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NIGERIAN DEFENCE ACADEMY

FACULTY OF ENGINEERING & TECHNOLOGY
DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING
SECOND SEMESTER EXAMINATION 2018/2019 SESSION

COURSE CODE: ELE 442

COURSE TIFLE: DIGITAL SIGNAL PROCESSING
REGULAR COURSE: 67RC
TIME ALLOWED: 2 Hours
INSTRUCTION: ANSWER ANY THREE(3) QUESTIONS (show all workings in detail)
DATE:

QUESTION ONE
a. Give one example each of two-dimensional and three dimensional signal respectively (2Marks)
b. Briefly explain two advantages and one disadvantage of digital signal processing.
(3Marks)
c. Convolve x (n) and h(n) using the Matrix method

{
1
n , 0≤ n ≤ 6
x ( n )= 3
0 , elsew h ere

{
h ( n ) = 1 ,−2≤ n ≤ 2
0 ,elsew h ere
(9 Marks)
d. Apply the bilinear transformation method to find the digital equivalent of an analog filter given a
transfer function of:
1
H ( s )=
s
+1
Ωc
π
The cut-off frequency of the analog filter is Ωc =± and a sampling rate of 10 Hz is required. (11 Marks)
3

QUESTION TWO
a. In a tabular format, state five differences between finite impulse response (FIR) filters and infinite
impulse response filters (IIR). (5 Marks)
b. List the special DS processor hardware units. (3 Marks)
c. Find the finite impulse response (FIR) output for the sinusoidal input shown below:

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x ( n )=4+3 cos ( πn3 − π2 )+3 cos( 7 8πn )

The filter coefficients are given to be b k =[ 1 ,2 , 1 ] (17 Marks)
QUESTION THREE
a. State three (3) areas in engineering, where correlation can be applied. (3 Marks)
b. State the mathematical expression for:
i. The Bilateral Z-transform.
ii. The Unilateral Z-transform.
iii. The Inverse Z-transform (3 Marks)

c. With the aid of a diagram, briefly describe the architecture of a DS processor (3 Marks)
d. Given the transfer function shown below:
−1
1+ z
X ( z )= −1 −2
1−z +0.5 z
i. Plot the poles and zeros of X ( z ) in the complex z-plane (6 Marks)
ii. Determine the inverse transform of X ( z ) (10 Marks)

QUESTION FOUR
a. State the conditions for the Fourier transform of the analogue signal x (t) to exist. ( 3 Marks)
b. What is the reason for finding the Fourier transform of a signal, x (t)? (2 Marks)
c. State mathematically, the components of the Fourier transform of an analogue signal x (t). (2 Marks)
d. Explain how the Discrete Fourier Transform (DFT) relates to the Z-transform of a discrete-time
LTI system. (3
Marks)
Given the discrete-time signal x ( n )={ 1 , 2 ,3,4 }, derive the N-point Discrete Fourier Transform (DFT)
¿
e.
using the following:
i. The Discrete Fourier Transform’s (DFT’s) linear transformation property. (8 marks)
ii. Decimation in frequency FFT algorithm. (3.5 Marks)
iii. Use the Decimation-in-Time Inverse Fast Fourier Transform (IFFT) algorithm to
determine the Inverse Discrete Fourier Transform (IDFT) of the answer obtained in
question e above. (3.5 Marks)

QUESTION FIVE
a. Briefly explain the following concepts as relates to digital signal processing.
i. Sampling
ii. Quantization
iii Coding (3 Marks)
b. With the aid of diagrams, explain the concept of correlation of digital signals. (3 Marks)
c. Determine the steady state response of the system:
1
y ( n )= [ x ( n )−x ( n−2 ) ]
2

(π
)
If the input signal is x ( n )=5+3 cos n+60 ° + 4 sin( πn+ 45 °).
2
(8 Marks)

