F 1379073256653

Signals and Systems
Textbooks: Fundamentals of signals and systems using the web and Matlab
Authors: Edward W. Kamen and Bonnie S. Heck
Prentice-Hall International, Inc.
Web site: http://users.ece.gatech.edu/~bonnie/book/
Domestic dealer: Chwa books Corp.
Lecture note compiler: Ping-Sung Liao
Chapter 1 Fundamental Concepts
1.1 Signals and Systems

Signals:
x(t) is a real-valued, or scalar-valued, function of the time variable t.
For instance, f (t ) = sin(wt)
The representation of a signal may be described by either a continuous-time signal or a
set of sample values.
For easily understanding the composition of a signal, the signal may be in terms of the
frequency spectrum through Fourier transform.
Signal Processing
Signal processing plays an important role in either the extraction of the information
carried in a signal or the reconstruction of a signal which has been corrupted by spurious
signals of noise.
Reconstruct the x(t) from m(t) (estimation/filtering) by canceling the noise n(t)
m(t ) = x(t ) + n(t )
Systems
A system is an interconnection of components with terminals or access ports through
which energy and information can be applied or extracted.
A mathematical model of a system is usually an idealized representation of the system.
There are two types basic of mathematical models: one is input/output representation; the
other is state model.
Four types of input/output representations are studied here,
1. The input/output differential equation or difference equation,
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2. The convolution model,

3. The Fourier transform representation,
4. The transfer function representation.
1.2 Continuous-Time Signals

A signal x(t) is said to be a continuous-time signal or analog signal when time variable
t takes its values from the set of the real number.
Step function
The unit-step function of u(t) is described mathematically by
1, t 0
u (t ) =
0, t < 0
The magnitude of unit-step function u(t) is equal to 1 for all t 0 .
Ramp function
The unit-ramp function of r(t) is described mathematically by
t , t 0
r (t ) =
0, t < 0
Note that for t 0 , the slope if r(t) is 1.
Unit-impulse function (t)
The unit impulse function, also called the delta function or the Dirac function, is defined
in generalized form by
(t ) = 0, t 0
(t ) = 1,
for any real number >0
Periodic signals
A continuous-time signal x(t) is periodic with period T if x(t + T) = x(t ) , for all t.
Note that the fundamental period is the smallest positive number T which satisfies the
foregoing definition of periodic signals.
Time-Shift signals
Given a continuous-time signal, x(t), the shifted version of x(t) usually is denoted as x(t-t1)
or x(t+t1) where t1 > 0. The signal of x(t-t1) is shifted to the right by t1 seconds and the
signal of x(t+t1) is shifted to the left by t1 seconds.
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Continuous and Piecewise-Continuous Signals

x(t) is discontinuous at t1 if x (t1 ) x (t1+ ) .
x(t) is continuous at t1 if x (t1 ) = x (t1+ ) .
x(t) is continuous signal if it is continuous at all points t.
Remark:
continuous
continuous
>>
>>
continuous-time signal
continuous-time signal is continuous as a function of t.
A continuous signal x(t) is said to be piecewise continuous if it is continuous at all points

t except at a finite or countably infinite collection of points t ,i = 1, 2, 3, .....
Derivative of a continuous-time signal
A continuous-time signal x(t) is said to be differentiable at point t1 if its ordinary
derivative
x(t + h) x(t1 )
dx(t )
= lim 1
dt t =t1 h 0
h
has a limit as h->0.

Generalized derivative of x(t) (if x(t) discontinuous at t1 )
Piecewise continuous signal x(t) may have a derivative in the generalized sense such as
dx(t )
+ [ x(t1+ ) x(t1 )1 ] (t t1 )
dt t t1
1.3 Discrete-Time Signals
A discrete-time signal is a signal that is a function of the discrete-time variable tn; in other
words, a discrete-time signal has values only at the discrete-time points t= tn, where n is
integer number.
The waveform of x[n] is usually depicted by stem plot.
Sampling
One of the most common ways in which discrete-time signals arise is in sampling
continuous-time signals.
x[n] = x(t ) t = nT = x(nT ) where T is the sampling period.

Note that nonuniform sampling is sometimes utilized in practical applications but is not
considered here.
Discrete-time unit step unction
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1, n =0, 1, ...
u[n] =
0, n =-1, -2, ...
Discrete-time Ramp function
n, n =0, 1, ...
r[ n ] =
0, n=-1, -2, ...
Discrete-time unit-pulse function [n]
1, n =0
[ n] =
0, n 0
Discrete-Time Periodic Signals
x[n + r] = x[n], for all integer n and r is called the period (fundamental period).
For instance, x[n] = Acos(n + )
The signal is periodic if Acos[(n+r) + ]= Acos(n + ).
In other words, r=2q for some integer q.
1.4 Examples of Systems

Four examples of continuous-time systems
RC circuit
dy (t ) 1
+ y (t ) = i(t ) = x(t )
dt
R
Assumption : initial value y (t0 ), x(t ) = u (t )
1 (t / RC )(t )
e
d
0 C
= y (t0 ) + R[1 e (t / RC )t ], t 0
y (t ) =
Car on a level surface
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d 2 y (t )
M
+ k f y (t ) = x(t )
dt 2
Let v(t ) = dy (t ) / dt
Thus,
dv(t )
M
+ k f v(t ) = x(t )
dt
Mass-Spring-Damper system
d 2 y (t )
+ Dy (t ) + Ky (t ) = x(t )
dt 2
Simple pendulum
d 2 (t )
+ MgL sin (t ) = Lx(t )
dt 2
If the magnitude of the angle (t ) is small, so that sin (t ) is approximately equal to
I
(t ) and the foregoing equation can be approximated by

I
d 2 (t )
+ MgL (t ) = Lx(t )
dt 2
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Example of discrete-time system

Repayment of the a bank loan
The input x[n] is the amount of the loan payment in the nth month, and the output y[n] is
the balance of the loan after the nth month, and I is the yearly interest rate.
I
y[ n] 1 + y[ n 1] = x[ n], n =0, 1, 2, ...

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1.5 Basic System Properties

Causality
A system is said to be causal if for any time t1, the output response y(t1) resulting from
the input signal x(t) is only dependent on values of the input x(t) for t > t1.
Memoryless system and memory system
A causal system is memoryless or static system if for any time t1, the value of the output
at time t1 is depends only the value of the input at time t1.
A causal system that is not memoryless is said to have memory.
Linearity
A system is said linear if it is both additive and homogeneous.
x1 (t ) 6 y1 (t )
x2 (t ) 6 y2 (t )
additive: x1 (t ) + x2 (t ) 6 y1 (t ) + y2 (t )
homogeneous: ax1 (t ) 6 ay1 (t ), where a is a scalar
linearity: a1 x1 (t ) + a2 x2 (t ) a a1 y1 (t ) + a2 y2 (t ), where a1 and a2 are any real scalar
A system that is not linear is said to be nonlinear.
Note that the system analysis on more widely used systems are based on linearization
method.
Time-invariance
The system is said to be time-invariant if for any input x(t) and any time variable t1, the
response to the shifted input x(t-t1) is equal to y(t-t1), where y(t) is the response to x(t)
with zero initial energy.
A system is time varying or time variant if it is not time invariant.
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Finite dimensionality
Given a continuous system with input x(t) and output y(t), the system is finite dimensional
if for some positive integer N and nonnegative M, can be written in the form
y ( N ) (t ) = f ( y (t ), y (1) (t ),..., y ( N 1) (t ), x(t ), x (1) (t ),...x ( M ) (t ), t ),
where N 1, M > 0
Given a discrete-time system with input x[n] and output y[n], the system is finite
dimensional if for some positive integer N and nonnegative M, can be written in the form
y[n] = f ( y[n 1], y[n 2],..., y[ n N ], x[ n], x[ n 1],..., x[ n M ], n),
where N 1, M > 0
A continuous-time system with memory is infinite dimensional if it is not finite

dimensional.
The examples of infinite dimensional system are often proved by the law of
contradiction.
For instance, system with time delay as given below,

Infinite dimensional system-1
dy (t )
+ ay (t 1) = x(t )
dt
In practice, it is seldom possible to express the solution of a delay-differential equation in
analytic form.
Infinite dimensional system-2
Consider the discrete-time system with the given input/output relationship,
n 1
y[n] =
i =0
1
x[i ]
ni
1
1
1
x[0] + (
) x[1] + ... + x[n 1] + 0 x[ n]
n
n 1
1
1
1
1
= 0 x[n] + x[n 1] + ... + (
) x[n (n 1)] + x[n n]
n 1
n
1
Because the last term in the right-side cannot be specified by x[n-M] in which M is a
deterministic non-negative integer, this discrete-time difference equation is infinite
dimensional.
y[n] =
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Linear finite-dimensional system

N -1
= ai (t ) y (i ) (t ) +
i =0
b (t ) y
i =0
(i )
(t )
where N 1, M > 0
y[n] = f ( y[n 1], y[n 2],..., y[n N ], x[ n], x[n 1],..., x[ n M ], n),
N 1
i =1
i =0
= ai (n) y[n i ] + bi (n) x[n i ]

where N 1, M > 0
Linear time-invariant finite-dimensional system

N -1
= ai (t ) y (i ) (t ) +
i =0
b (t ) x
i =0
(i )
(t )
i, ai (t ) and bi (t ) are constants, and N 1, M > 0

y[n] = f ( y[n 1], y[n 2],..., y[n N ], x[n], x[ n 1],..., x[ n M ], n),
N 1
i =1
i =0
= ai (n) y[n i ] + bi (n) x[n i ]

i, ai (n) and bi (n) are constants, and N 1, M > 0
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Signals and Systems

