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INTRODUCTION TO DIGITAL SIGNAL PROCESSING

1.0 What is Digital Signal Processing (DSP)


Signal is any variable that carries or contains some kind of information. Examples of the
types of signals are:
Speech signals telephony, radio
Biomedical signals brain signals, cardiac signals
Audio signals cd player, engine sound
Video and image television, radiography, camera
Radar signals range and bearing of distance target
Digital signal processing (DSP) is the processing of signals using digital technology. DSP
and analog signal processing are subfields of signal processing.
2.0 Why Digital Signal Processing
Why Process Signal:
Signals are corrupted by interference due to noise Remove interference and
noise.
To recover information present in the signal Enhance a signal.
To perform algorithms to obtain spectrum of the data, or transform signal into
more suitable form, or extract information from signals.
The goal of DSP is usually to Measure, Analyze or Filter continuous real-world analog
signals. In DSP, engineers usually study digital signals in many domains (discrete
transformations of signals) such as:
Time domain
Spatial domain (multi-dimensional signals)
Frequency domain (Frequency spectrum)
Autocorrelation domain
Wavelet domain, etc
They choose the domain to process a signal by making an educated guess or possibilities
of which domain best represents the characteristics of the signal.
3.0 Advantages and Disadvantages of DSP
Motivation of Going to Digital:
Easy to analyze - Using computers.
Efficient way of storing - Use of compact disc (CD, Hard Disk, Pendrive, etc).
Compact in size Digital processors are small in size, semiconductor technology.
Smaller size, lower cost, low power consumption, and higher speed.
System works better.
Advantages of DSP:
a) Easier Implementation of Adaptive Algorithms:
Once the signals are converted into a digital format they can be manipulated
mathematically. This gives the advantage that all the signals can be treated far more, and
this enables better filtering and general manipulation or transformations of the signals.
b) Programmability:
DSP systems can be programmed and reprogrammed to perform a variety of functions,
without modyfying the hardware.
c) Performance:
DSP can be used to perform functions not possible with analog signal processing. Any
complex filtering and algorithms can be implemented using DSP techniques.
Disadvantages of DSP:
a) Cost and Bandwidth Limitation Related to the Processor Cycle and Algorithm
Complexity:
At the present, fast ADCs and DACs are too expensive or do not have sufficient
resolution for wide bandwidth DSP applications. Currently, only specialized Ics can be
used to process signals in MHz range and these are quite expensive. Furthermore, most
DSP devices are still not fast enough and can only process signals of moderate
bandwidths. Bandwidths in the 100 MHz range are still processed by analog methods.
Nevertheles, DSP devices are becoming faster and faster.
4.0 DSP Structure
DSP systems use a DSP processor or other digital hardware, an analog-to-digital
converter (ADC) and a digital-to-analog (DAC) converter to replace analog devices such
as amplifiers, modulators and filters. The figure below shows a typical DSP system:
Digital
Processors
ADC DAC
Analog
signal
Analog
signal
Discrete-
Time
signal
Discrete-
Time
signal
The first step is to convert the signal from an analog to a digital form, by using an
ADC converter.
A processor performs digital operations based on a specific signal processing
algorithm implemented in software to process the digital signals.
The output digital signal sometime may be converted back to analog form using
the DAC.
Digital Processor Methodology:
HARDWARE
- FPGA
- ASIC
- Computer
- DSP Processor
- Microprocessor
- Microcontroller
SOFTWARE
Specific DSP algorithms required for DSP are sometimes can only be performed using
specialized microprocessor called Digital Signal Processor. This process signals in real
time.
5.0 Application of DSP
Image Processing pattern recognition, image enhancement, satellite weather
map
Instrumentation / Control spectrum analysis, noise reduction, data compression
Speech / Audio speech recognition, text to speech, digital audio, speech
systhesis
Military secure communication, radar processing
Telecommunications data communication, digital and cellular mobile phone,
spread spectrum
Biomedical patient monitoring, EEG brain mapper, ECG analysis, x-ray storage
and enhancement
Power power quality analysis and monitoring
A REVIEW OF SIGNALS AND SYSTEMS

1.0 Signals

A signal x(t) is a continuous time function if time t can be expressed as a real number. It can be
classified as periodic and aperiodic.

a) periodic signal is
x(t)=x(t+T) -<t< (1)

b) non periodic (aperiodic) signal is
x(t) defined within -T/2<t<T/2, and T< (2)

example of aperiodic signals

1) Delta function
2) Step function
3) Ramp function
4) Pulse function
5) Pulse sinusoid