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d. A digital system is represented by the difference equation:
y ( n )=0.5 y ( n−2 )+ x(n−1).
If the system has a forced input expressed by x ( n )=( 0.5 )n u ( n ) and the initial conditions is given as
x (−1 )=−1 , y (−2 )=1 , y (−1 )=0.
Calculate the total response y (n) of the system for the points 0 ≤ n ≤3 (11 Marks)

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MARKING GUIDE
Question One
Question 1:
a) Two dimensional signal
i. Image 1 Mark
Three dimensional signals
i. Colour TV 1 Mark
b) Two Advantages of digital signal processing:
i. Flexibility: digital systems are programmable, modular and can be modified easily but a complete
redesign is needed to modify analog systems.
ii. Accuracy: It is very easy to determine the depth of accuracy needed when processing signals
digitally. Analog systems have very little tolerance and are very challenging to control in terms of
word length, floating points and similar factors.
iii. Storage: digital signals can be economically stored in magnetic and flash media without any
deterioration in the fidelity of the stored signal. Analog signals require very large storage spaces
due to their enormous bandwidth.
iv. Cost: digital systems in comparison with analog systems are generally cheaper to implement. This
may be because of the inherent ease in manufacturing digital systems and the added advantage of
being able to manipulate the algorithms that operate them.
v. Mathematical processing: It is very easy to manipulate digital signals. Complex mathematical
operations can be performed on digital signals with great precision. Analog signals do not lend
themselves easily to complex mathematical manipulations.
vi. Repeatability: It is easy to repeat digital operations with the same level of accuracy and obtain
exactly the same results. Analog systems degrade with time and may not be able to reproduce
exactly the same results for corresponding analog operations 2 Marks
Disadvantages of digital signal processing:
i. It is more complicated than ASP.
ii. It requires more components.
iii. Due to its increased number of components, it will cost more than using its ASP equivalence.
iv. Loss of some information in the natural signal due to the conversion from analog to digital and then
back to analog.
v. When the frequency of the natural signal becomes too high, it cannot be processed digitally
1Mark
c) For x (n):
1
Case 1: n ;0 ≤ n ≤6
3
1
When: n=0 ; x ( 0 )= × 0=0
3
1
n=1; x ( 1 )=
3
2
n=2; x ( 2 )=
3
n=3 ; x ( 3 )=1
4
n=4 ; x ( 4 )=
3

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5
n=5 ; x ( 5 )=
3
n=6 ; x ( n )=2 1 Mark
Case 2: 0 , elsew h ere

{ ¿ 1 2

3 3
4 5
∴ x ( n ) = … 0,0 , 0 , , , 1 , , , 2,0 , … .
3 3 }
For h ( n ) :
Case 1: 1 ;−2 ≤ n ≤2
When n=−2 ; h ( n )=1
n=−1 ; h ( n )=1
n=0 ; h ( n )=1
n=1; h ( n )=1
n=2; h ( n )=1 1 Mark
Case2: 0 , elsew h ere
∴ h ( n )= { ….0,0,1,1 , 1 ,1,1,0,0 … . }
¿

y ( n )=x ( n )∗h(n)
i.
x (n)
¿
0 1 2 1 4 5 2
3 3 3 3
1 0 1 2 1 4 5 2
3 3 3 3
1 0 1 2 1 4 5 2
h(n) 3 3 3 3
1¿ 0 1 2 1 4 5 2
3 3 3 3
1 0 1 2 1 4 5 2
3 3 3 3
1 0 1 2 1 4 5 2
3 3 3 3
5 Marks

y (−2) y (−1) y (0) y (1) y (2) y (3) y ( 4) y (5) y (6) y (7) y (8)
y (n) 0 1 1 2 10 5 20 6 5 11 2
3 3 3 3

{ 1 10
3
20
3
11
∴ y ( n )= ..0 , , 1,2 , ,5 , , 6,5 , , 2 , … .
3 3 }
1 Mark

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1 Mark

d) Prewarp the analog prototype critical frequency so as to coincide with the critical frequency of the
corresponding digital filter.
' w Ωc T
Ωc =tan =tan 1 Mark
2 2
π 1
Ωc =± ∧T = =0.1 sec 1 Mark
3 f