Chapter 2 System defined by Differential or difference equations

2.1 Linear Input/Output Differential Equations with Constant Coefficients
N -1
y ( N ) (t ) + ai (t ) y (i ) (t ) =
i =0
b (t ) x
i =0
(i )
(t )
Here it is assumed that i, ai (t ) and bi (t ) are real constants,

and N 1, M > 0
In other words,
N -1
y ( N ) (t ) + ai y (i ) (t ) =
i =0
b x
i =0
(i )
(t )
Here it is assumed that i, ai and bi are real constants,

and N 1, M > 0
Initial condition
To solve the foregoing equation, it is necessary to specify the N initial conditions, either
y (0), y (1) (0),... y ( N 1) (0) for most cases or y (0 ), y (1) (0 ),... y ( N 1) (0 ) if the Mth
derivative of the input x(t) contains an impulse k(t) or a derivative of an impulse.
First-order case
Canonical formation
dy (t )
(2.4)
+ ay (t ) = bx(t )
dt
Its output response y(t) for the initial condition y(0) and input x(t) is given by
t
y (t ) = y (0)e ( at ) + e a (t )bx( )d , t 0
0
(2.5)
or
t
y (t ) = y (0 )e ( at ) + e a ( t )bx( )d , t 0
0
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READING SKILL
dy (t )
dx(t )
+ ay (t ) = b1
+ b0 x(t )
dt
dt
(2.7)
d [ y (t ) b1 x(t )]
+ ay (t ) = b0 x(t ),
dt
d [ y (t ) b1 x(t )]
+ a[ y (t ) b1 x(t )] = b0 x(t ) ab1 x(t ),
dt
dq (t )
+ aq (t ) = (b0 ab1 ) x(t )
dt
where q (t ) = y (t ) b1 x(t )
(2.11)
This gives that

t
q(t ) = q(0 )e ( at ) + e a (t ) (b0 ab1 ) x( )d , t 0

0
(2.12)
Thus, y(t) can be yields by substituting y (t ) = q(t ) + b1 x(t ) into (2.12)

t
y (t ) = b1 x(t ) + [ y (0) b1 x(0)]e ( at ) + e a (t ) (b0 ab1 ) x( )d , t 0

0
(2.13)
Note that if the input x(t) is the unit-step function u(t), the response of output y(t) in
(2.13) is expressed as
t
y (t ) = b1 x(t ) + [ y (0 ) b1 x(0 )]e ( at ) + e a (t ) (b0 ab1 ) x( ) d , t 0

0
= b1u (t ) +y (0 )e ( at ) + + e a ( t ) (b0 ab1 )d , t 0

0
= b1u (t ) +y (0 )e
=
( at )
+ (b0 ab1 )e at (1/ a)[1 e a ( t ) ]d , t 0
2.2 System Modeling
In building the input/output equation of a system, one must be familiar with the laws of
physics and the methodologies of mathematics.
Electrical circuits
Resistor: v (t ) = Ri (t )
dv(t )
1 t
Capacitor : i (t ) = C
or v(t ) =
i ( ) d
dt
C -
di (t )
1 t
or i (t ) = v( ) d
Inductor : v(t ) = L
dt
L -
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where v(t)and i(t) are the terminal voltage across the component such as R, L, C.
Mechanical systems
There are three types of forces that resist the translation motion.
Inertia force x I (t ): xI (t ) = M
d 2 y (t )
dt 2
Damping force x d (t ): xd (t ) = kd
d y (t )
dt
Spring force x s (t ): xd (t ) = k s y (t )
Dalemberts principle
Any fixed time the sum of all external forces applied to a body in a given direction and
all the forces resisting the motion in that direction must be equal to zero.
Rotational mechanical systems

In analogy with the three types of forces resisting translation motion, there are three types
of forces resisting rotational motion.
Inertia torque x I (t ): xI (t ) = I
d 2 (t )
dt 2
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Damping torque x d (t ): xd (t ) = kd
d (t )
dt
Spring torque x s (t ): xd (t ) = k s (t )
2.3 Linear Input/Output Difference Equations with Constant Coefficients

N 1
i =1
i =0
y[n] + ai y[n i ] = bi (n) x[n i ]
(2.35)
i, ai and bi are real constants, and N 1, M > 0
The foregoing difference equation is linear, time-invariant, causal and finite dimensional.
The solution of (2.35) can be computed recursively as follows.
N 1
i =1
i =0
y[n] = ai y[n i ] + bi x[n i ] , n = 0,1, 2,....
For an infinite dimensional difference equation its solution cannot be computed

recursively as above.
Example:
By considering the first-order linear difference equation
y[ n] = ay[ n 1] + bx[ n]
with initial condition y[0].
First, setting n=1, n=2, n=3 in (2.39) gives

y[1] = ay[0] + bx[1]
y[2] = ay[1] + bx[2]
= a(ay[0] + bx[1]) + bx[2]
= a 2 y[0] abx[1] + bx[2]
y[3] = ay[2] + bx[3]
= a(a 2 y[0] abx[1] + bx[2]) + bx[3]
= a 3 y[0] + a 2bx[1] abx[2] + bx[3]
Finally, it can be seen that n 1
n
y[ n] = ( a ) n y[0] + ( a ) n i bx[ n]
i =1
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2.4 Discretization in Time of Differential Equations

The discretization in time actually results in a discrete-time representation of the
continuous-time system governed by the given input/output differential equation.
First-order case
dy (t )
+ ay (t ) = bx(t )
dt
(2.46)
The derivative of y(t) can be approximated by
dy (t )
y[(n + 1)T ] y[nT ]
=
dt t =nT
T
(2.48)
Inserting the approximation (2.46) into (2.45) gives

y[( n + 1)T ] y[nT ]
= ay[ nT ] + bx[nT ]
T
(2.49)
Let x[n] = x(t ) t =nT and y[n] = y (t ) t =nT

In terms of this notation, (2.49) becomes
y[( n + 1)] y[ n]
= ay[ n] + bx[ n]
T
Be careful, the sampling period T in the denominator cannot be omitted here.
Finally, the discrete approximation to (2.46) can be expressed as
y[ n] y[ n 1] = aTy[ n 1] + bTx[n 1]
or
y[ n] = (1 aT ) y[n 1] + bTx[n 1]
(2.51)
Second-order case
Consider a linear time-invariant continuous-time system with the second-order

input/output differential equation
d 2 y (t )
dy (t )
dx(t )
+ a1
+ a0 y (t ) = b1
+ b0 x(t )
2
dt
dt
dt
(2.62)
Dealing with the above equation, the following approximations are needed to
approximate the first-order and the second-order derivatives.
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dy (t )
y[(n + 1)T ] y[nT ]
=
dt t = nT
T
d 2 y (t )
y[(n + 2)T ] 2 y[(n + 1)T ] + y[nT ]
=
2
dt t = nT
T2
Setting t=nT in (2.62) and using the approximations given above result in the following
time discretization of (2.62).
y[ n + 2] 2 y[n + 1] + y[ n]
y[n + 1] y[ n]
x[n + 1] x[ n]
+ b0 x[n]
+ a1
+ a0 y[ n] = b1
2
T
T
T
After summarizing,
y[n] + (a1T 2) y[n 1] + (1 a1T + a0T 2 ) y[n 2]

= b1Tx[n 1] + (b0T 2 b1T ) x[n 2]
2.5 Systems Defined by Time-Varying or Nonlinear Equations

First-order time-varying system
dy (t )
+ a (t ) y (t ) = b(t ) x(t )
dt
The time-varying property of capacitance C(t) is a result of changing the position of the
dielectric. The charge-voltage relationship of capacitor is given by
q (t ) = C (t )vc (t )
Thus, taking the derivative of both sides of the previous equation
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ic (i ) =
v (t )
dq (t )
= C (t ) c + C (t )vc (t )
dt
dt
The solution of a linear input/output differential equation with time-varying coefficients

cannot be obtained exactly by analytic form in general, but can be obtained by
numerical-solution techniques.
In general, due to the nonlinearity if is not possible to derive an analytic expression for
the output response y(t) of a nonlinear system with an initial condition y(0) and the input
x(t). Similarly, the response y(t) of a nonlinear system is often computed using a
numerical solution techniques as the linear input/output differential equation with
time-varying coefficients does.
The system behavior around the nominal conditions ynorm(0) for a nonlinear system can
be linerized into the linearized equation with respect to the nominal functions ynorm(t) and
xnorm(t).
Time-varying discrete-time system
y[n] = f ( y[n 1], y[n 2],..., y[ n N ], x[ n], x[ n 1],..., x[ n M ], n),

N 1
i =1
i =0
= ai (n) y[n i ] + bi (n) x[n i ]

i, ai (n) and bi (n) are constants, and N 1, M > 0
For instance,
y[n] + a( n) y[n 1] = b0 (n) x[n] + b1 (n) x[n 1]
Load-payment process
I ( n)
y[n] 1 +
y[ n 1] = x[ n]
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Nonlinear input/output difference equations

For instance,
y[ n] = f ( y[ n 1], x[ n],[ n 1])
If the explicit expression of function f(.) is given, the solution can be obtained as the
recursion process (program) for the linear system does.
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Signals and Systems

Chapter 3 Convolution Representation
3.1 Convolution Representation of Linear Time-Invariant Discrete-Time Systems
Throughout this section, there is no initial energy in the discussed linear time-invariant
discrete system which is also causal, but not necessarily finite dimensional.
Let h[n] is the output response when a unit pulse (impulse) [n] is applied to the system
with no initial energy at time n=0.
If the input signal is shift to right with delay time t=i, the shifted unit-pulse response with
respect to the same system shown in Fig. 3.1 is given below.
In general case, the input signal x[n] can be expressed in the form
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x[n] = x[0] [n] + x[1] [n 1] + x[2] [n 2] + ...
(3.3)
x[n] = x[i ] [n i ]
i =0
The response to x[i ] [n i] is given by

x[n] = x[0] [n] + x[1] [n 1] + x[2] [n 2] + ...
yi [n] = x[i ] [n i ]h[n i ]
(3.4)
= x[i ]h[n i ]
By additive property, the response to the sum given by (3.3) must be equal to the sum of
the individual sum yi[n]. Thus the response to x[n] is
y[n] = yi [n]
i =0
= x[i ]h[n i ], n 0
(3.5)
i =0
3.2 Convolution of Discrete-Time Signals
In this section, the convolution operation is defined for arbitrary discrete-time signals x[n]
and v[n] that are not necessarily zero for n<0.
The convolution of x[n] and v[n] is defined by
x[n]* v[n] =
x[i]v[n i]
i =
(3.7)
v[i]x[n i]
i =
If x[n] is zero for n<0, this convolution operation is given by

0,
x[n]* v[n] =
x[i ]v[n i ],
i =0
n<0
n0
Similarly, if v[n] is zero for n<0, the convolution operation is given by
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0,
x[n]* v[n] =
x[i ]v[n i ],
i =0
n<0
n0
The graphical illustration of the convolution operation is given below.
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Compute the convolution by tabulation for special cases.