1.1 Delta Function

1 ) ( = t x 0 = t
=0 elsewhere
) (t o = (3)

1.2 Step Function

1 ) ( = t x 0 > t
=0 0 s t
) (t u = (4)

1.3 Ramp Function

t t x = ) ( 0 > t
=0 0 s t
) (t r = (5)

1.4 Pulse Function

1 ) ( = t x
1 0
t t t s s
=0 elsewhere (6)

1.5 Pulse Sinusoid Function

) 2 cos( ) (
1
| t = t f t x
1 0
t t t s s
=0 elsewhere (7)

1.6 Energy and Power

The energy and power of a signal x(t) is defined as

(8)
) )

= =
2 /
2 /
2 /
2 /
2
| ) ( | ) ( * ) (
T
T
T
T
x
dt t x dt t x t x E

x
T
T
T
T
x
E
T
dt t x
T
dt t x t x
T
P
1
| ) ( |
1
) ( * ) (
1
2 /
2 /
2 /
2 /
2
= = =
) )

(9)

where T is the duration of the signal.

2.0 Systems

A system operates on an input signal and output the results.
Characteristics of systems are

1. time invariant
2. shift invariant
3. causal
4. stability
5. linear

2.1 Basic System Characteristics

Time and shift invariant means that the system characteristics and shift do not change with time.
Given a system impulse response h(t), the system is causal if

h(t)=0 t>0
=0 t<0 (10)

and stable if

(11) <
)


dt t h | ) ( |

2.2 Linearity

The basic system input-output characteristics is defined by the following transformation

)] ( [ ) ( t x T t y = (12)

If x
0
(t) and x
1
(t) represents two different inputs to the system, the system is considered linear if

)] ( ) ( [ )] ( [ )] ( [
1 0 1 0
t x t x T t x T t x T + = + (13)
) ( ) ( ) (
01 1 0
t y t y t y = +

The assumption of linearity also allows the following

)]] ( [ [ )]] ( [ [ ) ( t x S T t x T S t y = = (14)

where S[ ] and T[ ] are linear transformations.

2.3 Convolution

Many naturally occurring phenomena can be modeled as a convolution process. It is defined as

y(t)=h(t)*x(t) (15)
y(t)=x(t)*h(t) (16)

where h(t) and x(t) are the system impulse response and input signal respectively. If defined in
integral form, it can be expressed as

(17)
)


= d t x h t y ) ( ) ( ) (
(18)
)


= d t h x t y ) ( ) ( ) (

Example:


Consider the LTI system with an impulse response as

1 ) ( = t h 2 0 s s t
= 0 elsewhere

If the input to the system is given by

1 ) ( = t x 1 0 s s t
= 0 elsewhere

then the output of the system is

)


= t t d t h t x t y ) ( ) ( ) (


The convolution procedure described in graphical form is as shown below.



0 2
h(t)
t
0 1
x(t)
t









a) The definition of the system impulse response h(t) and the input signal x(t).


1 -2 0
1
1 0
1
-2 0
1
1
1 1
h(0-t)
t
x(t)
t

x(t)*h(0-t)






t


Then
)
= = 0 ) 0 ( * ) ( ) 0 ( t t t d h x y

b) The result at t=0.



1 0
h(1-t)
-1 1 0
x(t)
x(t)*h(1-t)





1 0 -1
t
t
t


Then
)
= = 1 ) 1 ( * ) ( ) 1 ( t t t d h x y



c) The result at t=1.


x(t)
h(2-t)
x(t)*h(2-t)
2

0
1

1 0
1
2 1 0
1




t
t
t


Then
)
= = 1 ) 2 ( * ) ( ) 2 ( t t t d h x y

d) The result at t=2.



x(t)
h(3-t) x(t)*h(3-t)


0 1 0 1 3 1 0
1
3
t
t
t
1
1








Then
)
= = 0 ) 3 ( * ) ( ) 3 ( t t t d h x y

e) The result at t=3.


0 1 2 3
y(t)
t
1








f) The overall results.

Figure 1 The steps and overall results for the computation of convolution.


