EC 35 SIGNALS AND SYSTEMS 3 1 0 4

AIM
To study and analyse characteristics of continuous, discrete signals and systems.
OBJECTIVES
To study the properties and representation of discrete and continuous signals.
To study the sampling process and analysis of discrete systems using ztransforms.
To study the analysis and synthesis of discrete time systems.
1. CLASSIFICATION OF SIGNALS AND SYSTEMS 9
Continuous time signals (CT signals), discrete time signals (DT signals) - Step, Ramp,
Pulse, Impulse, Exponential, Classification of CT and DT signals - periodic and periodic,
random singals, CT systems and DT systems, Basic properties of systems - Linear Time
invariant Systems and properties.
2. ANALYSIS OF CONTINUOUS TIME SIGNALS 9
Fourier series analysis, Spectrum of C.T. singals, Fourier Transform and Laplace
Transform in Signal Analysis.
3. LINEAR TIME INVARIANT CONTINUOUS TIME SYSTEMS 9
Differential equation, Block diagram representation, Impulse response, Convolution
integral, frequency response , Fourier and Laplace transforms in analysis, State variable
equations and matrix representation of systems
4. ANALYSIS OF DISCRETE TIME SIGNALS 9
Sampling of CT signals and aliasing, DTFT and properties, Z-transform and properties of
Z-transform.
5. LINEAR TIME INVARIANT - DISCRETE TIME SYSTEMS 9
Difference equations, Block diagram representation, Impulse response, Convolution
sum,
LTI systems analysis using DTFT and Z-transforms , State variable equations and matrix
representation of systems.
Tutorial = 15
Total No of periods: 45 + 15 = 60
TEXT BOOK:
1. Allan V.Oppenheim, S.Wilsky and S.H.Nawab, Signals and Systems, Pearson
Education, 2007.
2. Edward W Kamen & Bonnies Heck, Fundamentals of Signals and Systems,
Pearson Education, 2007.
REFERENCES:
1. H P Hsu, Rakesh Ranjan Signals and Systems, Schaums Outlines, Tata McGraw
Hill, Indian Reprint, 2007
2. S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing, McGraw Hill
International/TMH, 2007.
3. Simon Haykins and Barry Van Veen, Signals and Systems John Wiley & sons , Inc,
2004.
4. Robert A. Gabel and Richard A.Roberts, Signals & Linear Systems, John Wiley, III
edition, 1987.
5. Rodger E. Ziemer, William H. Tranter, D. Ronald Fannin. Signals & systems, Fourth
Edition, Pearson Education, 2002.






UNIT I
CLASSIFICATION OF SIGNALS AND SYSTEMS
Continuous time signals (CT signals), discrete time signals (DT signals)
A signal is a function representing a physical quantity or variable, and typically it
contains information about the behavior or nature of the phenomenon. For instance, in a
RC circuit the signal may represent the voltage across the capacitor or the current flowing
in the resistor. Mathematically, a signal is represented as a function of an independent
variable t. Usually t represents time. Thus, a signal is denoted by x(t).
A. Continuous-Time and Discrete-Time Signals:
A signal x(t) is a continuous-time signal if t is a continuous variable. If t is a discrete
variable, that is, x(t) is defined at discrete times, then x(t) is a discrete-time signal. Since a
discrete-time signal is defined at discrete times, a discrete-time signal is often identified as
a sequence of numbers, denoted by {x,) or x[n], where n = integer. Illustrations of a
continuous-time signal x(t) and of a discrete-time signal x[n] are shown in Fig. 1-1.



A discrete-time signal x[n] may represent a phenomenon for which the independent
variable is inherently discrete. For instance, the daily closing stock market average is by its
nature a signal that evolves at discrete points in time (that is, at the close of each day). On
the other hand a discrete-time signal x[n] may be obtained by sampling a continuous-time
signal x(t) such as

or in a shorter form as


where we understand that

and x
n
s are called samples and the time interval between them is called the sampling
interval. When the sampling intervals are equal (uniform sampling), then


where the constant T, is the sampling interval.
B. Analog and Digital Signals:
If a continuous-time signal x(l) can take on any value in the continuous interval (a, b),
where a may be - 03 and b may be + m, then the continuous-time signal x(t) is called an
analog signal. If a discrete-time signal x[n] can take on only a finite number of distinct
values, then we call this signal a digital signal.
C. Real and Complex Signals:
A signal x(t) is a real signal if its value is a real number, and a signal x(t) is a complex
signal if its value is a complex number. A general complex signal x(t) is a function of the
form
where x,( t ) and x2( t ) are real signals and
Note that in Eq. (I.l) t represents either a continuous or a discrete variable.
D. Deterministic and Random Signals:
Deterministic signals are those signals whose values are completely specified for any
given time. Thus, a deterministic signal can be modeled by a known function of time I .
Random signals are those signals that take random values at any given time and must be
characterized statistically. Random signals will not be discussed in this text.
E. Even and Odd Signals:
A signal x ( t ) or x[n] is referred to as an even signal if
x ( - t ) = x ( r )
x [ - n ] = x [ n ]
A signal x ( t ) or x[n] is referred to as an odd signal if
x ( - t ) = - x ( t )
x [ - n ] = - x [ n ]
Examples of even and odd signals are shown in Fig. 1-2
. Any signal x(t) or x[n] can be expressed as a sum of two signals, one of which is even
and one of which is odd. That is,