( )
π
× 0.1
' 3 1 Mark
∴ Ωc =tan =0.05241
2
Select the filter used and substitute the new cut-off frequency Ω'c

1 0.05241
H ( s )= =
s s+ 0.05241 2 Marks
+1
0.05241

Applying bilinear transformation: s →

2 z−1
T z +1 ( )
when T =0.1 1 Mark
0.05241 0.05241
H ( z )= =
( )
2 z−1
T z +1
+ 0.05241
z +1(
20 z−20
)
+ 0.05241
1

Mark

0.05241(z +1) 0.0026205( z+1)

H ( z )= = 2 Marks
20 z−20+ 0.05241 z−0.90738

Dividing numerator and denominator by1.26205 z, we have:

−1
0.0026205(1+ z )
H ( z )= −1 2 Marks
1−0.99738 z
Question Two
a.
IIR FILTER FIR FILTER
The impulse response of an IIR filter is The impulse response of an FIR filter is
infinite in duration. finite in duration.
The transfer function of an IIR filter has a The transfer function of an FIR filter does
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denominator. not have a denominator.

The difference equation of an IIR filter is The difference equation of an FIR filter is
RECURSIVE which means the current NON-RECURSIVE which means the current
output depends upon the previous output. output depends only upon the current and
previous input.
A designed IIR filter could be UNSTABLE. FIR filters are always STABLE because all
its poles are at the origin.
The IIR filter’s transfer function can be The transfer function can be expanded in a
expanded using a geometric series that is finite series.
infinite.
IIR filters can simulate a predefined analog FIR filters cannot simulate a predefined
filter. analog filter.
The phase response of an IIR filter is non- The phase response of FIR filters is linear.
linear.
IIR filters cannot be guaranteed to be stable. FIR filter design is stable. 5 Marks
b. The special DS processor hardware units are
i. Multiplier and Accumulator (MAC)
ii. Shifters
iii. Address Generators 3 Marks
c From the filter coefficients, it is clear that the impulse response is:
h ( n ) =δ ( n )+2 δ ( n−1 ) +δ (n−2) 1 Mark
Thus, the z-transform of the system function will be:
H ( z )=z { h(n) }=1+2 z−1 + z−2 1 Mark
Thus, the frequency response of the system is:
H ( z=e )=1+2 e +e
jω jω −j2ω
1 Mark
H ( e )=e e + 2 e + e
jw − jω jω − jω −j2ω − jω jw
=e (e +2+e ) − jω
1 Mark

Where e jω +e− jω =2 cosω

H ( e )=e ( 2+2 cosω )=|H ( e )|< H ( e )

jω − jω jω jω

¿|H ( e )|=2+2 cosω ;< H ( e )=−ω

jω jω
2 Marks
So, the magnitude of each component of the input is scaled by ¿ and the phase is added to
the phase of ¿ H ( e jω )
y ( n )=x ( n ) . H (e jω )
Firstly, y 1 ( n )=x 1 ( n ) .|H (e jω )|< H ( e jω )
For:
x 1 ( n )=4 ;ω=0
∴|H (e )| at ω=0 will be:
jω

¿ 2+2 cosω=2+2 cos 0=4 1 Mark

∴< H ( e ) at ω=0 will be :
jω

¿−ω=0 1 Mark
Thus, y 1 ( n )=( 4 × 4 ) <0=16 1 Mark

Secondly,

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y 2 ( n )=x 2 ( n )|H ( e jω )|< H ( e jω ) ∧x2 ( n )=3 cos ( πn3 − π2 )

π
|H (e jω)| at ω= 3 will be:
π
2+2 cosω=2+2 cos =3 1 Mark
3
π
¿ H ( e ) at ω= will be:
jω
3
−π
−ω= 1 Mark
3