For instance, suppose that x[n]=u(n-N) for all n<N and v[n] = u(n-M) for all n< M, where
N and M are positive or non-negative integers.
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The convolution of x[n] and v[n] written in the form

n < (N + M )
0,
y[n] = x[n]* v[n] = N M

x[i ]v[n i ],
i=N
n (N + M )
can be computed by the multiplication of two rows of x[n] and v[n] using array structure.
x[ N ]
v[ M ]
x[ N + 1]
x[ M + 1]
x[ N + 2]
x[ M + 2]
x[ N + 3]
x[ M + 3]
...
...
----------------------------------------------------------------------------------------------------x[ N ]v[ M ] x[ N + 1]v[ M ] x[ N + 2] v[ M ]

x[ N + 3] v[ M ]
...
x[ N ]v[ M + 1] x[ N + 1]v[ M + 1] x[ N + 2] v[ M + 1] x[ N + 3] v[ M + 1]
x[ N ]v[ M + 2]
x[ N + 1]v[ M + 2] x[ N + 2] v[ M + 2]
...
----------------------------------------------------------------------------------------------------y[ M + N ] y[ M + N + 1] y[ M + N + 2]
y[ M + N + 3]
...
...
Properties of the convolution operation

Associativity: x[n]*(v[n]* w[n]) = ( x[n]* v[n]) * w[n]
Commutativity: x[n]* v[n] = v[n]* x[n]
Distributive with addition: x[n]*(v[n] + w[n]) = ( x[n]* v[n]) + ( x[n]* w[n])
Shift property:
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w[n q] = xq [n]* v[n] = x[n]* vq [n]

where xq [n] and vq [n] are q - step right shifts (q - step left shifts)
of x[n] and v[n] when q > 0 (q < 0).
Convolution with unit pulse: x[n]* [n] = x[n]

Convolution with the shifted unit pulse: x[n]* q [n] = x[n q]
Example: given a causal linear time-invariant discrete-time system with unit-pulse
response h[n] and its initial energy being zero, when the input x[n] with x[n]=0 for n <0 is
applied to this system, as shown in (3.5), the output response y[n] will be
y[n] = x[i ]h[n i ], n 0
(3.23)
i =0
Since h[n]=0 for n <0 (by causality), h[n-i]=0 for n <i.

n
y[n] = x[i ]h[n i ], n 0

i =0
Note that the system is causal and the input signal is zero before n=0.
Noncausal systems:
y[n] = x[n]* h[n] =
x[i]h[n i]
i =
Since it is bi-infinite sum, the result of convolution sum cannot be evaluated in a finite
number of computations.
3.3 Convolution Representation of Linear Time-Invariant Continuous-Time Systems
For a causal linear time-invariant continuous-time system, the impulse response h(t)
could be determined experimentally by applying a large-amplitude short-duration pulse
(as an approximation to (t)), but in practice it is usually not possible to apply such an
input to the system.
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The convolution of a system h(t) with an input x(t) is denoted by
y (t ) = x(t ) * h(t ) = x( )h(t )d
Consider a causal linear time-invariant continuous-time system h(t)

with input x(t)=0 for t<0,
y (t ) = x(t ) * h(t ) = x( )h(t ) d
= x( )h(t )d + x( )h(t )d
= x( )h(t )d
0
0,
= t
0 x( )h(t )d ,
(by causality )
t<0
t0
3.4 Convolution of Continuous-Time Signals
Given two continuous-time signals x(t) and v(t), the convolution of x(t) and v(t) is defined
by
x(t ) * v(t ) = x( )v(t )d
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If x(t)=0 and v(t)=0 for t<0, the foregoing equation becomes
x(t ) * v(t ) = x( )v(t )d
= x( )v(t )d + x( )v(t )d
= x( )v(t )d
0
0,
= t
0 x( )v(t )d ,
(by causality )
(3.34)
t<0
t0
The integral in (3.34) exists for all t>0 if the functions of x(t) and v(t) are integrable for
all t>0; that is,
x( ) d < and
v( ) d < for all t > 0
Graphical illustration for the convolution of two continuous-time signals
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Properties of Convolution for continuous-time signals

Properties of the convolution operation
Associativity: x(t ) *(v(t ) * w(t )) = ( x(t ) * v(t )) * w(t )
Commutativity: x(t ) * v(t ) = v(t ) * x(t )
Distributive with addition: x(t ) *(v(t ) + w(t )) = ( x(t ) * v(t )) + ( x(t ) * w(t ))
Shift property:
w(t c) = xc (t ) * v(t ) = x(t ) * vc (t )
where xc (t ) and vc (t ) are c - sec ond right shifts (c - sec ond left shifts)
of x(t ) and v(t ) when c > 0 (c < 0).
Convolution with unit pulse: x(t ) * (t ) = x(t )

Convolution with the shifted unit pulse: x(t ) * c (t ) = x(t c)
Derivative property:
If the signal x(t) has an ordinary first derivative,
d
[ x(t ) * v(t )] = x (t ) * v(t )
dt
d
[ x(t ) * v(t )] = x(t ) * v(t )
dt
If the signals x(t) and v(t) have an ordinary first derivatives,

d2
[ x(t ) * v(t )] = x (t ) * v(t )
dt 2
Integration property
Let x ( 1) (t ) and v ( 1) (t ) denote the integrals of the signals x(t) and v(t) ; that is
t
x ( 1) (t ) = x( )d and v ( 1) (t ) = v( )d
The convolution of x ( 1) (t ) and v ( 1) (t ) is represented by

( x(t ) * v(t ))( 1) = ( x ( 1) (t ) * v(t )) = ( x(t ) * v ( 1) (t ))
Proof:
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( x(t ) * v(t ))( 1)

=
=
=
=-
[x( ) * v( )]d
[
=-
(A-1)
x( )v( )d ] d
{ x( ) [
v( )d ] }d
t
Because of v ( 1) (t ) = v( )d
( x(t ) * v
=
=-
=-
=-
( 1)
(t ))
{ x( )v ( 1) (t )}d
{ x( ) [
{ x( ) [
=
=
(A-2)
v( )d ] }d
v( )d ] }d
Comparing the (A-1) with (A-2) yields
( x(t ) * v(t ))( 1) = ( x(t ) * v ( 1) (t ))

The detail of proving ( x(t ) * v(t ))( 1) = ( x ( 1) (t ) * v(t )) is left to the reader.
Example: given a causal linear time-invariant continue-time system with unit-impulse

response h(t) and its initial energy being zero, when the input with x(t) =0 for t <0 is
applied to this system, the output response y[n] will be
y (t ) = x(t ) * h(t ) = x( )h(t )d , t 0

0
(3.46)
By the commutativity of convolution,
y (t ) = x(t ) * h(t ) = h( ) x(t )d , t 0

0
(3.47)
Note that the system is causal and the input signal is zero before t=0.
Let g(t) is the output response of the system when the input x(t) is the unit-step function
u(t) with no initial energy in the system at time t=0. From the definition of convolution,
g (t ) = h(t ) * u (t )
Differentiating the both sides of the foregoing equation and using the derivative property
of convolution gives
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g (t ) = h(t ) * u (t )
= h(t ) * (t )
= h(t )
Hence the impulse response h(t) of a linear system is equal to the derivate of the step
response g(t) of the system. Also, the step response of the system can be said to be the
integral of the impulse response of the system.
Noncausal systems:
y (t ) = x(t ) * h(t ) = x( )h(t )d
3.5 Numerical Convolution
Consider a causal linear time-invariant continuous-time system given by the convolution

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

0
For time instant at nT, that is, t=nT
y (nT ) = x( )h(nT )d
0
The integral of the above equation can be obtained by
y (nT ) =
i =0
( i +1)T
=iT

i =0
( i +1)T
=iT
x( )h(nT )d
x(iT )h(nT iT )d
x(iT )h(nT iT )
i =0
( i +1)T
=iT
(1)d
Tx(iT )h(nT iT )
i =0
Similarly,
y (nT ) T h(iT ) x((n i )T )

i =0
In general, the approximation is more accurate the smaller the duration T is.
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3.6 Linear Time-Varying Systems

For discrete-time time-varying system
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It is IMPOSSIBLE to express the output response of a linear time-varying system as

the convolution of the input signal with the impulse response of this linear time-varying
system.
Given a causal time varying system, let h[n,i] denote the output response when the unit
pulse is applied to the system.
By causality,
h[n, i ] = 0, n < i
For an input signal x[n]=0 for n<0, where n is integer, the output response y[n] resulting
from x[n] with no initial energy at time n=0 is given by
y[n] =
x[i] h[n, i]
i =
(3.69)
y[n] = x[i ] h[n, i ], n 0

i =0
Note that if the given causal time-invariant discrete-time system can be possibly and
reasonably reduce to a causal time-invariant discrete-time system, the foregoing equation
can be rewritten by
n
y[n] = x[i ] h[n i ], n 0

i =0
because of h[ n, i ] = h[n i ], for all n i .
For continuous-time time-varying system
Given a causal time varying system, let h(t,) denote the output response when the unit
impulse is applied to the system.
By causality,
h(t , ) = 0, t <
For an input signal x(t)=0 for t<0, the output response y(t) resulting from x(t) with no
initial energy at time t=0 is given by
t
y (t ) = x( )h(t , )d , t 0
(3.73)
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Note that if the given causal time-invariant continuous-time system can be possibly and
reasonably reduce to a causal time-invariant continuous-time system, the foregoing
equation can be rewritten by
t
y (t ) = x( )h(t )d , t 0
0
because of h(t , ) = h(t ), for all t .
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Signals and Systems

CHAPTER 4 THE FOURIER SERIES AND FOURIER TRANSFORM

As will be seen, the frequency spectrum is used to fetch out the frequency counterparts of
a real signal. In general, the frequency spectrum is a complex-valued function of the
frequency variable, and thus it is usually specified in terms of an amplitude spectrum and
a phase spectrum.
Fourier series is a powerful mathematical tool to analyze the spectrum of a periodic
signal. Oppositely, Fourier transform is only suitable to analyze the spectrum of an
aperiodic ( a nonperiodic) signal.
4.1 Representation of Signals in Terms of Frequency Components