where xe(t) = $ { x ( t ) + x ( - t ) ] even part of x ( t )
xe[n] = i { x [ n ] + x [ - n ] ) even part of x [ n ] (1.5)
x0(t) = $ { x ( t ) - x ( - t ) ) odd part of x(t )
x0[n] = $ { x [ n ]- x [ - n ] ) odd part of x [ n ] ( 1.6 )
Note that the product of two even signals or of two odd signals is an even signal and
that the product of an even signal and an odd signal is an odd signal.
F. Periodic and Nonperiodic Signals:
A continuous-time signal x ( t ) is said to be periodic with period T if there is a positive
nonzero value of T for which
x(t + T ) = x ( t ) all t (1.7)
An example of such a signal is given in Fig. 1-3(a). From Eq. (1.7) or Fig. 1-3(a) it follows
That
for all t and any integer m. The fundamental period T, of x ( t ) is the smallest positive
value of T for which Eq. (1.7) holds. Note that this definition does not work for a constant

signal x(t) (known as a dc signal). For a constant signal x(t) the fundamental period is
undefined since x(t) is periodic for any choice of T (and so there is no smallest positive
value). Any continuous-time signal which is not periodic is called a nonperiodic (or
aperiodic ) signal.
Periodic discrete-time signals are defined analogously. A sequence (discrete-time
signal) x[n] is periodic with period N if there is a positive integer N for which
x[n + N] =x[n] all n (1.9)
An example of such a sequence is given in Fig. 1-3(b). From Eq. (1.9) and Fig. 1-3(b) it
follows that
for all n and any integer m. The fundamental period No of x[n] is the smallest positive
integer N for which Eq. (1.9) holds. Any sequence which is not periodic is called a
nonperiodic (or aperiodic sequence.
Note that a sequence obtained by uniform sampling of a periodic continuous-time
signal may not be periodic (Probs. 1.12 and 1.13). Note also that the sum of two
continuous-time periodic signals may not be periodic but that the sum of two periodic
sequences is always periodic .
G. Energy and Power Signals:
Consider v(t) to be the voltage across a resistor R producing a current dt). The
instantaneous power p( t) per ohm is defined as

Total energy E and average power P on a per-ohm basis are

For an arbitrary continuous-time signal x(t), the normalized energy content E of x(t) is
defined as


The normalized average power P of x(t) is defined as

Similarly, for a discrete-time signal x[n], the normalized energy content E of x[n] is
defined as


The normalized average power P of x[n] is defined as

Based on definitions (1.14) to (1.17), the following classes of signals are defined:
1. x(t) (or x[n]) is said to be an energy signal (or sequence) if and only if 0 < E < m, and
so P = 0.
2. x(t) (or x[n]) is said to be a power signal (or sequence) if and only if 0 < P < m, thus
implying that E = m.
3. Signals that satisfy neither property are referred to as neither energy signals nor power
signals.
Note that a periodic signal is a power signal if its energy content per period is finite, and
then the average power of this signal need only be calculated over a period.
A. The Unit Step Sequence:
The unit step sequence u[n] is defined as

which is shown in Fig. 1-10(a). Note that the value of u[n] at n = 0 is defined [unlike the
continuous-time step function u(f) at t = 01 and equals unity. Similarly, the shifted unit step
sequence ii[n - k] is defined as

which is shown in Fig. 1-lO(b).

B. The Unit Impulse Sequence:
The unit impulse (or unit sample) sequence 6[n] is defined as

which is shown in Fig. 1-ll(a). Similarly, the shifted unit impulse (or sample) sequence
6[n - k] is defined as

which is shown in Fig. 1-1 l(b).

C. Complex Exponential Sequences:
The complex exponential sequence is of the form

Again, using Euler's formula, x[n] can be expressed as

Thus x[n] is a complex sequence whose real part is cos Ron and imaginary part is sin
Ron
. General Complex Exponential Sequences:
The most general complex exponential sequence is often defined as

where C and are in general complex numbers. Note that Eq. (1.52) is the special case
of
Eq. (1.57) with C = 1 and

Real Exponential Sequences:
If C and a in Eq. (1.57) are both real, then x[n] is a real exponential sequence. Four
distinct cases can be identified: a > 1,0 < a < 1, - 1 < a < 0, and a < - 1. These four real
exponential sequences are shown in Fig. 1-12. Note that if a = 1, x[n] is a constant
sequence, whereas if a = - 1, x[n] alternates in value between +C and -C.
D. Sinusoidal Sequences:
A sinusoidal sequence can be expressed as