Thus, y 2 ( n )=( 3 ×3 )< ( πn3 − π2 − π3 )=9 cos ⁡( πn3 − π2 − π3 )

y 2 ( n )=9 cos { π (n−1) π
3
−
2 } 1 Mark

Thirdly,
y 3 ( n )=x 3 ( n )| H ( e )|< H ( e ) ∧x 3 ( n )=3 cos
jω jω
( 78π )
7π
|H (e jω)| at ω= 8
will be:
7π
2+2 cosω=2+2 cos =0.152 1 Mark
8
7π
¿ H ( e ) at ω=
jω
will be:
8
−7 π
−ω= 1 Mark
8

Thus, y 3 ( n )=( 3 × 0.152 )< ( 78πn − 78π )=0.456 cos ⁡( 7 π (n−1)

8
)

y 3 ( n )=0.456 cos { 7 π (n−1)

8 } 1 Mark

Therefore, the total output is y n= y1 ( n ) + y 2 ( n ) + y 3 (n) 1 Mark

y ( n )=16+ 9 cos { π (n−1) π

3
− +0.456 cos
2 } 8 {
7 π ( n−1)
} 1 Mark

Question Three
a) Any 3 areas in engineering where correlation can be applied.
i) Radar
ii) Sonar
iii) digital communication
iv) geology
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v) seismology
vi) satellite communication 3 Marks
b) Mathematical expression for:
∞
i) X ( z )= ∑ x(n) Z−n – The Bilateral z-transform
n=∞
∞
+¿ ( z) = ∑ x (n) Z ¿
−n

ii) – The Unilateral z-transform

X n=0

❑
1
iii) x ( n )= ∮
2 πj c
n−1
X (z ) Z dz – The Inverse z-transform

W h ere :
∮ c →Integral over a closed contour within the ROC of X ( z )which encloses the origin.
x n → Discrete-time signal.
X ( z ) → Z-transform of the discrete-time signal x ( n )
Z →The complex variable.
x (n)→ The discrete time signal to be transformed. 3 Marks

c) To accelerate the execution speed of digital signal processing, DS processors are designed based on the
Harvard architecture, which originated from the Mark 1 relay-based computers built by IBM in 1944 at
Harvard University. This computer stored its instructions on punched tape and data using relay latches.
The DS processor has two separate memory spaces. One is dedicated to the program code, while the
other is employed for data. Hence, to accommodate two memory spaces, two corresponding address
buses and two data buses are used.
In this way, the program memory and data memory have their own connections to the program memory
bus and data memory bus, respectively. This means that the Harvard processor can fetch the program
instruction and data in parallel at the same time, the former via the program memory bus and the latter
via the data memory bus. 1 Mark

2 Marks
d.
i. Remove all negative powers by multiplying the numerator and denominator by the highest power of the
denominator ( z 2 ¿ :
2
z +z
X ( z )= 2 1 Mark
z −z+ 0.5
Divide both sides by z :
X ( z) z 2+ z 1
= 2 ×
z z −z +0.5 z

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X ( z) z
= 2
z z −1.5 z+ 0.5

X (z ) z +1
= 2 1 Mark
z z −z +0.5

Determine the poles by equating the denominator to zero

∴ poles are at P1 =0.5+ j 0.5 ; P2=0.5− j 0.5 2 Marks

Determine the zeros by equating the numerator to zero:

z +1=0 ;
∴ zerois z=−1 1 Mark

1 Mark
X (z) z +1 A B
ii) = 2 = +
z z −z +0.5 z−P1 z−P2

z +1= A ( z−P2 ) + B ( z−P1 )

z +1= Az−A P2+ Bz−B P 1
Equating coefficients of z :
A+ B=1
Equating constants:
− A P2−B P1=1
1+ P2
B=
P2−P1
1+ ( 0.5− j 0.5 )
B=
( 0.5− j 0.5 )−( 0.5+ j 0.5 )

B=0.5+ j 1.5 1 Mark

A=0.5− j1.5 1 Mark
X ( z ) 0.5− j 1.5 0.5+ j1.5
= +
z z−P1 z−P2
( 0.5− j 1.5 ) z ( 0.5+ j1.5 ) z
X ( z )= + 1 Mark
z−P 1 z −P 2