READ SKILL: What is introduced here does not induce the original idea about the
decomposition of a signal by Fourier series.
For a large class of signals, the original time waveform can be synthesized (decomposed)
by (into) the composition of sinusoidal signals.
N
x(t ) = Ak cos(k t + k ),
- < t <
(4.1)
k =1
The characteristics or features of a signal given in (4.1) can be studied in terms of the
frequencies, the amplitudes, and the phases of the sinusoidal terms comprising the signal.
In particular, the amplitudes Ak, k=1,..,N, are the major factors in determining the shape
of the signal.
NOTE THAT the direct component of a signal is not included in this Section.
Examples Sum of Sinusoids
x(t ) = A1 cos(t ) + A2 cos(4t + / 3) + A3 cos(8t + / 2), - < t <
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Eulers formula
e j (k +k ) = cos( jk + k ) + j sin( jk + k )
cos( jk + k ) = Re[e j (k +k ) ]
N
Hence, x(t ) = Ak cos(k t + k ),

k =1
N
x(t ) = Re[Ak e j (k t +k ) ], - < t <

k =1
In addition, s = a + jb, s = a jb yield a = s + s . Thus

A
A
Re[ Ak e j (k t +k ) ] = k e j (k t +k ) + k e j (k t +k )
2
2
Ak jk
Defining ck =
e ,
2
yields
k =1
k =1
x(t ) = ck e jk t + c k e jk t
Then x(t ) =
ce
j
k = N
kt
, <t <
where ck is in the complex form as k n , for some integer n , and the amplitude of
|ck| and |c-k| are equal.
DONT BE SILLY TO THINK A REAL-DOMAIN SIGNAL IN THE E
4.2 Fourier Series Representation of Periodic Signals
Let x(t) is a periodic signal with period T, that is
x(t ) = x (t + T ), for all t , < t < .
Then the signal can be expressed as a sum of complex exponentials

x(t ) =
ce
k =
jk0t
< t <
(4.11)
where c0 is a real number and ck for k0 are in general complex numbers, and 0 is the
fundamental frequency by 0=2/T.
Obviously, the frequency content of a periodic signal is line spectra.
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NOTE THAT the direct component is included because of k=0 being considered. In
particular, (4.11) is not equal to (4.1).
The coefficients of ck for k0 are computed using the formula
T
ck = x(t ) e jk0t dt = 2T x(t ) e jk0t dt ,

0
k= , -2, -1, 0, 1, 2,
(4.12)
and the DC term c0 of is given by

T
ck = x(t )dt ,
(4.13)
The proof of (4.12) and (4.13) is left to reader.

It is difficult to believe that a pulse train signal with corners can be represented as the
sum of sinusoids that are infinitely smooth functions.
It should be noted that the finite sum xN(t) CAN BE CALCULATED BY TRUCATING
THE EXPONENTIAL FOURIER SERIES DIRECTLY:
x(t ) =
ce
k = N
jk0t
< t <
A periodic signal has a Fourier series if it satisfies the Dirichlet conditions given by
1. x(t) is absolutely integral over any period; that is
a +T
x(t ) dt < , for any a
2. x(t) has only a finite number of maxima and minimum over any period.
3. x(t) has only a finite number of discontinuities over any period.
IMPORTANT SENSE
Fourier series representation of a periodic signal can be in three different forms: (1)
trigonometric form, (3) compact trigonometric form and (3) complex exponential form.
(1) x(t ) = a0 + (an cos k0t + bn sin k0t )
(LATHI, p. 595 trigonometric form)
k =1
(2) x(t ) = A0 + Ak cos( k0t + k ),
- < t <
k =1
(3) x(t ) =
ce
k =
jk0 t
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The trigonometric representation of the pulse train shown in Fig. 4.6 is
x(t ) =
1 2
+ ( cos(k t + [(1)( k 1) / 2 1] ),
2 k =1 k
2
< t <
(4.16)
k odd
To approach the plot of x(t), let xN (t ) denote the finite sum

xN (t ) =
1 N 2
+ ( cos(k t + [(1)( k 1) / 2 1] ),
2 k =1 k
2
< t <
k odd
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Gibbs phenomenon
It can be seen the magnitude of the
overshoot is approximately equal to
9% at the corners of pulse train.
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Parsevals theorm
Let x(t) is a periodic signal with period T, the average power of the signals is
P = x 2 (t ) dt =
0
k =
ck
4.3 Fourier Transform

How does the aperiodic signal be represented in terms of frequency content? Is it also
true to deal with an aperiodic signal using Fourier series? If not, what do we do to get the
spectrum content of an aperiodic signal? Fourier transform, in contrast to Fourier series,
is used to the above questions.
Again let us discuss the pulse train of x(t) shown in Fig. 4.6, as the period T is infinite, it
will be one-second rectangular pulse.
It is known that x(t) can be expressed in the form,

x(t ) =
ce
k =
jk0t
, <t<
where the coefficients ck is equal to

k 0
), k = 1, 2,...
k0T
2
and since 0 = 2 / T , the foregoing equation can be rewritten by
ck = (
) sin(
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ck = (
k
1
) sin( 0 ), k = 1, 2,...
2
k
The plots of |ck| for T=2, 5, 10 and infinite are display below.
It is obvious that as T is infinite, the spectrum of the one-shot pulse signal appears in
continuous form.
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The definition of Fourier transform is given by
X ( ) = x(t )e jt dt , -< <
(4.36)
and the inverse Fourier transform is expressed in the form

x(t ) =
1
2
X ( )e jt dt ,
(4.36A)
A signals x(t) is said to have a Fourier transform in the ordinal sense if the integral in
(4.36) converges (exists). In other words, the signal x(t) is well-behaved and the signal x(t)
is absolutely integrable,
x(t ) dt <
(4.37)
The term well-behaved means has a finite number of discontinuities, maxima, and
minima within any finite integral of time.
Except that impulse signal, most signals of interesting are well-behaved.
Example 4.5 Constant Signal

Consider the constant signal x(t)=1, - <t < . Please find the Fourier transform of
this constant signal.
Note that the condition of (4.37) is not obeyed for constant signal.
Example 4.6 Exponential signal
Now consider the exponential signal
x(t ) = e bt u (t ) .
Please find the Fourier transform of this constant signal.
( X ( ) =
1
1
, X ( ) =
, (X ( ) = tan 1 ( ) )
2
2
b + j
b
b +
Using Eulers formula

X ( ) = Re[ X ( )] + j Im[ X ( )]
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Defining R ( ) = Re[ X ( )]
I ( ) = Im[ X ( )]
Thus the rectangular form of Fourier transform for the signal x(t) is
X ( ) = R ( ) + jI ( )
(4.39)
R( ) = x(t ) cos tdt
I ( ) = x(t ) sin tdt
The polar form of a complex function X ( ) is
X ( ) = X ( ) exp( j(X ( ))
(4.40)
The relationship between (4.39) and (4.4) can be connected using the following formula.
X ( ) = R 2 ( ) + I 2 ( )
(X ( ) = tan 1 (
I ( )
)
R ( )
For even signals x(t)= x(-t)
R( ) = x(t ) cos tdt R( ) = 2 x(t ) cos tdt

I ( ) = x(t ) sin tdt = 0
For odd signals x(t)= -x(-t)
R( ) = x(t ) cos tdt = 0
I ( ) = x(t ) sin tdt I ( ) = 2 x(t ) sin tdt
Example 4.7 Rectangular pulse

1, - t <
p (t ) =
0, all other t
Please compute its Fourier transform. ( X ( ) =
sin(
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NOTE THAT there are two types of definition about sinc function
(http://en.wikipedia.org/wiki/Sinc_function).
1. In digital signal processing and information theory, the normalized sinc function
is commonly defined by
sinc( x ) =
sin( x)
x
2. In mathematics, the historical unnormalized sinc function (for sinus cardinalis),

is defined by
sinc( x) =
sin( x)
x
4.4 Properties of the Fourier Transform

It is well known x (t ) X ( ) and v(t ) V ( ) .
Linearity: ax (t ) + bv(t ) aX ( ) + bV ( ) , for any real or complex scalars a and b.
Left or right shift in Time: x(t c) X ( ) e jc for any real scalar c.
Time scaling:
1
X ( ) for any positive real scalar a.
a
a
1
x(at ) X ( ) for any nonzero real scalar a, either positive or negative.

a
a
x( at )
Time reversal: x( t ) X ( )
if x(t) is a real-valued signal, the Fourier transform can be rewritten as
x ( t ) X ( )
where is X ( ) the complex conjugate of X ( ) .
Multiplication by a power of t
dn
t x(t ) ( j )
X ( )
d n
n
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Multiplication by a complex exponential

x(t ) e j0t X ( 0 )
Multiplication by a complex exponential

1
x(t ) cos(0t ) [ X ( + 0 ) + X ( 0 )]
2
j
x(t ) sin(0t ) [ X ( + 0 ) X ( 0 )]
2
The above equations can be proved by Eulers formula.
Differential in the time domain

dn
x(t ) ( j ) n X ( )
n
dt
Convolution of two signals in the time domain
x(t ) * v (t ) X ( )V ( )
Integral in the time domain
x( )d
X ( )
+ X (0) ( )
j
In general, the integral of x(t) doesnt have a Fourier transform in the ordinary sense,
but it does have the generalized transform.
Multiplication of two signals in the time domain

x(t )v (t )
1
1
[ X ( ) *V ( )] =
2
2
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x(t )v(t )e jt dt
t =
1
2
1
=
2
1
=
2
1
=
2
=
1
2
t =
x( )e jt d } v(t )e jt dt
x( )e jt v(t )e jt dt} d
x( ){
x( ){
t =
t =
v(t )e jt e jt dt} d
v(t )e j ( )t dt} d
x( )V ( )d
Parsevals theorem
x (t )v (t ) dt
1
2
X ( )V ( ) d =
1
2
V ( ) X ( ) d
If v(t)= x(t),
x 2 (t )dt
1
2
X ( )X ( )d =
1
2
X ( ) d
2
Duality Property
If x (t ) X ( ) is sure,
X (t ) 2 x( )
The duality property is easily proved by the institution of variables and the
definitions of Fourier transform and inverse Fourier transform.
4.5 Generalized Fourier Transform
In the ordinary sense, there are not the Fourier transform for constant signals, unit-step
function and sinusoidal functions because the absolute integral property doesnt hold for
them.
x(t ) dt <
But for the engineering needs, the generalized Fourier transform is considered here.
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The Fourier transform of (t) is given by