If n is dimensionless, then both R, and 0 have units of radians. Two examples of
sinusoidal sequences are shown in Fig. 1-13. As before, the sinusoidal sequence in Eq.
(1.58) can be expressed as

As we observed in the case of the complex exponential sequence in Eq. (1.52), the same G.
observations [Eqs. (1.54) and (1.5611 also hold for sinusoidal sequences. For instance, the
sequence in Fig. 1-13(a) is periodic with fundamental period 12, but the sequence in Fig.
l-13( b) is not periodic.
SYSTEMS AND CLASSIFICATION OF SYSTEMS
A. System Representation:
A system is a mathematical model of a physical process that relates the input (or
excitation) signal to the output (or response) signal.
Let x and y be the input and output signals, respectively, of a system. Then the system
is viewed as a transformation (or mapping) of x into y. This transformation is represented
by the mathematical notation

where T is the operator representing some well-defined rule by which x is transformed
into y. Relationship (1.60) is depicted as shown in Fig. 1-14(a). Multiple input and/or
output signals are possible as shown in Fig. 1-14(b). We will restrict our attention for the
most part in this text to the single-input, single-output case

B. Continuous;Time and Discrete-Time Systems:
If the input and output signals x and p are continuous-time signals, then the system is
called a continuous-time system [Fig. I - 15(a)]. If the input and output signals are discrete-
time signals or sequences, then the system is called a discrete-time system [Fig. I - 15(b)].


Causal and Noncausal Systems:
A system is called causal if its output y ( t ) at an arbitrary time t = t,, depends on only
the input x ( t ) for t It o. That is, the output of a causal system at the present time depends
on only the present and/or past values of the input, not on its future values. Thus, in a
causal system, it is not possible to obtain an output before an input is applied to the
system. A system is called noncausal if it is not causal. Examples of noncausal systems are

Note that all memoryless systems are causal, but not vice versa
. Linear Systems and Nonlinear Systems:
If the operator T in Eq. (1.60) satisfies the following two conditions, then T is called a
linear operator and the system represented by a linear operator T is called a linear system:
1. Additivity:
Given that Tx, = y, and Tx, = y,, then

for any signals x, and x2.
2. Homogeneity (or Scaling):

for any signals x and any scalar .
Any system that does not satisfy Eq. (1.66) and/or Eq. (1.67) is classified as a
nonlinear system. Equations (1.66) and ( 1.67) can be combined into a single condition as

where 1 and 2 are arbitrary scalars. Equation (1.68) is known as the superposition
property. Examples of linear systems are the resistor [Eq. (1.6111 and the capacitor [Eq.
( 1.62)]. Examples of nonlinear systems are

Note that a consequence of the homogeneity (or scaling) property [Eq. (1.6711 of linear
systems is that a zero input yields a zero output. This follows readily by setting = 0 in
Eq.(1.67). This is another important property of linear systems.
F. Time-Invariant and Time-Varying Systems:
A system is called rime-inuariant if a time shift (delay or advance) in the input signal
causes the same time shift in the output signal. Thus, for a continuous-time system, the
system is time-invariant if


for any real value of T. For a discrete-time system, the system is time-invariant (or
shift-incariant ) if

for any integer k. A system which does not satisfy Eq. (1.71) (continuous-time system) or
Eq. (1.72) (discrete-time system) is called a time-varying system. To check a system for
time-invariance, we can compare the shifted output with the output produced by the
shifted input (Probs. 1.33 to 1.39).
G. Linear Time-Invariant Systems
If the system is linear and also time-invariant, then it is called a linear rime-invariant
(LTI) system.
H. Stable Systems:
A system is bounded-input/bounded-output (BIBO) stable if for any bounded
input x defined by

the corresponding output y is also bounded defined by

where k , and k, are finite real constants. Note that there are many other definitions of
stability. (See Chap. 7.)


PROPERTIES OF CONTINUOUS-TIME LTI SYSTEMS
A. Systems with or without Memory:
Since the output y(t) of a memoryless system depends on only the present input x(t),
then, if the system is also linear and time-invariant, this relationship can only be of the
form

where K is a (gain) constant. Thus, the corresponding impulse response h(f) is simply

Therefore, if h(tJ # 0 for I,, # 0, the continuous-time LTI system has memory.
B. Causality:
As discussed in Sec. 1.5D, a causal system does not respond to an input event until that
event actually occurs. Therefore, for a causal continuous-time LTI system, we have

Applying the causality condition (2.16) to Eq. (2.101, the output of a causal continuous-time
LTI system is expressed as

Alternatively, applying the causality condition ( 2.16) to Eq. (2.61, we have

Equation (2.18) shows that the only values of the input x(t) used to evaluate the output
y( t ) are those for r 5 t.
Based on the causality condition (2.161, any signal x(t) is called causal if

and is called anticausal if

Then, from Eqs. (2.17), (2. I8), and (2. Iga), when the input x(t) is causal, the output y(t )
of a causal continuous-time LTI system is given by