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Recall,

z−1
( AK
1−Pk z
−1
+
A ¿k
1−P ¿
k
) =2| A k|r nk cos ( β nk + α k ) u ( n )

If ROC is |z|>|P k|=r k

¿
Ak =A k =0.5− j1.5 1 Mark

P K =P¿k =0 .5+ j 0 .5 1 Mark

Expressing Ak ¿ PK in polar form we have:

(
j tan
−1 b
)
a+ jb=√ a +b . e2 2 a
=r e j (arg )
For Ak :a=0 . 5 ,b=−1 . 5

r =√ 0 .52 + (−1 . 5 ) =1 .58=| Ak|

2

arg=tan −1 ( ba )=tan ( −10 .5.5 )

−1

arg=−71. 57=α k

Ak in polar form will be:

Ak =0 .5+ j 1. 5=1 . 58 e− j 71 .57 2 Marks

For Pk :a=0 .5 , b=0 . 5

r =√ 0 .52 +0 . 52=0 . 7071=r k

arg=tan −1 ( 00 .. 55 )=45=β k

∴ P k in polar form will be:

j 45
Pk =0 .5+ j 0. 5=0 .7071 e 2 Marks

n
¿ 2. ( 1.58 )( 0.7071 ) cos ( 45 n−71.57 ) u ( n )

n
¿ 31 .6 ( 0 . 7071 ) cos ( 45 n−71. 57 ) u(n) 1 Mark

Question Four
a. These conditions are known as the Dirichlets conditions and its stated below

i. The will be absolutely integrable i.e.

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1 Mark

ii. The should have a finite number of maxima and minima within any finite interval 1
Mark

iii. can have a finite number of discontinuities within any interval. 1 Mark

b. The purpose for finding the Fourier Transform of a signal is to obtain the frequency spectrum of
the signal. The frequency spectrum of the signal consists of the Magnitude and phase of the signal. Hence the

Fourier transform of the signal can be obtained by 2 Mark

c. The component of the spectrum of the Fourier transform of an analog signal can be represented
mathematically as:

Magnitude = or 1 Mark

Phase = 1 Mark

d. The Z-Transform of N-point sequence is given by

1 Mark

Let be evaluated at N equally spaced points on a unit circle, i.e. at , therefore

But by the definition of N-point DFT, we get

1 Mark
Comparing both equations, we can deduce that

1 Mar k

e. Given the 4-point DFT

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i) Using the Discrete Fourier Transform’s (DFT’s) linear transformation property the formula for

DFT is given by . When we consider the linear transformation property,

the formula becomes where, .

Substituting into the formula, we have

1 Mark

Substituting the values when , we have:

3 Marks

Where Twiddle matrix which the value can be derived from a unity complex Fourier
transform -plane and is given below:

2 Marks

Therefore 1 Mark

Hence 1 Mark
ii) Finding the fourier transform with the decimation in frequency method, we have

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3.5 Marks
iii) Using the Decimation in Time to obtain the IFFT, we have

3.5 Marks
Question Five
a.
i. Sampling: This is the process of converting an analog signal into a discrete-time continuous
amplitude signal. It is a process that takes the highest frequency of the analog signal to
regenerate the digital signal by multiplying by a number minimum twice the highest
frequency, a phenomenon known as Shannon Sampling theorem. This process is done to
eliminate the problem of aliasing 1
mark
ii. Quantization: The process of converting a discrete-time continuous-amplitude signal into a
digital signal by expressing each sample value as a finite (instead of infinite) number of
digits is called quantization. 1 mark
iii. Coding: The coding process in an A/D converter gives a unique binary number to each
quantization level. If there are “L” levels, L different binary numbers are needed to
represent all quantized samples. Therefore if ‘b’ bits are used to represent each sample, then
b
2 different values can be coded. 1 mark
b. Correlation is a mathematical operation that closely resembles convolution in which 2 signals are
involved. The main objectives of performing correlation between 2 signals that are similar is to measure the
degree to which the signals are similar and thus to extract some information that depends on a large extent to
the application. Correlations of signals are often encountered in radar, sonar, digital communication, geology
and other areas of science and engineering.

