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EC6303 Signals and Systems

Lecture Notes

ECE 3rd Semester

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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 xn 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.

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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

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. 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

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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

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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).

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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

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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.

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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 discretetime 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

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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.)

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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

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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,

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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). an and bn are called
Fourier coefficients and are given by

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

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Solution to the above example
Coefficient a0 is given by

Coefficients an is given by

And coefficients bn is given by

A computation of the above coefficients gives

a0 = 0 , an = 0 and bn = [ 2 / (n*pi) ] [ 1 - cos (n pi) ]
Note that cos (n pi) may be written as
cos (n pi) = (-1)n
and that bn = 0 whenever n is even.
The given function f(t) has the following Fourier series

Continuous Time Fourier Transform

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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

becomes a non-periodic signal

, the periodic signal

and the following will result:

Interval between two neighboring frequency components becomes zero:

Discrete frequency becomes continuous frequency:

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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.

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Comparing Fourier coefficient of a periodic signal

a non-periodic signal

we see that the dimension of

If

with with Fourier spectrum of

is different from that of

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:

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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:

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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

larger. Also note

and it gets taller and narrower as

When

approaches infinity,

Recall that the Fourier coefficient of

for all , and the spectrum becomes

is

gets

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which represents the energy contained in the signal at

frequency), and the spectrum
which is infinity at zero frequency.

(DC component at zero

is the energy density or distribution

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 0k by the weights Xk
and summing:

p1

x(n) X k e

x(t) X k e ik0t

ik0n

k 0

We can perform a similar reconstruction for aperiodic signals

1
1
x(t) X ()eit d
x(n) X ()ein d
2
2
These are called the inverse transforms.
X () x(t)eit dt (t)eit dt ei0 1
Fourier Transform of Impulse Functions
Find the Fourier transform of the Dirac delta function:

X () x(n)ein (n)ein ei0 1

Find the DTFT of the Kronecker delta function:

X () x(n)e

in

(n)ein ei0 1
n

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

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Application of Laplace Transform to Circuit Analysis

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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

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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:

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Block diagram representations

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

Impulse Response
(t)

LTI System
H[.]

h(n)=H[ (t)]

A very important way to analyse a system is to stud y 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

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Properties of System Impulse Response

Stable
A system is stable if the impulse response is absolutely summable

Causal
h(t)
t
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

x(t) x()(t )d

Convolution Integral- Properties

Commutative

Associative

x(t)* h(t) h(t)* x(t)

[x(t)*h1 (t)]*h2 (t) x(t)*[h1 (t)*h2 (t)]

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Distributive x(t)*[h1 (t) h2 (t)] [x(t)*h1 (t)] [x(t)*h2 (t)]

Thus, using commutative property:

x(t) x()h(t )d h()x(t )d

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):

x (t)
x 1(t)
x(t) 2 x (t) x2 (t)
x (t)t
1
N

Note that inxth(its) development, v(t) will be the input, y(t) will be the
N
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

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Consider the CT differential equations:

y(t) a1 y(t) a0 y(t) b0 v(t)

A second-order differential equation requires two state variables:

x1 (t) y(t)
x2 (t) y(t)
We can reformulate the differential equation as a set of three equations:

x1 (t) x2 (t)
x2 (t) a0 x1 (t) a1 x2 (t) b0 v(t)
y(t) x (t)
We can write these in1 matrix form as:
x1 (t) 0
x t a
2 ( ) 0

1 x1 (t) 0
v(t)
a x
1 2 (t ) b0
x (t)
1
th order
This can be extendeyd(tt)oan
N0equation of this type:
x1 (t)differential

2
y N (t) N 1 a yi (t) b v(t)

x t y(i1) t i
N
, 1, 2, ...,
i ( )

i0

The state variables are defined as

The resulting state equation is

x1 (t) x2 (t)
x2 (t) x3 (t)
xN 1 (t) xN (t)
N 1

xN (t) ai xi1 (t) b0 v(t)

i0

y(t) x1 (t)
Matrix representation

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1
0

0
0
A

0
0

a
a
0
1
C 1 0 0

0
1
0
a
2

0
0

0
B

1
0
a
b
N 1
0
D 0

UNIT IV
ANALYSIS OF DISCRETE TIME SIGNALS
SAMPLING OF CT SIGNALS AND ALIASING,DTFT AND PROPERTIES,ZTRANSFORM 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 fs is defined as the number of samples obtained
in one second, or fs = 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 fN can be observed in the sampled signal, but
their frequency is ambiguous. That is, a frequency component with frequency f cannot be

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distinguished from other components with frequencies NfN + f and NfN 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

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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 < 2m.