(1) (t ) 1
Proof:
(4.75)
0+
0+
(t )e jt dt = (t )e jt dt = (t ) 1dt = 1
(2) 1 2 ( )
(4.76)
Proof: Applying the duality to (4.75) yields the result of (4.76).
1
+ ( )
j
Proof: Applying the integration property to (4.75) yields the result of (4.81).
(3) u (t )
(4.81)
(3) cos(0t ) [ ( + 0 ) + ( 0 )]
(4.77)
(4) sin(0t ) j [ ( + 0 ) ( 0 )]
(4.78)
(4.79)
(5) e j0t 2 ( 0 )
Proof: Applying the Eulers formula, (4.77) and (4.78) yields the result of (4.79).
Interestingly, if a signal x(t) can be decomposed and denoted by
x (t ) =
ce
k =
jk0 t
, <t<
Thus, the Fourier transform of x(t) can be expressed by

X ( ) =
k =
2 ( k0 )
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Common Fourier transform pairs

(1) (t ) 1
(2) 1 2 ( )
1
(3) u (t )
+ ( )
j
(4) (t c) 1 e jc , c is any real number
1
(5) e bt u (t )
, b >0
j + b
(6) 1 e j0t 2 ( 0 )
1
j
1
1
+ ( ) ,
Proof: 0.5 + u (t ) ( )2 ( ) +
2
j
(A1) 0.5 + u (t )
(7) cos(0t ) [ ( + 0 ) + ( 0 )]
(8) sin(0t ) j [ ( + 0 ) ( 0 )]
(A2) e j0t 2 ( 0 )
cos(0t + ) [e j ( + 0 ) + e j ( 0 )]
(complex function)
(10) sin(0t + ) [e j ( + 0 ) e j ( 0 )]
(complex function)
(9)
(A3)
cos(0t + ) = [e j (0t + ) + e j (0t + ) ] / 2
= [e j (0t + ) / 2 + e j (0t + ) / 2]
= (e j e j0t ) / 2 + (e j e j0t ) / 2
sin(0t + ) = [e j (0t + ) e j (0t + ) ] / 2

= [e j (0t + ) / 2 e j (0t + ) / 2]
= (e j e j0t ) / 2 (e j e j0t ) / 2
[cos(0t ) + j sin(0t )]e j = e j (0t + ) = e j0 (t + / 0 ) = 1 e j0t
j / 0
t = t + / 0
2 ( 0 ) = [cos( / 0 ) + j sin( / 0 )]2 ( 0 )
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(What is the mistake for this deduction? Please find out the cause.)
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Signals and Systems

5 FREQUENCY-DOMAIN ANALYSIS OF SYSTEMS

FOUCUS ON LINEAR TIME-INVARIANT SYSTEM, CAUSALITY NOT
ABSOLUTELY NEEDED.
TIME_DOMAIN CONCEPT
For a sinusoidal signal as the input, the output response of the linear time-invariant
system is also a sinusoidal having the same frequency, but which is amplitude-scaled and
phase-shifted.
FREQUENCY-DOMIAN CONCEPT
The Fourier transform of the system output is the product of the Fourier transform of the
input and the frequency response function ( or system function) of the system.
5.1 Response to a Sinusoidal Input
TOPIC: the behavior of the system function of a linear time-invariant system
Consider a linear time-invariant continuous-time system with impulse response h(t). The
output response y(t) resulting from input x(t) is given by the convolution relationship
y (t ) = x( )h(t )d = h( ) x(t )d
(5.1)
NOTE THAT the assumption that h(t) is absolutely integrable is considered in this
Chapter.
h( ) d <
(5.2)
Suppose that the input xc(t) is

xc (t ) = Ae j (0t + )
Applying (5.1) to the input xc(t) yields
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yc (t ) = h( ) Ae j (0 (t ) + ) d
yc (t ) = [ h( )e j0 d ] Ae j (0t + )
= [ h( )e
j0
(5.5)
d ]xc (t )
The response yc(t ) can be expressed in terms of the function H() defined by
H ( ) = [ h( )e j d ]
(Fourier Transform)
(5.6)
Using the definition (5.6) of H(), the response given by (5.5) can be written in the form
yc (t ) = H (0 ) xc (t ),
<t <
Expressing H(0) in polar form gives
H (0 ) = H (0 ) e j(H (0 )
( Also denoted as H (0 ) = H (0 ) (H (0 ) )
yc (t ) = H (0 ) e j(H (0 ) Ae j (0t + )
(5.13)
(5.14)
= A H (0 ) e j (0t + + (H (0 ))
By Eulers formula,
xc (t ) = Ae j (0t + ) = A cos(0t + ) + jA sin(0t + )
Define two sinusoidal functions with frequency 0 and amplitude A, which is real
number.
xR (t ) = Re[ Ae j (0t + ) ] = A cos(0t + )
xI (t ) = Im[ Ae j (0t + ) ] = A sin(0t + )
and
yc (t ) = yR (t ) + jyI (t ) = H (0 ) xc (t ),
Hence, yR (t ) = Re[ H (0 ) xc (t )],
yI (t ) = Im[ H (0 ) xc (t )],
<t <
<t <
<t <
Applying (5.14) to the forgoing equations gives

yR (t ) = Re[ Ae j (0t + ) ] = AH (0 ) cos(0t + + (H (0 ))
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yI (t ) = Im[ Ae j (0t + ) ] = AH (0 ) sin(0t + + (H (0 ))
The system function H() relating to Fig. 5.1 is given by

H ( ) =
1/ Cs
R + 1/ Cs
H ( ) =
s = j
1/ RC
j + 1/ RC
From the foregoing equation, the magnitude function and the phase function of H() are
H ( ) =
1/ RC
+ (1/ RC ) 2
2
(H ( ) = tan 1 ( RC )
For the case of 1/RC=1000, Fig. 5.2 shows the system function of H() in polar form.
The RC circuit as shown in Fig. 5.1 is an example of a low pass filter.
In practice, the frequency with amplitude attenuation being 1/ 2 of the magnitude of
dc component is called the cutoff frequency (bandwidth) of the system.
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5.2 Response to Periodic Inputs

TOPIC: First, the behavior of the system function of a linear time-invariant system
is assumedly well-known. This section will discuss the computation of the response
of the system with a periodic input.
Suppose that the input signal x(t) to the system is periodic so that has the complex
exponential Fourier series
x(t ) =
ce
k =
jk0t
< t <
Hence by linearity and the results in Section 5.1, the output response y(t) is
y (t ) =
y (t ) =
H ( k )c e
k =
k =
jk0t
<t <
H (k0 ) ck e j ( k0t + (H ( k0 )) ,
<t <
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Because the coefficients ck may be complex number, it can be written by
ck = ck e j(ck
and thus
y (t ) =
H (k0 ) ck e j ( k0t + (H ( k0 ) + (ck ) ,
< t <
H (k0 ) ck e j ( (H ( k0 )+ (ck ) e jk0t ,
<t <
k =
or
y (t ) =
k =
Obviously, the output of a linear time-invariant system with periodic input is also
periodic.
Applying the waveform shown in Fig. 5.4 to the RC circuit in Fig. 5.1 yields the output in
the form
y (t ) = 0.5 +
(1)(|k |1) / 2
k =1, 3, 5, 7,...
1
1/ RC
(
)e jk t ,
k jk + 1/ RC
< t <
(The proof is left to the reader.)

and the plots of the output y(t) (terminal voltage across the capacitor) for 1/RC=1,
1/RC=10 and 1/RC=100 are sketched below.
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5.3 Response to Aperiodic Inputs

For a linear time-invariant system, even the input x(t) in not a periodic signal, the output
y(t) resulting from the aperiodic signal x(t) can be found by the convolution property.
y (t ) = x( )h(t )d = h( ) x(t )d Y ( ) = H ( ) X ( )
(5.37)
Applying the inverse Fourier transform to both sides of the -domain representation
(5.37) gives
y (t ) =
1
2
H ( ) X ( )e jt d
(5.40)
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Example 5.5 Response of RC Circuit to a Pulse

Impulse signal in Example 5.5 and its amplitude spectrum and phase spectrum are given
below. The Fourier transform of the input x(t) given in Fig. 5.8 is
X ( ) =
sin( / 2)
( / 2)
The Fourier transform of the RC circuit for 1/RC=1 is given.
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H ( ) =
1/ RC
j + 1/ RC
=
RC =1
1
1
1
e ( tan ( )
=
2
j + 1
1+
In other words, the impulse response of the system h(t) is

0,
h(t ) = 1 (1/ RC )t
,
RC e
t<0
t 0, and RC =1
From (5.37), the Fourier transform of the output is shown below.
Figure 5.12 Output response when RC=1.
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5.4 Analysis of Ideal Filters

An ideal filter is a system that completely rejects sinusoidal inputs of the form
x(t)=A cos(0t), for0 in certain frequency ranges, and does not attenuate sinusoidal
inputs whose frequencies are outside these ranges.
The four popular types of ideal filter are shown below.
In the above discussion on ideal filters, nothing bas been said regarding the phase of the
ideal filters. It turns out that to avoid phase distortion in the filtering process, a filter
should have a linear phase characteristic over the passband of the filter. That is,
(H ( ) = td for all in the filter passband
where td is a fixed positive number.
For example, the output response of a linear phase filter resulting from the input
x(t)=A cos(0t), is given by
y (t ) = H (0 ) e j(H (0 ) A cos(0t 0td )
= A H (0 ) cos(0 (t td ))
Thus the linear phase characteristic results in a time delay of td seconds through the
filter.
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Note that if the input signal is

x(t ) = A0 cos(0t ) + A1 cos(1t )
The output response of the first system is expressed by
y (t ) = A0 H (0 ) cos(0 (t td )) + A1 H (1 ) cos(1 (t td ))
and the output response of the second system is denoted by
y (t ) = A0 H (0 ) cos(0t + C ) + A1 H (1 ) cos(1t + C )
where C is a nonzero constant.
According to the linear phase characteristic, the first system is a linear phase system; but
the second system is phase distortion system because its output is not a time-delayed
version of the input.
Ideal linear-phase lowpass filter

Consider the ideal lowpass filter with the frequency function
e jtd , B B
H ( ) =
others
0,
(5.48)
where td is a positive real number.