3.0 Fourier Series

The Fourier series describes that any periodic signal can be defined as a weighted sum of
complex exponentials. The series can be defined in terms of the analysis and generation
functions

)

=
2 /
2 /
0
0
0
0
) 2 exp( ) (
1
T
T
k
dt t kf j t x
T
c t (19)
) 2 exp( ) (
0
t kf j c t x
k
k
t
_

=
= (20)

where f
0
is the fundamental frequency, T
0
is the period and the fundamental frequency is the
inverse of the period. The analysis function estimates the frequency content of a signal that is
represented by the coefficients c
k
. The generation function can be used to reconstruct the signal
based on the coefficients.

3.1 Characteristic of Fourier Series

Time domain
x(t)
continuous in time
periodic

Frequency domain
a
k
discrete frequency
aperiodic (non periodic)







The properties of the Fourier series are shown in Table 1.

Properties x(t) c
k
Differentiation
n
n
dt
t x d ) (
| |
kt f j
n
n
k
e
dt
d
c
0
2t

Integration
)

2 /
2 /
0
0
) (
T
T
dt t x dt e c
T
T
kt f j
k )

2 /
2 /
2
0
0
0
t

Time-shift ) (
d
T t x d
T f j
k
e c
0
2t

Frequency-shift
t f k j
e t x
0 1
2
) (
t

t f k c
k k 0 1 1
2 where
1
t e =
+

Even symmetry ) ( ) ( t x t x =
Odd symmetry ) ( ) ( t x t x =
Linearity ) ( ) ( t y t x +
k k
d c +
Product of two signals ) ( ) ( t y t x
_

l
l k l
d c
Convolution of two signals ) ( * ) ( t y t x
k k
d c
Parsevals theorem
| |
)

=
2 /
2 /
2
0
0
0
) (
1
T
T
x
dt t x
T
P
_

=
=
k
k x
c P
2


Table 1 Fourier series properties.


3.2 Sample Calculations for the Fourier Series

A periodic signal defined within a period T is

x(t)=1 -T
0
/4<t<T
0
/4
=0 -T
0
/2<t<-T
0
/4, T
0
/4<t<T
0
/2

Based on the analysis function, the Fourier series coefficients are


4 /
4 /
0
2
0
4 /
4 /
2
0 0
0
0
0
0
0
2
1
) 1 (
1
T
T
t kf j
T
T
t kf j
k
kf j
e
T
dt e
T
c

= =
)
t
t
t


(

+
=

0
4 / 2 4 / 2
0
2
1
0 0 0 0
kf j
e e
T
T kf j T kf j
t
t t


since f
0
T
0
=1, further simplifications results in


2 /
) 2 / sin(
2
1 ) 2 / sin(
k
k
k
k
c
k
t
t
t
t
= =


f
0
=1/T

Figure 2 Time domain and frequency domain representation of a signal

3.3 Derivation of the Parsevals Theorem

The Fourier series is an example of an orthogonal transformation. As a result, the energy is
conserved in both the time and frequency domain. If a periodic signal is defined as a Fourier
series

(21)
t kf j
k
k
e c t x
0
2
) (
t
_

=
=




The power of the signal is

dt e c e c
T
dt t x t x
T
P
t lf j
l
l
t kf j
k
k
T
T
T
T
x
*
1
) ( * ) (
1
0 0
0
0
0
0
2 2
2 /
2 /
0
2 /
2 /
0
(

= =
_ _
) )

=

t t
(22)

Rearranging the terms will result in

dt e e c c
T
T
T
t lf j t kf j
k
l
l
k
)
_ _

=
=
2 /
2 /
2 2 *
0
0
0
0 0
1
t t
(23)

Since the Fourier series is linear transformation, the integration and summation order can be
interchange that results in

dt e
T
c c dt e c c
T
k l
T
T
t f l k j
l k
t f l k j
k
l
l
T
T
k _ _
)
_ _
)

= =
2 /
2 /
) ( 2
0
* ) ( 2 *
2 /
2 /
0
0
0
0 0
0
0
1 1
t t

(24)

For k=l, the integration of the complex exponentials is

1
1
2 /
2 /
) ( 2
0
0
0
0
= =
)

dt e
T
T
T
t f l k j t
(25)

and is zero otherwise. Thus, the results is now

(26)
_ _

=
= =
k
k
k
k k
c c c
2 *

Thus, it is proven that relationship between the power in the time and frequency domain is


_ _
)

= = =
k
k
k
k k
T
T
x
c c c dt t x t x
T
P
2 *
2 /
2 /
0
0
0
) ( * ) (
1
(27)

4.0 Fourier Transform

The Fourier series is not practical because a signal that is periodic does not exist in practice. A
modification in the assumption of the Fourier series results in the Fourier transform that
applicable for practical signals.