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 discretetime Fourier transform (or DTFT) of
is usually written:

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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

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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

Property

is the complex conjugate of the function x[n]

represents the correlation between x[n] and y[n].

Time domain

Frequency domain

Remark
s

Linearity
Shift in time
Shift in
frequency
(modulation
)
Time
reversal
Time
conjugation
Time
reversal &
conjugation
Derivative
in frequency

integer k
real
number a

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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:

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

H[z] hnz n
Bilateral inversn eZ transform
1
h[n]
H[z] zn1dz
2 j R

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

H[z] nz
n

nz n 1
n0

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

h[n-1] z H[z] H[z] n 1z

-1

n 1z n z1
n1

Inverse z-transform
cj

f n 1 F z z n1dz
2j c
j

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

Z (x1(nT) x2 (nT)) Z (x1(nT)) Z (x2 (nT))

2. Initial Value

3.

x(0) lim X (z)

z

X (z) x(0) x(1)z1

1
Final value x() lim(1 z ) X (z)
x() lim sX (s)
s 0

z1

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s 0

1
s
s
sX (s)

lim sX (s)

s 0

z 1
1
1 z 1
1 z 1
(1 z 1 ) X (z)
lim (1 z 1 ) X (z)
z 1

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
yna1yn 1a2 yn 2b0xn

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LTI systems with rational system function can be represented as constantcoefficient 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
N

a yn k b xn k
k 0

k 0

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Alternative equivalent form
N

k 1

k 0

yn ak yn k bk x n k

Direct Form II

Cascade form
General form for cascade implementation
z 1g z 1g z
H z A1f
M1

k 1
N1

M2

1
k

k 1
N2

1c z 1d z 1d z
1

k 1

k 1

1
k

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4
0

Parallel form
Represent system function using partial fraction expansion
NP
Bk 1ek z 1
Ak
HEzSPO

IMPULSE R
NSCEk z

1
ankdzwe
k a LTI
can have the
1ckcterize
z 1
1dskystem,
z 1 1d
Impulse response h[n]
can fullyk 1chara
k 0
1

NP

NP

output of LTI system as

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

y[n] x[k] h[n k]

k

Linear time-yi[nnv]ariax[nnt ]syhst[enm

] s can be described by the convolution sum
Convolution is conmutative

x[n] h[n] = h[n] x[n]

Convolution is distributive

x[n] (h1[n] + h2[n]) = x[n] h1[n] + x[n] h2[n]

Cascade connection:

y[n] = h1[n] [ h2[n] x[n] ] = [ h1[n] h2[n] ] x[n]

Parallel connection

y[n] = h1[n] x[n] + h2[n] x[n] ] = [ h1[n] + h2[n] ] x[n]

LTI systems are stable iff

h[k]

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LTI systems are causal if

h[n] = 0 n < 0
LTI SYSTEMS ANALYSIS USING DTFT

Consider
, then

X (e j) X (e j) e jX ( e j)

and H (e j) H (e j) e jH (e j)

magnitude

Y (e j) X (e j) H (e j)
phase

Y (e j) X (e j) H (e j)
Frequency response at

is valid if ROC includes

H e j
Hz

z 1

z 1,

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

Y zX zH z.
General form of LCCDE
N

a yn k b xn k
k 0
N

k 0
M

a z Y (z)

b zk X z

Compute thekz0-transform

k 0

Y z
H
z
kN0
X z

bk z

k k

k 0

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

1c z
M

b
H z 0
a0

k 1
N

1d z
k 1

k 1

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