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1 mark
A specially designed antenna transmits burst of radio waves energy in a selected direction. If the
propagating waves strikes an object i.e. The airplane in the diagram, a small fraction of the energy is
reflected towards a receiver, near the transmitter. The transmitted pulse is a specific shape that is selected
by the user, which can be a triangular wave or a square wave. The received signal will consist of parts:
a). A shifted and scaled version of the of the transmitting pulse,
b). Random noise, resulting from interfering radio waves, thermal noise in electronics, etc.
From the figure, the correlation of the transmitted signal to the received signal in relation to the noise
generated can be represented by the equation given below:
y ( n )=αx ( n−D ) +ω (n)
Where α = Attenuation Factor
x(n) = Input signal
D = Round Trip Delay
ω = Additive Noise
y(n) = Output Signal 2 marks
1
c. given that y ( n )= [ x ( n )−x ( n−2 ) ], Taking the Z-Transform of both sides, we have
2
−2
Y ( Z )=0.5 X ( Z )−0.5 X ( Z ) z ½ mark
Y ( Z )= X ( Z ) { 0.5−0.5 z } ½ mark
−2

Y (Z ) −2
But the transfer function H ( Z )= =0.5−0.5 z ½ mark
X (Z )
But H ( e )=H ( Z ) ¿ z=e
jω
jω ½ mark
Therefore, the equation becomes
H ( e )=0.5−0.5 e
jω −2 jω

H ( e )=0.5−0.5 ( cos 2 ω− j sin 2 ω ) ½ mark

jω

H ( e jω )=0.5−0.5 cos 2 ω− j 0.5 sin 2ω ½ mark

Therefore, the magnitude response is given as

|H ( e jω )|=√(0.5−0.5 cos 2 ω)2+(−0.5 sin 2 ω)2 … … … … .1 ½ mark
While the phase response is given as
−1 −0.5 sin 2 ω
∠ H ( e )=tan
jω
… … … … … … ..2½ mark
0.5−0.5 cos 2 ω
π
( )
If the input to the system is x ( n )=5+3 cos n+60 ° + 4 sin(πn+ 45 °), the steady state response y ss (n)
2
is obtained by:
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For the first part, when ω=0 , substituting into equation 1 and 2, we have
|H ( e jω )|=√(0.5−0.5 cos 2× 0)2+(−0.5 sin 2 ×0)2=√ 0+0=0 ½ mark
−0.5 sin 2 ×0
∠ H ( e jω )=tan−1 =0 ° ½ mark
0.5−0.5 cos 2 ×0
π
For the second part when ω=
2

√
2 2
π π
|H e |= (0.5−0.5 cos(2 × 2 )) +(−0.5 sin(2 × 2 )) =√12 −02=1 ½ mark
( jω
)
π
−0.5 sin 2 ×
2
∠ H ( e jω )=tan−1 =0 ° ½ mark
π
0.5−0.5 cos 2 ×
2
For the third part when ω=π
|H ( e jω )|=√( 0.5−0.5 cos 2 π )2+ (−0.5 sin 2× π )2 =√0+ 0=0 ½ mark
−1 −0.5 sin 2 × π
∠ H ( e )=tan
jω
=0 ° ½ mark
0.5−0.5 cos 2 × π
Therefore
y ss ( n )=(5 ×0)+¿

( )
π
y ss ( n )=3 cos n+60 ° . 1 marks
2
d). The homogenous solution y h (n) is obtained by making x ( n )=0
Therefore, from the equation
y ( n ) +0.75 y ( n−1 ) +0.125 y ( n−2 )=x ( n )
y ( n ) +0.75 y ( n−1 ) +0.125 y ( n−2 )=0
n
Putting y ( n )= λ , we have ½ mark
n n−1 n−2
λ + 0.75 λ +0.125 λ =0
λ ( λ + 0.75 λ+0.125 ) =0 ½ mark
n−2 2