C. Stability:
The BIBO (bounded-input/bounded-output) stability of an LTI system (Sec.
1.5H) is readily ascertained from its impulse response. It can be shown (Prob. 2.13) that a
continuous-time LTI system is BIBO stable if its impulse response is absolutel integrable,
that is,

PROPERTIES OF DISCRETE-TIME LTI SYSTEMS
A. Systems with or without Memory:
Since the output y[n] of a memoryless system depends on only the present input x[n],
then, if the system is also linear and time-invariant, this relationship can only be of the form

where K is a (gain) constant. Thus, the corresponding impulse response is simply

Therefore, if h[n,] # 0 for n, # 0, the discrete-time LTI system has memory
B. Causality:
Similar to the continuous-time case, the causality condition for a discrete-time LTI
system is

Applying the causality condition (2.44) to Eq. (2.391, the output of a causal discrete-time
LTI system is expressed as

Alternatively, applying the causality condition (2.44) to Eq. (Z..V), we have


Equation (2.46) shows that the only values of the input x[n] used to evaluate the output
y[n] are those for k I n.
As in the continuous-time case, we say that any sequence x[n] is called causal if

and is called anticausal if


Then, when the input x[n] is causal, the output y[n] of a causal discrete-time LTI
system is given by


C. Stability:
It can be shown (Prob. 2.37) that a discrete-time LTI system is BIB0 stable if its
impulse response is absolutely summable, that is,







UNIT II
Analysis of continuous time signals
Fourier series analysis:
Fourier series: a complicated waveform analyzed into a number of
harmonically related sine and cosine functions
A two parts tutorial on Fourier series. In the first part an example is used
to show how Fourier coefficients are calculated and in a second part you may
use an applet to further explore Fouries series of the same function.

Fourier series may be used to represent periodic functions as a linear
combination of sine and cosine functions. If f(t) is a periodic function of period T,
then under certain conditions, its Fourier series is given by:

where n = 1 , 2 , 3 , ... and T is the period of function f(t). a
n
and b
n
are called
Fourier coefficients and are given by



Example : Find the Fourier series of the periodic function f(t) defined by

Solution to the above example

Coefficient a
0
is given by



Coefficients a
n
is given by



And coefficients b
n
is given by



A computation of the above coefficients gives

a
0
= 0 , a
n
= 0 and b
n
= [ 2 / (n*pi) ] [ 1 - cos (n pi) ]

Note that cos (n pi) may be written as

cos (n pi) = (-1)
n


and that b
n
= 0 whenever n is even.

The given function f(t) has the following Fourier series

Continuous Time Fourier Transform
The Fourier expansion coefficient ( in OWN) of a periodic signal
is



and the Fourier expansion of the signal is:



which can also be written as:



where is defined as



When the period of approaches infinity , the periodic signal
becomes a non-periodic signal and the following will result:
- Interval between two neighboring frequency components becomes zero:



- Discrete frequency becomes continuous frequency:



- Summation of the Fourier expansion in equation (a) becomes an integral:



the second equal sign is due to the general fact:



- Time integral over in equation (b) becomes over the entire time axis:



In summary, when the signal is non-periodic , the Fourier
expansion becomes Fourier transform. The forward transform (analysis) is:



and the inverse transform (synthesis) is:



Note that is denoted by in OWN.
Comparing Fourier coefficient of a periodic signal with with Fourier spectrum of
a non-periodic signal :



we see that the dimension of is different from that of :



If represents the energy contained in the kth frequency component of a periodic
signal , then represents the energy density of a non-periodic signal
distributed along the frequency axis. We can only speak of the energy contained in
a particular frequency band :



Note on notations:
The spectrum of a time signal can be denoted by or to emphasize the fact
that the spectrum represents how the energy contained in the signal is distributed as a
function of frequency or . Moreover, if is used, the factor in front of
the inverse transform is dropped so that the transform pair takes a more symmetric form.
On the other hand, as Fourier transform can be considered as a special case of Laplace
transform when the real part of the complex argument is zero:



it is also natural to denote the spectrum of by (in OWN).
Example 0:
Consider the unit impulse function:







Example 1:
If the spectrum of a signal is a delta function in frequency domain
, the signal can be found to be:



i.e.,



Example 2:





The spectrum is



This is the sinc function with a parameter , as shown in the figure.

Note that the height of the main peak is and it gets taller and narrower as gets
larger. Also note



When approaches infinity, for all , and the spectrum becomes



Recall that the Fourier coefficient of is



which represents the energy contained in the signal at (DC component at zero
frequency), and the spectrum is the energy density or distribution
which is infinity at zero frequency.
The integral in the above transform is an important formula to be used frequently later:



which can also be written as



Switching and in the equation above, we also have



representing a superposition of an infinite number of cosine functions of all frequencies,
which cancel each other any where along the time axis except at where they add
up to infinity, an impulse.
Example 3:




The spectrum of the cosine function is










The spectrum of the sine function



can be similarly obtained to be



Again, these spectra represent the energy density distribution of the sinusoids, while the
corresponding Fourier coefficients



and



represent the energy contained at frequency .