4.1 Derivation of the Fourier Transform

By redefining the coefficients as

(28)
k
c T k X
0
) ( =

the Fourier series analysis and generation functions are now

)

=
2 /
2 /
2
0
0
0
) ( ) (
T
T
t kf j
dt e t x k X
t
(29)
t kf j
k
e k X
T
t x
0
2
0
) (
1
) (
t
_

=
= (30)

By making the period T
0
large, the result is


0
T
from discrete to continuous variable f kf
0
for the generation function
)
_


The Fourier series has now become the Fourier transform that is defined in terms of the analysis
and generation function as

)

= dt e t x f X
ft j t 2
) ( ) ( (31)
)


= df e f X t x
ft j t 2
) ( ) ( (32)
4.2 Characteristic of Fourier Transform

Time domain
x(t)
continuous in time
aperiodic

Frequency domain
X(f)
continuous frequency
aperiodic (non periodic)

The properties of the Fourier transform are shown in Table 2.

Properties x(t) X(f)
Differentiation in time
n
n
dt
t x d ) (
| |
ft j
n
n
e
dt
d
f X
t 2
) (
Integration in time
)

2 /
2 /
) (
T
T
dt t x dt e f X
T
T
ft j
)

2 /
2 /
2
) (
t

Time-shift ) (
d
T t x
d
fT j
e f X
t 2
) (


Frequency-shift
t f j
e t x
1
2
) (
t

) (
1
f f X
Linearity ) ( ) ( t y t x + ) ( ) ( f Y f X +
Product of two signals ) ( ) ( t y t x
)


dv v f Y v X ) ( ) (
Convolution of two signals ) ( * ) ( t y t x ) ( ) ( f Y f X
Time scaling ) (at x
|
.
|

\
|
a
f
X
a
1

Parsevals theorem
| |
)


= dt t x E
x
2
) ( | |
)


= df f X E
x
2
) (

Table 2 Fourier transform properties.

4.3 Example of Fourier Transform

A signal is defined as

) 2 cos( ) (
1
t f t x t = -T/2<t<T/2
= 0 elsewhere


x(t)
T/2
-T/2

Figure 3 Time domain representation of signal.

The analysis function is


)

= dt e t x f X
ft j t 2
) ( ) (

The signal x(t) can be represented in terms of complex exponentials


2
) 2 cos( ) (
1 1
2 2
1
t f j t f j
e e
t f t x
t t
t

+
= =

Substituting the signal into the analysis function result in

dt e e dt e e
dt e t f f X
ft j
T
T
t f j ft j
T
T
t f j
ft j
T
T
t t t t
t
t
2
2 /
2 /
2 2
2 /
2 /
2
2
1
2 /
2 /
1 1
2
1
2
1
) 2 cos( ) (

) )
)
+ =
=


dt e dt e
T
T
t f f j
T
T
t f f j
) )

+ =
2 /
2 /
) ( 2
2 /
2 /
) ( 2
1 1
2
1
2
1
t t


2 /
2 /
1
) ( 2
2 /
2 /
1
) ( 2
) ( 2 2
1
) ( 2 2
1
1 1
T
T
t f f j
T
T
t f f j
f f j
e
f f j
e

+
+

+
(

=
t t
t t


(

+
+
(

=
+ +
+

) ( 2 2
1
) ( 2 2
1
1
2 / ) ( 2 2 / ) ( 2
1
2 / ) ( 2 2 / ) ( 2
1 1
1 1
f f j
e e
f f j
e e
T f f j T f f j
T f f j T f f j
t
t
t t
t t


The equation is further simplified by using the complex exponential representation for the sine
function


T f f
T f f T
T f f
T f f T
f X
) (
) ) ( sin(
2 ) (
) ) ( sin(
2
) (
1
1
1
1
+
+
+

=
t
t
t
t




Figure 4 Frequency domain representation of signal.

If T=> then

X(f) = (f-f
1
)+ (f+f
1
)



Figure 5 Frequency domain representation as T approaches infinity.