From (5.48), the amplitude function of H() is
1,
H ( ) =
0,
B B
others
and the phase function of H() is
td , B B
(H ( ) =
others
0,
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The impulse response of the linear-phase lowpass filter defined by (5.48) can be
computed by taking the inverse Fourier transform of the frequency function H(). First
using the definition of the rectangular pulse, H()can be expressed in the form
H ( ) = p2 B ( )e jtd ,
< <
(5.49)
From Table 4.2 the following transform pair can be found
t
sinc( ) p ( )
2
2
(5.50)
Setting =2B in the foregoing equation gives

B
sin c(
Bt
) p2 B ( )
(5.51)
Applying the time-shift property to the transformation pair (5.51) gives

B
sin c(
B (t td )
) p2 B ( )e jtd
(5.52)
Since the right-hand side of the transform pair (5.52) is equal to H(), the impulse
response of the ideal linear-phase lowpass filter is
h(t ) =
sin c(
B (t td )
),
<t <
(5.53)
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The impulse response h(t) of the ideal linear-phase lowpass filter is plotted in Fig. 5.20.
In Fig. 5.20, it is clear the impulse response h(t) is not zero before t=0, and thus the
ideal linear phase lowpass filter is a noncausal system. In fact, any ideal filter is
noncausal and thus cannot be realized.
Ideal linear-phase bandpass filter

The frequency function of an ideal linear-phase bandpass filter is given in polar form
jtd
e
H ( ) =
0,
, B1 B2
others
The phase function of the ideal linear-phase bandpass filter is plotted in Fig. 5.23.
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For a band-limit ideal linear-phase filter, lets investigate the output response resulting
from different inputs.
t
CASE (I): x(t ) = sin c( ),
<t <
(CASE-1)
From Table 4.2, the input spectrum is

X ( w) = p2 ( w) with bandwidth being equal to 1.
e jtd , B B
For the ideal linear-phase system with H ( ) =
,
others
0,
The output spectrum is
Y ( ) = H ( ) X ( ) = p2 B ( )e jtd p2 ( )
If 2B>2 or B>1, p2 B ( ) p2 ( ) = p2 ( )
and thus the output response is
Y ( ) = p2 ( )e jtd
Using time-shift property gives

y (t ) = sin c(
t td
) = x(t td ),
<t <
If 2B<2 or B<1,
p2 B ( ) p2 ( ) = p2 B ( )
Y ( ) = p2 B ( )e jtd = H ( )
Therefore,
y (t ) = h(t ) =
sin c(
B (t td )
) = B sin c(
B (t td )
),
< t <
Note that the output is not the time-delayed version of the input since in this case the
bandwidth is not wide enough to pass all frequency components of the input (CASE-1).
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t
CASE (II): x(t ) = sin c( ) cos(2t ),
<t <
(CASE-2)
By the modulation property of Fourier transform, the input spectrum is

X ( ) =
[ p2 ( + 2) +p2 ( 2)] with bandwidth being equal to 3.
e jtd , B B
For the ideal linear-phase system with H ( ) =
,
0,
others
The output spectrum is

Y ( ) = H ( ) X ( ) = p2 B ( )e jtd [ p2 ( + 2) + p2 ( 2)]
If B>3, X ( ) = 0 for >3

Y ( ) =
[ p2 ( + 2) + p2 ( 2)]e jtd
Using time-shift property gives

y (t ) = x(t td ) = sin c(
t td
) cos(2(t td )),
<t <
If B<1, X ( ) = 0 for <1. Thus the output spectrum is given

Y ( ) = 0
which implies that

y (t ) = 0,
<t <
If 1<B<3, the product of Fourier transform of the input signal and that of the ideal
linear-phase lowpass system is given in Fig. 5.22 and it is expressed by
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p2 B ( )[ p2 ( + 2) + p2 ( 2)] = pB 1 ( +
B +1
B +1
) + pB 1 (
)
2
2

Y ( ) =
[ pB 1 ( +
B +1
B + 1 jtd
) + pB 1 (
)]e
2
2
Therefore, using the time-shifted and modulation properties over the foregoing equation
yields
B 1
B 1
B 1
y (t ) =
)(t td ) cos (
)(t td ) ,
sinc (
2
2
<t <
Note that the output is not the time-delayed version of the input since in this case the
bandwidth is not wide enough to pass all frequency components of the input (CASE-2).
TRICK
sin c(
t
) 2 p ( )
2
( B 1) sin c(
( B 1)t
) 2 pB 1 ( )
2
F 1 { pB 1 ( )} =
( B 1)
( B 1)t
sin c(
)
2
2
B +1
B +1
1
B +1
) + pB 1 (
)] F 1 { pB 1 ( )} cos(
[ pB 1 ( +
t)
2
2
2
2
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5.5 Sampling Therorm

A band-limited signal can be completely reconstructed from a sampled version of
the signal if the sampling frequency is suitably fast.
In uniform sampling, the sampling operation, also named impulse train, can be expressed
in the mathematical form
p (t ) =
(t nT )
n =
where T is the sampling period.

Hence, the sampled waveform of x(t) is denoted as xT(t) and its representation is given by
x(t ) p (t ) =
x(t ) (t nT ) =
n =
x(nT ) (t nT )
n =
To determine the Fourier transform of x(t) p(t), there are three ways to find it.
(1) Linear property and time-shift property
(2) Convolution operation in frequency domain
(3) The expansion of Fourier series of p(t) time x(t)
The first two ways are left to readers.
Since p(t) is a periodic signal, it has the complex exponential Fourier series
p (t ) =
ce
k =
jks t
, <t <
(5.55)
The coefficients ck of the Fourier series are computed as follows:
1 T /2
p(t ) e jks t dt
T
/
2
T
1 T /2
= (t ) e jkst dt
T T / 2
1
=
T
ck =
Inserting ck into (5.55) yields
1
p (t ) = e jkst , < t <
k = T
and thus
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x(t ) p (t ) =
T x (t ) e
jks t
(5.56)
k =
With Xs() equal to the Fourier transform of xT(t), the Fourier transform of x(t) p(t) is
X s ( ) =
T X ( k )
k =
(5.57)
From (5.57) it is seen that Xs() which is the Fourier transform of x(t) p(t) consists of a
sum of frequency-shifted replica of X() sitting at integer multiples ks for all integer k.
Suppose that a signal x(t) has bandwidth B, that is
X ( ) = 0,
for > B
Then if s 2 B , for Xs() the replica of X() do not overlap in frequency. By this result,
the sampling theorem states that the original input x(t) can be completely reconstructed
from xs(t) by an ideal linear-phase lowpass filter with cutoff frequency B if the sampling
frequency s is chosen to be greater than or equal to 2B. The minimal sampling
frequency s=2B is named the Nyquist (sampling) frequency.
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For an ideal lowpass filter with bandwidth B, (linear-phase)
T (0D , B B
H ( ) =
others
0,
(5.48)
The impulse response of h(t) of this filter is given by

h(t ) =
TB
sinc(
Bt
),
<t <
It can be proved that

y (t ) =
TB
x(nT )
n =
sinc(
B (t nT )
),
B (t nT )
TB
y (t ) =
), < t <
x(nT ) sinc(
n =
This expression is called the interpolation formula for the signal x(t). It shows that if the
sampling frequency is greater than or equal to twice the bandwidth of the input signal, the
input signal can be obtained by the infinite sum of the sinc functions sinc( Bt / ) with
TB
amplitude weighs of
x(nT ) and time shifts at nT for all integer n.

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Signals and Systems

7 FOURIER ANALYSIS OF DISCRETE-TIME SIGNALS AND SYSTEMS
Continuous-time signal
Periodic signal Aperiodic signal
Fourier series
Fourier
(CTFS)
transform (CTFT)
Discrete-time signal
Periodic signal
Aperiodic signal
Discrete-time
Discrete-time
Fourier series
Fourier transform
(DTFS)
(DTFT)DFTFFT
Harmonics
analysis;
Periodic signal
Line spectrum
Since the signal

is aperiodic, the
frequency
spectrum is
continuous.
x[n]= x[n+N0]
X k = X () =2 k / N
X k = X k+N
Fundamental
frequency and
its multiples
X ( ) = x(t )e jt dt
X()= X(+2)
N 1
x[n] = ck e jk 0 n
x[n]e
j n
n =
k =0
Let = 2 / NT
X ( ) = k =
X ( ) =
where 0 =2/ N0
1 e jk T
Xk
jk
N 1
X k = x[n]e jk 2 n / N
n=0
2r
X k = x[n]e jk 2 n / N ,
where
k=0,1,2,,(N-1)
n=0
where N 2 r
z
z
DTFT is also named after Discrete-time Fourier integral (infinite data).

DFT for finite data and FFT for finite data with length of a power of 2
The connection between the bilateral z-transform and the DTFT is similar to
that between Laplace transform and CTFT.
Use of DFT and FFT is for the numerical computation of DTFT.
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Whereas the z-transform is superior to the DTFT for analysis of linear

time-invariant, discrete-time systems, the DTFT is preferable in signal
analysis.
Supplement: Periodic signal representation by discrete-time Fourier series

A periodic signal x[n] with period N0 can be written in the form
x[n] = x[n + N 0 ]
The smallest value of N0 for which this equation holds is the fundamental period. The
fundamental frequency is 0 =2/ N0 rad/sample. An N0periodic signal x[n] can be
represented by a discrete-time Fourier series made up of sinusoids of fundamental
frequency and its harmonics. Unlike the continuous-time Fourier series, the discrete-time
Fourier series has only a finite number (N0) of terms (individual independent harmonics)
due to the fact e jn = e j ( 2 m ) n .
For a continuous-time periodic signal x(t), it can be represented by the periodic complex
exponentials in the form
x(t ) =
ce
k
k =
jk0t
< t <
(4.11)
where c0 is a real number and ck for k0 are in general complex numbers, and 0 is the
fundamental frequency by 0=2/T (CTFS).
Similarly, the discrete-time periodic signal x[n] can be written by
x[ n] =
ce
k =
jk 0 n
where 0 is equal to 2/ N0 rad/sample.