Inverse Transforms
If we have the full sequence of Fourier coefficients for a periodic signal, we can
reconstruct it by multiplying the complex sinusoids of frequency
0
k by the weights X
k

and summing:



We can perform a similar reconstruction for aperiodic signals


These are called the inverse transforms.

Fourier Transform of Impulse Functions
Find the Fourier transform of the Dirac delta function:



Find the DTFT of the Kronecker delta function:

=
=
1
0
0
) (
p
k
n ik
k
e X n x
e

=
=
k
t ik
k
e X t x
0
) (
e

=
t
t
e
e e
t
d e X n x
n i
) (
2
1
) (

= e e
t
e
d e X t x
t i
) (
2
1
) (
1 ) ( ) ( ) (
0
= = = =

e e e
o e
i t i t i
e dt e t dt e t x X
1 ) ( ) ( ) (
0
= = = =

e e e
o e
i
n
n i
n
n i
e e n e n x X
1 ) ( ) ( ) (
0
= = = =

e e e
o e
i
n
n i
n
n i
e e n e n x X

The delta functions contain all frequencies at equal amplitudes.
Roughly speaking, thats why the system response to an impulse input is important: it
tests the system at all frequencies.

Laplace Transform

Lapalce transform is a generalization of the Fourier transform in the sense that it allows
complex frequency whereas Fourier analysis can only handle real frequency. Like
Fourier transform, Lapalce transform allows us to analyze a linear circuit problem, no
matter how complicated the circuit is, in the frequency domain in stead of in he time
domain.
Mathematically, it produces the benefit of converting a set of differential equations into
a corresponding set of algebraic equations, which are much easier to solve. Physically, it
produces more insight of the circuit and allows us to know the bandwidth, phase, and
transfer characteristics important for circuit analysis and design.
Most importantly, Laplace transform lifts the limit of Fourier analysis to allow us to
find both the steady-state and transient responses of a linear circuit. Using Fourier
transform, one can only deal with he steady state behavior (i.e. circuit response under
indefinite sinusoidal excitation).
Using Laplace transform, one can find the response under any types of excitation (e.g.
switching on and off at any given time(s), sinusoidal, impulse, square wave excitations,
etc.



















Laplace Transform

Application of Laplace Transform to Circuit Analysis







UNIT III
LINEAR TIME INVARIANT CONTINUOUS TIME SYSTEMS
Differential equation, Block diagram representation, Impulse response, Convolution
integral, frequency response, State variableequations and matrix representation of
systems.
System:
A system is an operation that transforms input signal x into output signal y.


LTI Systems
Time Invariant
X(t) y(t) & x(t-to) y(t-to)
Linearity
a1x1(t)+ a2x2(t) a1y1(t)+ a2y2(t)
a1y1(t)+ a2y2(t)= T[a1x1(t)+a2x2(t)]
Meet the description of many physical systems
They can be modeled systematically
Non-LTI systems typically have no general mathematical procedure to
obtain solution




Differential equation:
This is a linear first order differential equation with constant coefficients
(assuming a and b are constants)

The general nth order linear DE with constant equations is


Linear constant-coefficient differential equations
In RC circuit




To introduce some of the important ideas concerning systems specified by linear
constant-coefficient differential equations ,let us consider a first-order differential
equations:


Block diagram representations
Block diagram representations of first-order systems described by differential and
difference equations



Impulse Response

(t) h(n)=H[ (t)]

A very important way to analyse a system is to study the output signal when a unit
impulse signal is used as an input.
This impulse response signal can be used to infer properties about the systems structure
(LHS of difference equation or unforced solution).
The system impulse response, h(t) completely characterises a linear, time invariant
system
LTI System
H[.]
Properties of System Impulse Response
Stable
A system is stable if the impulse response is absolutely summable

Causal
A system is causal if
h(t)=0 when t<0
Finite/infinite impulse response
The system has a finite impulse response and hence no dynamics in y(t) if there exists
T>0, such that:
h(t)=0 when t>T
Linear
ad(t) ah(t)
Time invariant
d(t-T) h(t-T)


Convolution Integral
An approach (available tool or operation) to describe the input-output relationship
for LTI Systems


In a LTI system
d(t) h(t)
Remember h(t) is T[d(t)]
Unit impulse function the impulse response
It is possible to use h(t) to solve for any input-output relationship
Any input can be expressed using the unit impulse function



Convolution Integral - Properties
Commutative
) ( * ) ( ) ( * ) ( t x t h t h t x =


Associative )] ( * ) ( [ * ) ( ) ( * )] ( * ) ( [
2 1 2 1
t h t h t x t h t h t x =
t t o t d t x t x ) ( ) ( ) ( =

<

= t
t h ) (
Distributive )] ( * ) ( [ )] ( * ) ( [ )] ( ) ( [ * ) (
2 1 2 1
t h t x t h t x t h t h t x + = +
Thus, using commutative property:




State variables and Matrix representation
State variables represent a way to describe ALL linear systems in terms of a
common set of equations involving matrix algebra.
Many familiar properties, such as stability, can be derived from this common
representation. It forms the basis for the theoretical analysis of linear systems.
State variables are used extensively in a wide range of engineering problems,
particularly mechanical engineering, and are the foundation of control theory.
The state variables often represent internal elements of the system such as
voltages across capacitors and currents across inductors.
They account for observable elements of the circuit, such as voltages, and also
account for the initial conditions of the circuit, such as energy stored in
capacitors. This is critical to computing the overall response of the system.
Matrix transformations can be used to convert from one state variable
representation to the other, so the initial choice of variables is not critical.
Software tools such as MATLAB can be used to perform the matrix
manipulations required.
Let us define the state of the system by an N-element column vector, x(t):




Note that in this development, v(t) will be the input, y(t) will be the
output, and x(t) is used for the state variables.
Any system can be modeled by the following state equations:
This system model can handle single input/single output systems,
or multiple inputs and outputs.
The equations above can be implemented using the signal flow
graph shown to the below
t t t t t t d t x h d t h x t x ) ( ) ( ) ( ) ( ) ( = =

| |
t
N
N
t x t x t x
t x
t x
t x
t ) ( ) ( ) (
) (
) (
) (
) (
2 1
2
1

= x



Consider the CT differential equations:

A second-order differential equation requires two state variables:

We can reformulate the differential equation as a set of three equations:



We can write these in matrix form as:




This can be extended to an N
th
-order differential equation of this type:



The state variables are defined as


The resulting state equation is
) ( ) (
) ( ) ( ) (
) ( ) (
) ( ) (
) ( ) (
1
0
1
0
1
1
3 2
2 1
t x t y
t v b t x a t x
t x t x
t x t x
t x t x
N
i
i i N
N N
=
+ =
=
=
=

=
+

Matrix representation
) ( ) ( ) ( ) (
0 0 1
t v b t y a t y a t y = + +
) ( ) ( ) ( ) (
2 1
t y t x t y t x = =
) ( ) (
) ( ) ( ) ( ) (
) ( ) (
1
0 2 1 1 0 2
2 1
t x t y
t v b t x a t x a t x
t x t x
=
+ =
=

| |

) (
) (
0 1 ) (
) (
0
) (
) ( 1 0
) (
) (
2
1
0 2
1
1 0 2
1
t x
t x
t y
t v
b t x
t x
a a t x
t x

( ) ( )
) ( ) ( ) (
0
1
0
t v b t y a t y
N
i
i
i
N
= +

=
( ) N i t y t x
i
i
..., , 2 , 1 , ) (
) 1 (
= =

| | 0 0 0 0 1
0
0
0
1 0 0 0
1 0 0
0 0 1 0
0 1 2 1 0
= =

D C
B A

b a a a a
N














UNIT IV

ANALYSIS OF DISCRETE TIME SIGNALS

SAMPLING OF CT SIGNALS AND ALIASING,DTFT AND PROPERTIES,Z-
TRANSFORM AND PROPERTIES OF Z-TRANSFORM
SAMPLING
Sampling theory
Let x(t) be a continuous signal which is to be sampled, and that sampling is performed by
measuring the value of the continuous signal every T seconds, which is called the
sampling interval. Thus, the sampled signal x[n] given by:
x[n] = x(nT), with n = 0, 1, 2, 3, ...
The sampling frequency or sampling rate f
s
is defined as the number of samples obtained
in one second, or f
s
= 1/T. The sampling rate is measured in hertz or in samples per
second.
The frequency equal to one-half of the sampling rate is therefore a bound on the highest
frequency that can be unambiguously represented by the sampled signal. This frequency
(half the sampling rate) is called the Nyquist frequency of the sampling system.
Frequencies above the Nyquist frequency f
N
can be observed in the sampled signal, but
their frequency is ambiguous. That is, a frequency component with frequency f cannot be
distinguished from other components with frequencies Nf
N
+ f and Nf
N
f for nonzero
integers N. This ambiguity is called aliasing. To handle this problem as gracefully as
possible, most analog signals are filtered with an anti-aliasing filter (usually a low-pass
filter with cutoff near the Nyquist frequency) before conversion to the sampled discrete
representation.
The theory of taking discrete sample values (grid of color pixels) from functions
defined over continuous domains (incident radiance defined over the film plane)
and then using those samples to reconstruct new functions that are similar to the
original (reconstruction).
Sampler: selects sample points on the image plane
Filter: blends multiple samples together


Sampling theory
Sampling Theorem:
bandlimited signal can be reconstructed exactly if it is sampled at a rate atleast twice the
maximum frequencycomponent in it."
Consider a signal g(t) that is bandlimited.
Sampling theory




The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly from
its samples it has to be sampled ata rate fs _ 2fm.
The minimum required sampling rate fs = 2fm is called nyquist rate


Aliasing
Aliasing is a phenomenon where the high frequency components of the sampled signal
interfere with each other because of inadequate sampling
s
< 2
m
.