4.4 Fourier Transform and Time Domain Convolution

If the input-output relationship is a convolution, then the relationship can be expressed as
multiplication in the frequency domain. The input and output of a system h(t) is x(t) and y(t)
respectively. When expressed in terms of the Fourier transform is

(33)
) ) )

= = dt e d t x h dt e t y f Y
ft j ft j t t

2 2
) ( ) ( ) ( ) (

By changing variables t
1
=t-, t=t
1
+, and dt
1
=dt, the above equation becomes

) ) ) )


+
=
1
2 2
1 1
) ( 2
1
1 1
) ( ) ( ) ( ) ( dt d e e t x h dt d e t x h
ft j f j t f j

t t t

1
2
1
2
1
2 2
1
1 1
) ( ) ( ) ( ) ( dt e t x d e h dt d e e t x h
ft j f j ft j f j
) ) ) )



= =
t t t t

(34)

Based on the definition of the Fourier transform, the equation is further simplifies as

) ( ) ( ) ( f X f H f Y = (35)

This shows that time-domain convolution is equal to multiplication in the frequency domain.
Similarly, a multiplication in time-domain results in convolution in frequency domain.

Example

If a system h(t) has Fourier transform that is defined as

H(f)=1 -1000sfs1000
=0 elsewhere

it is desired to determine the output of the system y(t) when subjected to an input
) 1500 2 cos( ) 750 2 cos( ) ( t t t x t t + = s s T T t T , 2 / 2 /
=0 elsewhere

The Fourier transform of x(t) is

) 1500 (
2
1
) 1500 (
2
1
) 750 (
2
1
) 750 (
2
1
) ( + + + + + = f f f f f X o o o o

The Fourier transform of the output is

) 750 (
2
1
) 750 (
2
1
) ( ) ( ) ( + + = = f f f X f H f Y o o

By using the generation function, the output in time is


) )

+ + = = dt ft j f f dt ft j f Y t y ) 2 exp( ) 750 (
2
1
) 750 (
2
1
) 2 exp( ) ( ) ( t o o t

Using the properties of the delta function, the output is simplified as

) 750 2 cos( ) ( t t y t =

4.5 Derivation of the Parsevals Theorem

Similar to the Fourier series, the Parsevals theorem is also applicable to the Fourier transform.
Assuming the signal aperiodic, the energy is


) )
= =
2 /
2 /
2 /
2 /
2
) ( * ) ( ) (
T
T
T
T
x
dt t x t x dt t x E (36)

When expressed in terms of the analysis function of the Fourier transform, the energy can be
expressed as

(37) dt dv vt j v X df ft j f X
T
T
* ) 2 exp( ) ( ) 2 exp( ) (
2 /
2 /
(
(

(
(

=
) ) )


t t

The linearity of the Fourier transform allows changing the order of integration that results in

dfdv dt t v f j v X f X dt dfdv vt j ft j v X f X
T
T
) ) ) ) )


(
(

=
(
(

=
2 /
2 /
) ) ( 2 exp( ) ( * ) ( ) 2 exp( ) 2 exp( ) ( * ) ( t t t
(38)

Since the terms within the square brackets is non zero for f=v, then the equation simplifies as

(39) dfdv v X f X
) )


= ) ( * ) (

The functions X(f) and X(v) are essentially the same. Thus, the above equation becomes


) ) ) )


= = = df f X df f X f X dfdv v X f X
2
) ( ) ( * ) ( ) ( * ) ( (40)

Thus, it is shown that the energy in time and frequency is


) )


= = df f X df t x E
T
T
x
2
2 /
2 /
2
) ( ) ( (41)

By dividing both sides of the equation by the signal duration T, the theorem is extended to
power.
5.0 Laplace Transform

The Laplace transform originated from the Fourier transform. As known, the Fourier transform
enables us to understand the behaviour of a system in the frequency domain by allowing a signal
x(t) to be represented as a continuous sum of complex exponential. Thus, the introduction of the
convergence factor into the Fourier transform integral leads to the Laplace transform.
5.1 Derivation of Laplace Transform
Recalling the definition of the Fourier transform pair for a signal x(t),

dt e t x f X
ft j
)

=
t 2
) ( ) ( (42)

df e f X t x
ft j
)


=
t 2
) ( ) ( (43)

Defining a function y(t) as

(44) ) ( ) ( t x e t y
t o
=

The term is introduced known as the convergence factor. An appropriate choice for
t
e
o
o will
make y(t) absolutely convergent and its Fourier transform will exist for a large number of
functions. By substituting Equation 45 into Equation 43, the Fourier transform for y(t) is given
by

dt e t y f Y
ft
)

=
t 2
) ( ) (
(45)
)