λ n−2=0 , an d ( λ2 +0.75 λ+ 0.125 )=0

( λ 2+ 0.75 λ+0.125 ) =0
( λ 2+ 0.25 λ+0.5 λ+ 0.125 )=0
λ ( λ +0.25 ) +0.5 ( λ+ 0.25 )=0 ½ mark
( λ+ 0.5 ) ( λ+0.25 ) =0
λ 1=−0.5 , λ2=−0.251 mark
Therefore, y h ( n )=C 1 λn1 +C2 λ n2
n n
y h ( n )=C 1 (−0.5) +C2 (−0.25) ½ mark

The Particular solution is given by

x ( n )=0.5 n u ( n )
n
y p ( n )=K 0.5 u ( n ) ½ mark
Substituting into equation 1, we have
K 0.5n u ( n ) +0.75 K 0.5n−1 u ( n−1 )+ 0.125 K 0.5 n−2 u ( n−2 )=0.5n−1 u ( n−1 ) ½ mark
Placing n=2, so that none of the term vanishes, we have
K 0.52 u ( 2 ) +0.75 K 0.52−1 u ( 2−1 )+ 0.125 K 0.5 2−2 u ( 2−2 )=0.52−1 u ( 2−1 ) ½ mark
0.25 K ( 1 )+ 0.75 K 0.51 ( 1 ) +0.125 K 0.50 ( 1 ) =0.51 (1)
0.25 K +0.375 K +0.125 K=0.5
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0.75 K=0.5
K=0.667 ½ mark
Therefore,
y p ( n )=0.667 ¿
Hence the total response is given as:
y ( n )=C 1 (−0.5 )n +C2 (−0.25 )n +0.667 ¿

From initial conditions, the equation 1 becomes:

y ( n )=x ( n−1 )−0.75 y ( n−1 )−0.125 y ( n−2 )
Where x ( −1 ) =−1 , y ( −2 ) =2, y (−1 )=1
x ( 0 )=0.5 0 u ( 0 ) =1½ mark

When n = 0
y ( 0 )=x (−1 )−0.75 y (−1 )−0.125 y (−2 )
y ( 0 )=−1−0.75−0.25=−2 … … … … … … … … … … .3 ½ mark
When n = 1
y ( 1 )=x ( 0 )−0.75 y ( 0 )−0.125 y (−1 )
y ( 1 )=1−0.75 (−2 ) −0.125 ( 1 )
y ( 1 )=1+1.5−0.125=2.375 … … … … … … … ..4 ½ mark
Substituting into equation 2, we have:
n n
y ( n )=C 1 (−0.5 ) +C2 (−0.25 ) +0.667 ¿
When n = 0
0 0
y ( 0 )=C 1 (−0.5 ) +C2 (−0.25 ) +0.667 ¿
y ( 0 )=C 1 +C2 +0.667 … … … … … … … … … … ….5 ½ mark
When n = 1
1 1
y ( 1 )=C 1 (−0.5 ) +C 2 (−0.25 ) +0.667 ¿
y ( 1 )=−0.5 C 1−0.5 C2 +0.333 … … … … … … … … ..6 ½ mark
Equating the equation 3 and 5, we have
C 1+C 2=−2.667 … … … … … … … … … … 7 ½ mark
Equating the equation 4 and 6, we have
0.5 C 1+0.25 C 2=−2.042 … … … … … … … … 8 ½ mark
Solving equation 7 and 8 simultaneously, we have
C 1=−5.495∧C 2=2.828 ½ mark
Therefore, Total response is:
n n
y ( n )=−5.495 (−0.5 ) + 2.828 (−0.25 ) + 0.667 ¿
Hence y ( 0 )=−2, y ( 1 )=2.375 , y ( 2 )=−1.0312∧ y ( 3 )=0.7266 1 marks

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