Aliasing leads to distortion in recovered signal. This is the reason why sampling
frequency should be atleast twice the bandwidth of the signal.
DISCRETE TIME FOURIER TRANSFORM
In mathematics, the discrete-time Fourier transform (DTFT) is one of the specific
forms of Fourier analysis. As such, it transforms one function into another, which is
called the frequency domain representation, or simply the "DTFT", of the original
function (which is often a function in the time-domain). But the DTFT requires an input
function that is discrete. Such inputs are often created by sampling a continuous function,
like a person's voice.
Given a discrete set of real or complex numbers: (integers), the discrete-
time Fourier transform (or DTFT) of is usually written:

Often the sequence represents the values (aka samples) of a continuous-time
function, , at discrete moments in time: , where is the sampling interval
(in seconds), and is the sampling rate (samples per second). Then the DTFT
provides an approximation of the continuous-time Fourier transform:

To understand this, consider the Poisson summation formula, which indicates that a
periodic summation of function can be constructed from the samples of function
The result is:






(Eq.2)
The right-hand sides of Eq.2 and Eq.1 are identical with these associations:


comprises exact copies of that are shifted by multiples of
s
and combined
by addition. For sufficiently large
s
, the k=0 term can be observed in the region [
s
/2,

s
/2] with little or no distortion (aliasing) from the other terms.
Inverse transform
The following inverse transforms recover the discrete-time sequence:




The integrals span one full period of the DTFT, which means that the x[n] samples are
also the coefficients of a Fourier series expansion of the DTFT. Infinite limits of
integration change the transform into a continuous-time Fourier transform [inverse],
which produces a sequence of Dirac impulses. That is:

Properties
This table shows the relationships between generic discrete-time Fourier transforms. We
use the following notation:
- is the convolution between two signals
- is the complex conjugate of the function x[n]
- represents the correlation between x[n] and y[n].

Property
Time domain Frequency domain
Remark
s
Linearity



Shift in time


integer k
Shift in
frequency
(modulation
)

real
number a
Time
reversal



Time
conjugation


Time
reversal &
conjugation


Derivative
in frequency



Integral in
frequency


Convolve in
time



Multiply in
time



Correlation


Parseval's
theorem



SYMMETRY PROPERTIES
The Fourier Transform can be decomposed into a real and imaginary part or into an even
and odd part.

or

Time Domain

Frequency Domain





Z-transforms
Definition: The Z transform of a discrete-time signal x(n) is defined as the power
series:

( ) ( )
k
k
X z x n z

=
=


( ) [ ( )] X z Z x n =

where z is a complex variable. The above given relations are sometimes called the
direct Z - transform because they transform the time-domain signal x(n) into its
complex-plane representation X(z). Since Z transform is an infinite power series, it
exists only for those values of z for which this series converges. The region of
convergence of X(z) is the set of all values of z for which X(z) attains a finite value.
For discrete-time systems, z-transforms play the same role of Laplace transforms do
in continuous-time systems
Bilateral forward Z transform


Bilateral inverse Z transform
| |

=
n
n
z n h z H ] [

+
=
R
n
dz z z H
j
n h
1
] [
2
1
] [
t


Z-transform Pairs
h[n] = d[n]
Region of convergence: entire z-plane
| | | | 1 ] [
0
0
= = =

=

n
n
n
n
z n z n z H o o
h[n] = d[n-1]
Region of convergence: entire z-plane
h[n-1] z
-1
H[z] | | | |
1
1
1
1 1 ] [

=

= = =

z z n z n z H
n
n
n
n
o o
Inverse z-transform
| | | | dz z z F
j
n f
n
j c
j c
1
2
1

+

=
t

Using the definition requires a contour integration in the complex z-plane.
Fortunately, we tend to be interested in only a few basic signals (pulse, step, etc.)
Virtually all of the signals well see can be built up from these basic signals.
For these common signals, the z-transform pairs have been tabulated (see
Lathi, Table 5.1)


Z-transform Properties
Properties of z - transform
1. Linearity
)) ( ( )) ( ( )) ( ) ( (
2 1 2 1
nT x Z nT x Z nT x nT x Z + = +
2. Initial Value
) ( lim ) 0 ( z X x
z
=

+ + =
1
) 1 ( ) 0 ( ) ( z x x z X
3. Final value
) ( ) 1 ( lim ) (
1
1
z X z x
z

=

) ( lim ) (
0
s sX x
s
=

) ( ) 1 ( lim ) ( lim
) ( ) 1 ( ) (
1
1
1 1
1 0
1
1 0
1
1
1
z X z s sX
z X z s sX
z s
z s
z s
z s

UNIT V

LINEAR TIME INVARIANT DISCRETE TIME SYSTEMS
DIFFERENCE EQUATIONS,BLOCK DIAGRAM
REPRESENTATION,IMPULSE RESPONSE,CONVOLUTION SUM,LTI
SYSTEMS ANALYSIS USING DTFT AND Z-TRANSFORMS,STATE
VARIALE AND MATRIX REPRESENTATION OF SYSTEMS
DIFFERENCE EQUATIONS