+
= dt e t x
t f j ) 2 (
) (
t o

The equation can be rewritten as

(46)
)


+
= + dt e t x f j X
t f j ) 2 (
) ( ) 2 (
t o
t o

A variable s is used to represent f j t o 2 + is defined

f j s t o 2 + =

It is customary to call s as the complex frequency variable as f t 2 appears in the imaginary part
of the complex variable s.
Since , the Equation 46 can be written as jdf ds =
(47)
)

= dt e t x s X
st
) ( ) (

)
+

=
t o
t o
t
2
2
) (
2
1
) (
j
f j
st
ds e f X
j
t x (48)

The Equations 47 and 48 are known as the bilateral Laplace Transform. Since we consider a
causal system, unilateral Laplace transform are usually used.

(49)
)

=
0
) ( ) ( dt e t x s X
st
ds e f X
j
t x
f j
st
)
+
=
t o
t
2
0
) (
2
1
) ( (50)













5.2 Laplace Transform Properties

The properties of the Laplace transform are shown in Table 3.

Properties x(t) X(s)
Linearity
) ( ) (
2 2 1 1
t x a t x a + ) ( ) (
2 2 1 1
s X a s X a +
Shift in time ) ( t t x
) (s X e
st

Shift in s-plane
) (t x e
t o

) ( a s X
Time scaling ) (at x
) (
1
a
s
X
a

Differentiation
dt
dx

) 0 ( ) (

x s sX
Integration
)

t
d x t t ) (
)

+
0
) (
) (
t t d x
s
s X

Multiplication by t ) (t tx
ds
s dX ) (

Division by t
t
t x ) (

)

s
du u X ) (
Convolution in time
t t t d t x x ) ( ) (
2 1



) ( ) (
2 1
s X s X
Convolution in frequency
2 1
x x
du u s X u X
j
j c
j c
) ( ) (
2
1
2 1

)
+

t

Initial value theorem
) 0 (
+
x
) ( lim s sX
s

Final value theorem
) ( lim t x
t
) ( lim
0
s sX
s
(Valid only if sX(s)
has all its poles in
the left half of the
s-plane.)

Table 3 Laplace transform properties

5.3 Laplace Transform Examples

This section show some examples of calculations involving the Laplace transform.

A signal is defined as


at
e t x

= ) ( 0 > t
= 0 0 < t

) )

= =
0 0
) exp( ) ( ) ( dt e at dt e t x s X
st st

+
+
= =
)
0
) (
0
) (
) ( s a
e
dt e
t s a
t s a


) (
1
) (
exp
) (
0 ) ( ) (
a s s a s a
e
s a s a
+
=
+
+
+
=
+ +


A sinusoidal signal is expressed as

) 2 cos( ) (
1
t f t x t = 0 > t
= 0 0 < t


) )

= =
0
1
0
) 2 cos( ) ( ) ( dt e t f dt e t x s X
st st
t

By defining the cosine function in terms of complex exponentials, the Laplace transform is


) )

+ =
0
2
0
2
1 1
2
1
2
1
) ( dt e e dt e e s X
st t f j st t f j t t

) )

+ =
0
) 2 (
0
) 2 (
1 1
2
1
2
1
dt e dt e
t s f j t s f j t t

+
+

=
0
1
) 2 (
0
1
) 2 (
) 2 ( 2
1
) 2 ( 2
1
1 1
s f j
e
s f j
e
t s f j t s f j
t t
t t

) 2 ( 2
1
) 2 ( 2
1
) 2 ( 2
1
) 2 ( 2
1
1
0 ) 2 (
1
) 2 (
1
0 ) 2 (
1
) 2 (
1 1 1 1
s f j
e
s f j
e
s f j
e
s f j
e
s f j s f j s f j s f j
+
+
+

=
+ +
t t t t
t t t t
) 2 )( 2 (
) 2 ( ) 2 (
2
1
) 2 (
1
2
1
) 2 (
1
2
1
1 1
1 1
1 1
s f j s f j
s f j s f j
s f j s f j +
+ +
=
+
+

=
t t
t t
t t


( ) ( )
2
1
2
1 1 1 1
2
) 2 )( 2 ( ) 2 )( 2 (
2
2
1
f s
s
f j s f j s
s
s f j s f j
s
t
t t t t
+
=
+
=
+

