = 1 -2 z-1 + 2 z-2 

( ) ( ) ( ) 1
( ) ( ) ( ) ( ) ( )
1+0.5z 1

y(n) = 3 g(n) + g(n-2) + (-1/2)nu(n)

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 C- Consider an LTI causal system defined by ( ) ( ) with ROC : |z| > 1.
1- Is this system stable ? (2pt)

Unit circle outside the ROC  unstable

( )
2- Let ( ) . Show that ( ) and specify its ROC.
( )

Hint : ( ) . (2pts)

( )
( ) = -z/(z2+1) = -z-1/(1+z-2) ROC : |z| > 1 ( same as V(z) )

3- By using the appendix, compute the inverse z-transform of W(z). (3pts)

1/(1+z-2) ROC : |z| > 1  cos[n π/2] u(n) (from the table)
W(z)  w(n) = - cos[(n-)π/2] u(n-1)

4- Deduce the inverse z-transform of V(z). (2pts)

( )
( )  w(n) = n v(n)

v(n) = w(n) / n = - cos[(n-)π/2] u(n-1) /n

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Question 2: (28pts)

A- A discrete sinusoidal signal x(n) of length L is obtained by sampling an analog signal xa(t) at a rate
Fs. The Fourier transform of x(n) is denoted by X(w).

1) What is the tool to obtain a discrete spectrum? How it can be obtained from X(w)? (3pts)

A discrete spectrum is obtained by using a N-point discrete Fourier Transfrom DFT.

The N-point DFT X(k) are obtained by evaluating X(w) at w = 2pi k / N where 0 ≤ k ≤ N-1

2) If L-point DFT is not sufficient, how we can enhance the picture of the discrete-spectrum in order to
be able to reconstruct the continuous spectrum X(w). (3pts)

We have to evaluate X(w) at more closely spaced frequencies. This can be obtained by increasing N
the length of the DFT, so by padding x(n) by N-L zeros.

B- Let ( ) { } be the input signal to a discrete-time filter with impulse response ( )

{ } and let ( ) be the corresponding output.

1) Give the following sequences:

a- ( ) (( )) ( ). (2pt)

x1(n) ={-1 , 1 -3 }

b- ( ) (( )) . (2pt)

x2(n) ={1 , -1, -3 }

c- ( ) (( )) . (2pt)

x3(n) ={-3 , 1 -1 }

d- ( ) (( )) .(2pt)

x4(n) ={1 , -3 , -1, 0 }

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 2) Find the 4-point DFT of ( ). Deduce the 4-point DFT of ( ) (3pt)

X(k) = -1 -3w4k + w42k where k = 0 ; 1 ; 2 ; 3 and w4 = exp(-jπ/2) = -j

X(k) = { -3 ; -2+3j ; 3 ; -2 -3j }

X4(k) = w4-2k X(4-k) = (-1)k X(4-k) = { -3 ; 2+3j ; 3 ; 2 -3j }

3) Determine the 4-point circular autocorrelation ( ) of ( ). (4pts)

Hint: ( ) ( ) ( )

rxx(n) = x(n) x*(-n) = [ ][ ] = [11 0 –2 0]

4) For which sizes of the DFT the circular convolution of ( ) and ( ) is equivalent to their linear
convolution? (2pts)

x(n) ==> N1 = 3
h(n) ==> N2 = 2
y(n) = x(n) * h(n) ==> N1+N2-1 = 4

linear conv is equivalent to circular conv for DFT of size N ≥ 4

5) Compute ( ) using:
a- Circular definition. (2pts)

Compute the circular convolution of length 4 between x(n) and h(n)

v(n) = x’(n) h’(n) = = [-1 ; -5 ; -5 ; 2]

[ ][ ]

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 b- DFT. (3pts)

X’(k) = -1 -3w4k + w42k where k = 0 ; 1 ; 2 ; 3

H’(k) = 1 +2w4k where k = 0 ; 1 ; 2 ; 3

Y(k) = X’(k)H’(k) = -1 -3w4k + w42k -2w4k -6w42k + 2w43k = -1 -5w4k -5w42k + 2w43k

y(n) = [-1 ; -5 ; -5 ; 2] (1pt)

Question 3: (32pts)

A- We would like to design digital filter with the following specifications:

( )
{ ( )
( )

1) Sketch the filter response and specify the type of this filter. (4pts)

The sketch should show the frequency response with a transition bands, locating wp, wc, ws and
intuitively the passband and the stopband ripples.

2) Determine the stopband and passband ripples. (3pts).

dp1 = 0.01 and dp 2 = 0.02. ds = 0.0025.

3) Determine the passband and the stopband cutoff frequencies. (2pts)

wp1 = 0.1π ws1 = 0.25π wp2 = 0.8π and ws2 = 0.6π

4) Determine the attenuation in dB in the stopband. (2pts).

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 ds (dB) = 20 log10(0.0025) = -52dB

5) Determine the width of the transition band ∆f. (3pts).

∆w1 = 0.15 π and ∆w2 = 0.2π  ∆w = min(0.15π , 0.2π ) = 0.15π

∆f = ∆w/2π = 0.075

6) Determine the cutoff frequencies. (2pts)

wc1 = wp1 + ∆w/2 = 0.1π – 0.075π = 0.025 π;

wc2 = wp2 - ∆w/2 = 0.8π + 0.075π = 0.875 π.

7) Using the table of windows in the appendix, what are the appropriate window(s) to design this filter?
Justify your answer. (2pts)

Since the stop band attenuation should be -52 dB or better, hence either Hamming or Blackman
windows could be used.

8) Using the window method, design a FIR filter with minimum length that satisfies the above
specifications. Give a complete expression for the impulse response h(n). Specify any tapered
window by its type – no need to give the formula of the window. (5pts)

Since we would like to use a minimum length filter, and since for Hamming window N = 3.3/∆f
while for Blackman N = 5.5/∆f, we choose Hamming window.

∆f = ∆w / 2π = 0.05.

Therefore N = 3.3 / 0.075 = 44.

The ideal filter:

( ) {

hd(n) = δ[n] - sin(0.875π[n-22])/(π[n-22]) + sin(0.025π[n-22])/(π[n-22]) for 0 ≤ n ≤43

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 h(n) = hd(n) . wHamming(n)

9) What is the effect of increasing the window length (N) on the width of the main lobe and the peak
side-lobe amplitude of the frequency response of the window? (2pt)

Increasing N will lead to narrower main lobe but higher peak side-lobe.

10) What is the influence of not using a window on the unit sample response of the filter? (2pt)

The unit sample response would be of infinite length. Using a window leads to a truncated unit
sample response.

B- A digital lowpass IIR filter is required to fulfill the specifications given below:

- 3-dB cutoff frequency 1.75 kHz

- Attenuation of 40 dB at 2.5 kHz

The filter is to be designed using Butterworth filter.

1) Find the minimum order of IIR Butterworth filter. (3pts)

Given:
- Stopband edge: 2500π of attenuation 40 dB
- Cutoff freq:

The Butterworth filter:

- Stopband attenuation =
( )

2) Determine the poles of the Butterworth filter and give its system function H(s). (2pts)

The poles of H(s) are the seven poles of H(s)H(-s) that lie the left half s-plane

( )
==> , k = 0, 1,…, 6

( ) ∏

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 Appendix

One-sided z-Transform:

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