A discrete-time system is anything that takes a discrete-time signal as input and
generates a discrete-time signal as output.1 The concept of a system is very general. It
may be used to
model the response of an audio equalizer .
In electrical engineering, continuous-time signals are usually processed by electrical
circuits described by differential equations.
For example, any circuit of resistors, capacitors and inductors can be analyzed using
mesh analysis to yield a system of differential equations.
The voltages and currents in the circuit may then be computed by solving the equations.
The processing of discrete-time signals is performed by discrete-time systems.
Similar to the continuous-time case, we may represent a discrete-time system either by
a set of
difference equations or by a block diagram of its implementation.
For example, consider the following difference equation.
y(n) = y(n-1)+x(n)+x(n-1)+x(n-2)
This equation represents a discrete-time system. It operates on the input signal x(n)x(n)
to produce the output signal y(n).
BLOCK DIAGRAM REPRESENTATION
Block diagram representation of


| | | | | | | | n x b n y a n y a n y
0 2 1
2 1 + + =

LTI systems with rational system function can be represented as constant-
coefficient difference equation
The implementation of difference equations requires delayed values of the
input
output
intermediate results
The requirement of delayed elements implies need for storage
We also need means of
addition
multiplication
Direct Form I
General form of difference equation


| | | |

= =
=
M
k
k
N
k
k
k n x b k n y a
0 0

Alternative equivalent form

Direct Form II


Cascade form
General form for cascade implementation



| | | | | |

= =
=
M
k
k
N
k
k
k n x b k n y a n y
0 1
( )
( ) ( )( )
( ) ( )( )
I I
I I
=
-
=

=
-
=

=
2 1
2 1
1
1 1
1
1
1
1 1
1
1
1 1 1
1 1 1
N
k
k k
N
k
k
M
k
k k
M
k
k
z d z d z c
z g z g z f
A z H

Parallel form
Represent system function using partial fraction expansion


IMPULSE RESPONSE
Impulse response h[n] can fully characterize a LTI system, and we can have the
output of LTI system as

The z-transform of impulse response is called transfer or system function H(z).


CONVOLUTIO N SUM
The convolution sum provides a concise, mathematical way to express
the output of an LTI system based on an arbitrary discrete-time input signal and the system's
response. The convolution sum is expressed as




Linear time-invariant systems can be described by the convolution sum
Convolution is conmutative
x[n] - h[n] = h[n] - x[n]
Convolution is distributive
x[n] - (h
1
[n] + h
2
[n]) = x[n] - h
1
[n] + x[n] - h
2
[n]
Cascade connection:
y[n] = h
1
[n] - [ h
2
[n] - x[n] ] = [ h
1
[n] - h
2
[n] ] - x[n]
Parallel connection
y[n] = h
1
[n] - x[n] + h
2
[n] - x[n] ] = [ h
1
[n] + h
2
[n] ] - x[n]
LTI systems are stable iff

( )
( )
( )( )

= =
-

+ =
P P P
N
k
N
k k k
k k
k
k
N
k
k
k
z d z d
z e B
z c
A
z C z H
1 1
1 1
1
1
0
1 1
1
1
| | | | | | n h n x n y - =
( ) ( ) ( ). z H z X z Y =
[ ] [ ] [ ]
[ ] [ ] [ ]
k
y n x k h n k
y n x n h n

=
=
= -

[ ]
k
h k

=
<

LTI systems are causal if

h[n] = 0 n < 0
LTI SYSTEMS ANALYSIS USING DTFT
Consider and
, then
magnitude

phase

Frequency response at is valid if ROC includes



LTI SYSTEMS ANALYSIS USING Z-TRANSFORM
The z-transform of impulse response is called transfer or system function H(z).

General form of LCCDE




Compute the z-transform





System Function: Pole/zero Factorization
Stability requirement can be verified.
Choice of ROC determines causality.
Location of zeros and poles determines the frequency response and phase




) ( ) ( ) (
e e e j j j
e H e X e Y Z + Z = Z
) ( ) ( ) (
e e e j j j
e H e X e Y =
) (
) ( ) (
e
e e
j
e X j j j
e e X e X
Z
=
) (
) ( ) (
e
e e
j
e H j j j
e e H e H
Z
=
, 1 = z ( ) ( )
1 =
=
z
j
z H e H
e
( ) ( ) ( )
e e e j j j
e H e X e Y =
( ) ( ) ( ). z H z X z Y =
| | | | k n x b k n y a
M
k
k
N
k
k
=

= = 0 0
( ) z X z b z Y z a
k
M
k
k
N
k
k
k

= =

=
0 0
) (
( )
( )
( )

=
= =
N
k
k
k
k
M
k
k
z a
z b
z X
z Y
z H
0
0
( )
( )
( )
I
I
=

=
N
k
k
M
k
k
z d
z c
a
b
z H
1
1
1
1
0
0
1
1