=

( ) ( )
2
1
2
2
) (
f s
s
s X
t +
=

5.4 Solution of Differential Equation and Transfer Function

The Laplace transform can be used to determine the solution to differential equations. Given
that the input-output relationship of a linear time-invariant system is as follows

) ( ) ( ) 2 ( ) ( ) 1 ( ) (
2
2
t x t y a t y
dt
d
a t y
dt
d
= + + (51)

By taking the Laplace transform, the above equation becomes

(52) ) ( ) ( ) 2 ( ) ( ) 1 ( ) (
2
s X s Y a s sY a s Y s = + +

Putting all the s terms together results in

| | ) ( ) ( ) 2 ( ) 1 (
2
s X s Y a s a s = + +
) (
) 2 ( ) 1 (
1
) (
2
s X
a s a s
s Y
(

+ +
= (53)

The transfer function of the system obtained by taking the ratio of Y(s) and X(s) is


) 2 ( ) 1 (
1
) (
) (
) (
2
a s a s s X
s Y
s H
+ +
= = (54)

For a more general case, the transfer function is in the form of


_
_
=



=
+ + + +
+ + + +
=
N
n
n N
N
n
n N
N N N
N N N
s n a
s n b
N a s a s a s
N b s b s b s b
s H
0
) (
0
) (
2 1
2 1
) (
) (
) ( ...... ) 2 ( ) 1 (
) ( ...... ) 2 ( ) 1 ( ) 0 (
) ( (55)

where N is the polynomial order. The transfer function when factorized in terms of its roots is


( )( ) ( )
( )( ) ( )
( )
( )
[
[
=
=
+
+
=
+ + +
+ + +
=
N
n
N
n
n s
n s
N s s s
N s s s
s H
0
0
) (
) (
) ( ....... ) 1 ( ) 0 (
) ( ....... ) 1 ( ) 0 (
) (
o
|
o o o
| | |
(56)

The roots in the numerator |(0), -|(1) -|(N) are referred as the zeros while the roots in the
denominator o(0), -o(1) -o(N) are the poles.

The s-plane refers to the domain where all the possible values of H(s) occur. If the polynomial
order is 2 with real poles and zeros, the positions of the poles and zeros in the s-plane is shown
in Figure 6(a). Similar example for complex poles and zeroes are shown in Figure 6(b). For
causal systems, the system is stable if the poles are on the right hand side of the s-plane.

4
4
je je
o
o







(a) Real poles and zeros. (b) Complex poles and zeros.

Figure 6 Pole and zero plots.
5.5 The Inverse Laplace transform

By taking the inverse Laplace transform, the system impulse response h(t) is obtained from H(s).
If the following transfer function is used as example


( )( ) ) 1 ( ) 0 (
1
) 2 ( ) 1 (
1
) (
) (
) (
2
o o + +
=
+ +
= =
s s a s a s s X
s Y
s H (57)

the application of the partial fraction expansion results in


( )( ) ( ) ( ) ) 1 ( ) 0 ( ) 1 ( ) 0 (
1
) (
1 0
o o o o +
+
+
=
+ +
=
s
A
s
A
s s
s H (58)

Since
( )
at
e t h
a s
s H

=
+
= ) (
1
) ( , then the system impulse response obtained from the
transfer function is

(59)
t t
e A e A t h
) 1 (
1
) 0 (
0
) (
o o
+ =

If the poles are complex conjugate pole pairs, then the transfer function is

( )( ) | o | o j s j s a s a s s X
s Y
s H
+ + +
=
+ +
= =
1
) 2 ( ) 1 (
1
) (
) (
) (
2
(60)

By applying the partial fraction expansion, the transfer function is


( )( ) ( ) ( ) | o | o | o | o j s
A
j s
A
j s j s
s H
+
+
+ +
=
+ + +
=
1 0
1
) ( (61)

The resulting system impulse response is

(62)
t j t t j t
e e A e e A t h
| o | o
+ =
1 0
) (

If A
0
=A
1
, then the system impulse response is in the form of

(63) ) cos( 2 ) (
0 0 0
t e A e e A e e A t h
t t j t t j t
|
o | o | o
= + =

Thus, o contributes to the exponential decay in the amplitude level and | describes the
sinusoidal behavior of the signal.