Partitioning this infinite sum into an sum of a group items with length N0 yields
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x[n] =
ce
k =
j(
2 k
)n
N0
N0
= ... + ck e
j(
2 k
)n
N0
= ... + ck e
k = N 0 +1
j(
2 k
)n
N0
k =1
N0
= ... + ck e
= ... + ck e
ck e
j(
N0
+ ck + N0 e
2 k
)n
N0
3 N0
k = 2 N 0 +1
j(
2 ( k + N 0 )
)n
N0
k =+1
j(
2 k
)n
N0
k =1
N0
k =1
N0
2 N0
N0
+ ck + N0 e
k =1
N0
+ ck + N0 e
2 k
)n
N0
N0
+ ck + 2 N0 e
+...
j(
2 ( k + 2 N 0 )
)n
N0
+...
k =+1
j(
2 k
)n
N0
.e
j (2 ) n
k =+1
2 k
j(
)n
N0
ck e
j(
N0
+ ck + 2 N0 e
j(
2 k
)n
N0
. e j (4 ) n +...
k =+1
2 k
j(
)n
N0
k =+1
N0
+ ck + 2 N0 e
j(
2 k
)n
N0
+...
k =+1
The computation of ck is obtained by

T
ck = x[n] e jk0n dn
0
Thus it can be easily proved that

ck = ck + N0 = ck + AN0 , for any integer A
Finally, for convenience to conduct the analysis and synthesis of a signal x(t) and its
discrete Fourier (series) coefficients, the DTFS analysis/synthesis pair is expressed as
follows:
N0
Analysis: X [k ] = x[n]e jk 0 n
(frequency-domain, spectrum)
n =1
or X [k ] =
N 0 1
x[n]e
jk 0 n
n =0
N0
Synthesis: x[n] = X [k ]e jk 0n
(time-domain, space-domain, input sequence)
k =1
or x[n] =
N0 1
X [k ]e
jk 0 n
where 0 is equal to 2/ N0 rad/sample.
k =0
7.1 Discrete-Time Fourier Transform
The Fourier transform of a continuous-time aperiodic signal x(t) was defined by
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X ( ) = x(t )e jt dt , -< <
Given a discrete-time aperiodic signal x[n], the discrete-time Fourier transform of x[n] is
defined by
X ( ) =
x[n]e
j n
(7.2)
n =
The DTFT X() defined by (7.2) is in general a complex-value function of the real
variable . In (7.2), the frequency variable doesnt hold the property 2/ N0 since
the signal x[n] is aperiodic (DTFT).
A discrete-time signal x[n] is said to have a DTFT in the ordinal sense if the bi-infinite
sum in (7.2) converges for all real values of . A sufficient condition for x[n] to have a
DTFT in the ordinal sense is that x[n] is absolutely summable; that is,
x[ n]
(7.3)
n =
For example, if there is a positive integer N such that x[n]=0 for all n-N and nN, then
obviously the sum in (7.3) is finite, and thus any such discrete-time time-limited signal
has a DTFT in the ordinary sense.
For any discrete-time signal x[n], the DTFT X() is a periodic function of with
period 2. It is easy to prove that X()= X(+2) using the definition of DTFT ( Here
the proof is omitted). Therefore, the plot of X() is often given over a 2 interval such as
02 or -.
The rectangular form of X() is given by
X () = R () + jI ()
where
R ( ) =
x[n]cos(n)
n =
I () = x[n]sin(n)
n =
The polar form of X() is
X () = X () e j(X ( )
(7.8)
The computation of the amplitude function and phase function of X() is possible to get
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by using the relationships

X ( ) = R 2 ( ) + I 2 ( )
I ()
(X () = tan 1
R ()
Assuming that the discrete-time signal x[n] is real-valued, the amplitude function of X()
is an even function and the phase function of X() is an odd function; that is
X () = X ()
(X () = (X ()
Proof:
X ( ) =
x[n]e
j n
n =
Replacing the frequency variable by gives

X () =
x[n]e
j ( ) n
n =
= X ()
Using (7.8) with the substitution of with yields
X () = X () e j(X ( )
Taking the complex conjugate of X() gives
X ( ) = X ( ) e j (X ( )
Finally, combining (7.11)-(7.13) yields
X () = X () e j(X ( ) = X () e j(X ( )
Hence it must be true that

X () = X ()
(X () = (X ()
Example 7.2 Consider the discrete-time signal x[n] =(0.5)nu[n]. Find the DTFT of x[n].
Hint: X () =
0.5sin()
1
exp( j tan 1
)
1.25 cos()
1 0.5sin()
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The DTFT of even signals
x[n] = x[ n] R () = x[0] + 2 x[n]cos(n) and I () = 0

n =1
and X () = R(), (X () = 0
The DTFT of odd signals
x[n] = x[ n] R () = x[0] and I () = 2 x[n]sin( n)

n =1
The proofs are left to readers.
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HINTS:
X () =
n =
x[n]e jn = x[n]e jn + x(0) + x[n]e jn

n =
x[n]e
n =
jn
n =1
+ x(0) + x[n]e jn
n =1
n =1
n =1
= x[n]e jn + x(0) + x[n]e jn
= x[n](cos(n) + j sin(n)) + x(0) + x[n](cos(n) j sin(n))

n =1
n =1
n =1
n =1
= x(0) + [( x[n] + x[n]) cos(n)] j [( x[n] x[n]) sin(n)]
Example 7.3
1,
p[n] =
0,
DTFT of rectangular pulse
q n q
others
Find the DTFT of x[n].
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With comparison to Example 4.7, the result is the discrete-time counterpart to the
Fourier transform of the rectangular pulse in the continuous-time case. In addition, there
are ten points of zero crossing over the frequency interval 0.
Example 7.5 Consider the discrete-time signal x[n] =(-0.5)nu[n].
Hint: X () =
0.5sin()
1
exp( j tan 1
)
1.25 + cos()
1 + 0.5sin()
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DTFT transform pair for the discrete-time aperiodic signal
Analysis: X () =
x[n]e
j n
n =
Synthesis: x[n] =
1
2
X ()e jn d or x[n] =
1
2
X ()e
jn
Generalized DTFT
Usage for constant signal not obeying the summable condition given in (7.3)
Example 7.6 Consider the constant signal x[n] =1, for all integers n.
Since
x[n] =
n =
this signal does not have a DTFT in the ordinary sense. But it does have a DTFT in the
generalized form that is defined in the form
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X ( ) =
2 ( 2 k )
k =
The reason to choose the foregoing equation to be the generalized DTFT of the constant
signal follows the property that the inverse DTFT of X() is equal to the constant signal.
To see this, by (7.28),
1
2
1
=
2
x[n] =
[ 2 ( 2 k )]e jn d
k =
2 ()e
jn
= ()e j 0 d
= 1,
for all integers n
Properties of DTFT
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It should be noted that in contrast to the CTFT, there is no duality property for DTFT.
However, there is a relationship between the inverse CTFT and the inverse DTFT as
shown in the last property in Table 7.2.
7.2 Discrete Fourier Transform
Let x[n] is a discrete-time signal and its DTFT is X(). Since X() is in general a
continuous function of frequency variable , it cannot be stored and processed in a
digital computer unless X() can be expressed in a closed form. To implement DTFT in a
digital computer, it is absolutely necessary to discretize it in frequency. This leads the
introduction of discrete Fourier transform (DFT), which is defined below.
Suppose that the discrete-time signal x[n] is zero for all negative integers n<0 and all
positive integers n N, where N is a fixed positive integer. The N-point discrete Fourier
transform Xk is defined by
N 1
X k = x[n]e jk n ,
k = 0,1, 2,..., N 1
(7.33)
n=0
where =2 /N (DFT)
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The frequency spectrum in (7.33) can be expressed in polar form
X k = X k e j(X k
and in rectangular form
X k = R k + jI k
It is easily verified that
N 1
Rk = x[0] + x[n]cos(k n)
n =1
N 1
I k = x[n]sin(k n)
n =1
Inverse DFT
If Xk is the N-point discrete Fourier transform of x[n], then x[n] can be determined from
by applying the inverse DFT given by
x[n] =
1
N
N 1
X
k =0
e jk n ,
n = 0,1, 2,..., N -1
where =2 /N (DFT)
The connection between DFT and DTFT
Target signal: x[n] is zero for all negative integers n<0 and all positive integers n N,
X() is the DTFT of x[n] and Xk is the DFT of x[n]. They both can be expressed by
x[n]e
X ( ) =
j n
n =
N 1
= x[n]e
(7.38)
jn
n=0
and
N 1
X k = x[n]e jk n ,
k = 0,1, 2,..., N 1 with = 2 /N
n=0
That is,
N 1
X k = x[n]e jk 2 n / N
(7.38A)
n=0
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Comparing (7.38) and (7.38A) reveals that
X k = X () =2 k / N
Example 7.10
(7.39)
DTFT and DFT of finite-duration sinusoid
1, n = 0,1, 2,..., 2q
x[n] =
0, all other n
where 0 02
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DFT of Truncated signals

In most applications, we are interested in a part signal with finite samples; that is, only
x[n], for 0nN-1, is known, where N is some positive integer. When N-point DFT is
computed over the truncated signal x[n], X[k] can be easily obtained from (7.39). As
noted in Example 7.10, the bigger the length of the data, the better the frequency
spectrum. In order to obtain in order to get the best representation of X() of the signal
x[n], it may be necessary to extend the data length from N to L, where L>N, as large as
possibly required.
A new truncated signal x[ n] relating to the signal x[n] is denoted by
x(n) n = 0,1, 2,..., L -1
x[n] =
nL
0,
and let i
X () is the spectrum of the truncated signal x[ n] . From (7.39), the L-point DFT
of the truncated signal x[ n] is given by
i
Xk = i
X ( )
= 2 k / L
k = 0,1,..., L 1
(7.41)
Block effect arises from the boundary problem in which the values are not small for
n N (or n L ) so that the DTFT i
X ( ) of the truncated signal may differ
significantly from the DTFT X() of the actual signal x[n].
0nN-1
N-point x[n]
X k = X () =2 k / N , k [0, N 1]
0nN-1L-1
N-point x[n]
i
Xk = i
X ()
, k [0, L 1]
Boundary
problem
= 2 k / L
The DTFT X() of the signal x[n]

N-point DFT ( X k = X () =2 k / N ) of N-point x[n] ( x N [ n] )
L-point DFT ( j
Xk = i
X ()
= 2 k / L
) of L-point x[n] ( x L [ n] )
The mathematic analysis on the truncated signals

Consider the truncated signal with length of N which is an odd integer. By definition of
the rectangular pulse p[n], setting q=(N-1)/2 gives
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p[n
N 1 1, 0 n N 1
]=
2
0, all other n
Hence the truncated signal of x[n] can be expressed in the form

N 1
x (n) = x[ n] p[ n
]
2
(7.42)
Let P() denote the DTFT of p[n

p[n
N 1 sin( N / 2) j ( N 1) / 2
]=
e
2
sin( / 2)
N 1
] . From the results of Example 7.11, P() is
2
By the multiplication property in time domain, the DTFT of both sides of (7.42) yields
1
i
X ( ) = X ( ) * P ( ) =
2
x[ ] p[ ]d
Note that if x[n] is not suitably small for nN, in general the side lobes that exist in the
amplitude spectrum of |P()| will result in the side lobes in the amplitude spectrum of
|P() *X()|. This leakage phenomenon can be first observed in Example 7.11
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7.3 Properties of the DFT
7.4 System Analysis via the DTFT and DFT

Analysis of an ideal lowpass filter
x(n)
linear discrete-time
time-invariant system
h[n]
y(n)
X()
linear discrete-time
time-invariant system
H()
Y()
y[n] = h[n]* x[n] = h[i]x[n i]X () Y ()=H () X ()

i =
h[n] is the uni-pulse response of the linear discrete-time time-invariant system. Note that
the assumption of h[n] being causal is not absolutely necessary. It is assumed that the
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unit-pulse response is absolute summable; that is
h[ n]
n =
As a result of the sammability condition, the ordinary DTFT H() of h[n] is given by
H () =
h[n]e
j n
n =
Hence taking the DTFT of the convolution of the input signal x[n] and the unit-pulse
response h[n] yields
Y ()=H () X ( )
Illustration: H () =
k =
p2 B ( + 2 k )
x[n] = A cos(0 n)
That is,
X () =
A [ ( +
k =
2 k ) + ( 0 2 k )]
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From the plots of H() and X() given previously in Figures 7.29 and 7.30, it is clear
that Y() is equal to X() when0B and is identically zero if 0B. Thus in
mathematical form,
X [],
Y ( ) =
0,
, if 0 B <
, if B < 0 <
As a result of the periodicity of H() and X(), the output response y[n] is equal to x[n]
when
2 k B 0 2 k + B, where k is non-negative integer.
Digital-filter realization of an ideal analog lowpass filter
x(t)
x[n]= x(nT)
The concerning matter is the sampling period T ( i.e. the sampling frequency s=2/T)
and the bandwidth B of the input signal x(t). The reminder is similar to the treatment
discussed in the last topic (analysis of an ideal lowpass filter).
Unit-pulse response of ideal lowpass filter
The frequency spectrum of the ideal lowpass filter is
H () =
k =
p2 B ( + 2 k )
From the transform pairs in Table 7.1, the unit-pulse response of ideal lowpass filter is
given by
h( n) =
sinc(
nB
),
n = 0, 1, 2,...
From Figure 7.31, it is seen that h[n] is not zero for n<0, and thus the ideal lowpass
filter is noncausal.
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A filter with the following input/output relationship

y ( n) =
1
N
N 1
x[n i],
where N 3
i =1
is called mean filter which possesses the sharp frequency cutoff when N3. The mean
filter is an example of a causal lowpass discrete-time filter.
The plot of amplitude spectrum of the causal lowpass filter is left to readers.
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POPULAR CASE:
x[n]=0 for n<0 and nN and h[n]=0 for n<0 and n>Q. Find the output response y[n]
x[n]=0 for n<0 and nN

h[n]=0 for n<0 and n>Q
y[n]
Yk = H k X k Y (
y ( n) =
n ranges from 0 to N-1

n ranges from 0 to Q
n ranges from 0 to N+Q-1
2 k
2 k
2 k
) = H(
)X (
),
N +Q
N +Q
N +Q
N
Q+1
N+Q
k = 0,1,..., ( N + Q 1)
1 N + Q 1
H k X k e2 kn /( N +Q ) , where n = 0,1,..., ( N + Q 1)
N + Q k =0
(7.64)
It is important to note that the right-side of (7.64) is only coarse approximation of y[n] if
x[n] is not zero for nN and/or h[n] is not zero for n>Q.
The expression of (7.64) is a close approximation to the true values of y[n] only if x[n] is
samll for nN and/h[n] is samll for n>Q.
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7.5 FFT Algorithm
7.6 Applications of the FFT Algorithm

(A.) Computation of the Fourier Transform via the FFT
X ( ) = [ x( )e j d ]
(Fourier Transform)
It is assumed that x(t)=0 for all t<0 so that the Fourier transform of x(t) is given by
X ( ) = [ x(t )e jt dt ]
0
Suppose that the sampling period T is very small enough so that the variation in x(t) is
small over each T-period interval nTt<nT+T.
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X ( ) = [
nT
n=0
= [
n =0
nT +T
nT +T
nT
x(t )e jt dt ]
e jt dt ]x(nT )
nT +T
1
jt
x(nT )
=
e
n = 0 j
nT
jT
1 e
j nT
=
x(nT )e
j n =0
Now suppose that for some large positive integer N, the magnitude x(nT) is small for
nN (tNT). Then the Fourier transform of x(t) is
1 e jT N 1
j nT
X ( ) =
x(nT )e
j n =0
(7.77)
Evaluating both sides of (7.77) at = 2 k / NT
2 k 1 e j 2 k / N N 1
j 2 nk / N
X(
)=
x(nT )e
NT
j 2 k / NT n =0
(7.78)
By the definition of the DFT

N 1
X k = x[ n]e j 2 kn / N
(7.79)
n=0
Comparing (7.78) and (7.89) reveals that
X(
2 k 1 e j 2 k / N
)=
NT
j 2 k / NT
Xk
(7.80)
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(B.) Fast convolution via both FFT and IFFT
As mentioned before, when x[n]=0 for n<0 and nN and h[n]=0 for n<0 and n>Q,
the response of x[n] ( the convolution of x[n] * h[n]) is given by
1 N + Q 1
y ( n) =
H k X k e 2 kn /( N + Q ) , where n = 0,1,..., ( N + Q 1)
N + Q k =0
For instance, the unit-pulse response h[n] of the discrete-time system h[n] = (0.8) n u[n]
and the rectangular input x[n]=1 for r 0 < n 9. In this example, there is not finite
integer Q for which h[n]=0 for n>Q. However, it is true that h[n] is small for n>16, and
thus Q can be taken to be equal to 16. Since the input is zero for all n10, the integer N
defined above is equal to 10. Thus with Q=16, N+Q=26, and the smallest integer value
of r for which N+Q2r is 5. With L=25 =32, the L-point DFT (using FFT algorithm) can
be used to compute the DFT of y[n]. Consequently, the convolution of x[n] * h[n]) is
equal to the inverse DFT of Y[k].
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IMPORTANT TIPS
Fourier transform is a way to analyze the frequency component of a time-oriented

function; oppositely, the discrete-time Fourier transform is applied to analyze the
frequency component of a sequence-oriented function. The spectrum of the former spans
from infinite to finite; but the spectrum of the latter appears periodic with a periodic
spectrum ranging from to +. Both can come together through the periodic sampling
which bridges the gap between Fourier transform and Discrete-time Fourier transform by
Equation (7.80).
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Signals and Systems

1.1 Signals and Systems

2. The convolution model,

1.2 Continuous-Time Signals

for any real number >0

Continuous and Piecewise-Continuous Signals

A continuous signal x(t) is said to be piecewise continuous if it is continuous at all points

has a limit as h->0.

x[n] = x(t ) t = nT = x(nT ) where T is the sampling period.

1.4 Examples of Systems

Car on a level surface

(t ) and the foregoing equation can be approximated by

Example of discrete-time system

y[ n] 1 + y[ n 1] = x[ n], n =0, 1, 2, ...

1.5 Basic System Properties

A continuous-time system with memory is infinite dimensional if it is not finite

For instance, system with time delay as given below,

Linear finite-dimensional system

= ai (n) y[n i ] + bi (n) x[n i ]

Linear time-invariant finite-dimensional system

i, ai (t ) and bi (t ) are constants, and N 1, M > 0

= ai (n) y[n i ] + bi (n) x[n i ]

Signals and Systems

Chapter 2 System defined by Differential or difference equations

Here it is assumed that i, ai (t ) and bi (t ) are real constants,

Here it is assumed that i, ai and bi are real constants,

This gives that

q(t ) = q(0 )e ( at ) + e a (t ) (b0 ab1 ) x( )d , t 0

Thus, y(t) can be yields by substituting y (t ) = q(t ) + b1 x(t ) into (2.12)

y (t ) = b1 x(t ) + [ y (0) b1 x(0)]e ( at ) + e a (t ) (b0 ab1 ) x( )d , t 0

y (t ) = b1 x(t ) + [ y (0 ) b1 x(0 )]e ( at ) + e a (t ) (b0 ab1 ) x( ) d , t 0

= b1u (t ) +y (0 )e ( at ) + + e a ( t ) (b0 ab1 )d , t 0

+ (b0 ab1 )e at (1/ a)[1 e a ( t ) ]d , t 0

2.2 System Modeling

Rotational mechanical systems

2.3 Linear Input/Output Difference Equations with Constant Coefficients

y[n] + ai y[n i ] = bi (n) x[n i ]

i, ai and bi are real constants, and N 1, M > 0

y[n] = ai y[n i ] + bi x[n i ] , n = 0,1, 2,....

For an infinite dimensional difference equation its solution cannot be computed

First, setting n=1, n=2, n=3 in (2.39) gives

2.4 Discretization in Time of Differential Equations

The derivative of y(t) can be approximated by

Inserting the approximation (2.46) into (2.45) gives

Let x[n] = x(t ) t =nT and y[n] = y (t ) t =nT

Consider a linear time-invariant continuous-time system with the second-order

y[n] + (a1T 2) y[n 1] + (1 a1T + a0T 2 ) y[n 2]

2.5 Systems Defined by Time-Varying or Nonlinear Equations

Thus, taking the derivative of both sides of the previous equation

The solution of a linear input/output differential equation with time-varying coefficients

y[n] = f ( y[n 1], y[n 2],..., y[ n N ], x[ n], x[ n 1],..., x[ n M ], n),

= ai (n) y[n i ] + bi (n) x[n i ]

Nonlinear input/output difference equations

Signals and Systems

x[n] = x[0] [n] + x[1] [n 1] + x[2] [n 2] + ...

The response to x[i ] [n i] is given by

yi [n] = x[i ] [n i ]h[n i ]

3.2 Convolution of Discrete-Time Signals

If x[n] is zero for n<0, this convolution operation is given by

Similarly, if v[n] is zero for n<0, the convolution operation is given by

The graphical illustration of the convolution operation is given below.

Compute the convolution by tabulation for special cases.

The convolution of x[n] and v[n] written in the form