5.6 Inverse Laplace Transform Examples

A system is defined by a transfer function


( ) 1
1
) (
+
=
s
s H

If the input to the system is

t
e t x
1 . 0
) (

= 0 > t
= 0 0 < t

then it is the objective to determine the output of the system. The Laplace transform of the input
signal is


( ) 1 . 0
1
) (
+
=
s
s X

and the resulting output signal is


( )( ) 1 . 0 1
1
1 . 0
1
1
1
) ( ) ( ) (
+ +
= |
.
|

\
|
+
|
.
|

\
|
+
= =
s s s s
s X s H s Y

1 . 0 1
1 0
+
+
+
=
s
A
s
A


The constants A
0
and A
1
can be determined by applying partial fraction expansion

( ) ( )
( )( )
1 . 1
9 . 0
1
1 . 0 1
1
1 . 0 1
1
1 ) ( 1
1
1
0
= =
+
=
+ +
+ = + =
=
=
s
s
s s
s s Y s A
( ) ( )
( )( )
1 . 1
9 . 0
1
1 1 . 0
1
1 . 0 1
1
1 . 0 ) ( 1 . 0
1 . 0
1
1
= =
+
=
+ +
+ = + =
=
=
s
s
s s
s s Y s A

The output signal when expressed in terms of the Laplace transform is


1 . 0
1 . 1
1
1 . 1
) (
+
+
+

=
s s
s Y

and its inverse is


t t
e e t y
1 . 0 1
1 . 1 1 . 1 ) (

+ = 0 > t
= 0 0 < t

A transfer function of a system that has a complex conjugate pole pair is defined as


( )( ) 100 2 5 . 0 100 2 5 . 0
1
) (
t t j s j s
s H
+ + +
=

It is desired to determine the system impulse response. Solving the partial fraction expansion
results in


( )( ) 100 2 5 . 0 100 2 5 . 0 100 2 5 . 0 100 2 5 . 0
1
) (
1 0
t t t t j s
A
j s
A
j s j s
s H
+
+
+ +
=
+ + +
=

( ) ( )
( )( )
) 100 2 5 . 0 (
) 100 2 5 . 0 (
0
100 2 5 . 0 100 2 5 . 0
1
100 2 5 . 0 ) ( 100 2 5 . 0
t
t
t t
t t
j s
j s
j s j s
j s s H j s A
+ =
+ =
+ + +
+ + = + + =

100 4
1
100 2 5 . 0 100 2 5 . 0
1
t t t j j j
=
+
=

( ) ( )
( )( )
) 100 2 5 . 0 (
) 100 2 5 . 0 (
1
100 2 5 . 0 100 2 5 . 0
1
100 2 5 . 0 ) ( 100 2 5 . 0
t
t
t t
t t
j s
j s
j s j s
j s s H j s A
=
=
+ + +
+ = + =


100 4
1
100 2 5 . 0 100 2 5 . 0
1
t t t j j j
=
+ + +
=

The resulting transfer function becomes


100 2 5 . 0
1
100 4
1
100 2 5 . 0
1
100 4
1
) (
t t t t j s j j s j
s H
+
|
|
.
|

\
|
+
|
|
.
|

\
|
+ +
|
|
.
|

\
|
=

The system impulse response is


t j t t j t
e e
j
e e
j
t h
100 2 5 . 0 100 2 5 . 0
100 4
1
100 4
1
) (
t t
t t

|
|
.
|

\
|
+
|
|
.
|

\
|
=
) 100 2 sin(
100 2
1
5 . 0
t e
t
t
t

|
.
|

\
|
= 0 > t
= 0 0 < t

5.7 The Laplace Transform and its relation to the Fourier transform

The complex variable s in the Laplace transform consist of a real component o and complex
component j2tf. By making the real part zero, the Laplace transform becomes the Fourier
transform. For example, a transfer function is

( )
2 2
) 100 2 (
1
) (
t + +
=
s s
s H (64)

By making the substituting s= j2tf, the Fourier transform of the transfer function is


( ) ( ) ( ) ( ) ( ) ( ) ( ) f j f f j f
f H
t t t t t t 2 2 100 2
1
100 2 2 2
1
) (
2 2 2 2
+
=
+ +
= (65)

The magnitude and phase based on the above equation is


( ) ( ) ( ) ( )
2
2
2 2
2 2 100 2
1
) (
f f
f H
t t t +
= (66)

( ) ( )
|
|
.
|

\
|

2 2
1
2 100 2
2
tan ) (
f
f
f
t t
t
| (67)

The frequency response when represented as a function of magnitude and phase is shown in
Figure 7.



Figure 7 Frequency response represented as a function of magnitude and phase.

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