Signals and Systems Course Overview
Uploaded by
Rakesh Sharma.MSignals and Systems Course Overview
Uploaded by
Rakesh Sharma.MII YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
PREPARED BY
JAYANTHI.E,AP/ECE
MSAJCE
1
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
EC3354 SIGNALSANDSYSTEMS
OBJECTIVES:
To understand the basic properties of signal &systems
To know the methods of characterization of LTI systems in timedomain
To analyze continuous time signals and system in the Fourier and Laplacedomain
To analyze discrete time signals and system in the Fourier and Z transformdomain
UNITI CLASSIFICATION OF SIGNALSANDSYSTEMS 12
Standard signals- Step, Ramp, Pulse, Impulse, Real and complex exponentials and
Sinusoids_ Classification of signals – Continuous time (CT) and Discrete Time (DT)
signals, Periodic & Aperiodic signals, Deterministic & Random signals, Energy &
Power signals - Classification of systems- CT systems and DT systems- – Linear &
Nonlinear, Time-variant & Time-invariant, Causal & Non-causal, Stable & Unstable.
UNITII ANALYSIS OF CONTINUOUSTIMESIGNALS 12
Fourier series for periodic signals - Fourier Transform – properties- Laplace
Transforms and properties
UNITIII LINEAR TIME INVARIANT CONTINUOUSTIMESYSTEMS 12
Impulse response - convolution integrals- Differential Equation- Fourier and Laplace
transforms in Analysis of CT systems - Systems connected in series / parallel.
UNITIV ANALYSIS OF DISCRETETIMESIGNALS 12
Baseband signal Sampling – Fourier Transform of discrete time signals (DTFT) –
Properties of DTFT- Z Transform &Properties
UNITV LINEAR TIME INVARIANT-DISCRETETIMESYSTEMS 12
Impulse response – Difference equations-Convolution sum- Discrete Fourier
Transform and Z Transform Analysis of Recursive & Non-Recursive systems-DT
systems connected in series and parallel.
TOTAL: 60
PERIODS
OUTCOMES:
At the end of the course, the student should be able to:
To be able to determine if a given system islinear/causal/stable
Capable of determining the frequency components present in a deterministicsignal
Capable of characterizing LTI systems in the time domain and frequencydomain
To be able to compute the output of an LTI system in the time and frequencydomains
TEXT BOOK:
1. Allan [Link], [Link] and [Link], “Signals and Systems”, Pearson,
2015.(Unit 1- V)
REFERENCES
1. B. P. Lathi, “Principles of Linear Systems and Signals”, Second Edition, Oxford,2009.
2. [Link], [Link] and [Link], “Signals & Systems - Continuous
and Discrete”, Pearson,2007.
3. John Alan Stuller, “An Introduction to Signals and Systems”, Thomson,2007.
2
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Table of Contents
[Link] Contents [Link]
Unit I- Classification of Signals and Systems
1.1 Introduction 5
1.2 Continuous time and Discrete time signals 5
1.3 Elementary signals 6
1.3.1 Unit step signal 6
1.3.2 Ramp signal 6
1.3.3 Unit impulse signal 6
1.3.4 Sinusoidal signal 6
1.3.5 Exponential signal 6
1.4 Classification of CT and DT signals 7
1.5 CT and DT systems 8
1.6 Classification of systems 8
1.6.1 Static and Dynamic 8
1.6.2 System with memory and without memory 9
1.6.3 Time variant and time invariant 9
1.6.4 Linear and Non linear 10
1.6.5 Causal and non causal 10
1.6.6 Stable and Unstable 11
Unit II – Analysis of Continuous time signals
2 Fourier series analysis 21
2.1 Fourier transform 22
2.2 Inverse transform 22
2.3 Laplace transform 26
2.5 Properties of ROC of Laplace transform 50
Unit III – Linear Time invariant continuous time s ystem
3.1 System 51
3.2 LTI System 51
3.3 Block diagram representation 52
3.4 Impulse response 52
3.5 Properties of impulse response 53
3.6 Convolution integral 53
3.6.1 Convolution Integral Properties 53
i
3
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3.7 Properties of Laplace transform 53
3.8 Structure realization 55
Unit IV – Analysis of Discrete Time systems
4.1 Sampling Theory 72
4.2 Aliasing 73
4.3 Sampling of Non-bandlimited Signal: Anti-aliasing Filter 76
4.4 Discrete time fourier transform 77
4.5 Inverse transform 78
4.6 Properties 79
4.7 Symmetry properties 80
4.8 Z-transform 82
4.9 Properties 86
Unit V – Linear time invariant discrete time syste m
5.1 Introduction 99
5.2 Block Diagram representation 99
5.3 Convolution Sum 101
5.4 LTI system analysis using DTFT 108
5.5 LTI system analysis using Z transform 112
ii
4
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
UNIT – I
CLASSIFICATION OF SIGNALS AND SYSTEMS
1.1 INTRODUCTION:
A signal, as stated before is a function of one or more independent variables. A signal
is a quantitative description of a physical phenomenon, event or process. More precisely, a signal
is a function, usually of one variable in time. However, in general, signals can be functions of
more than one variable, e.g., image signals. Signals are functions of one or more variables.
Systems respond to an input signal by producing an output signal .
Examples of signals include:
1. A voltage signal: voltage across two points varying as a function oftime.
2. A force pattern: force varying as a function of 2-dimensionalspace.
3. A photograph: color and intensity as a function of 2-dimensionalspace.
4. A video signal: color and intensity as a function of 2-dimensional space andtime.
A continuous-time signal is a quantity of interest that depends on an independent
variable, where we usually think of the independent variable as time. Two examples are the
voltage at a particular node in an electrical circuit and the room temperature at a particular spot,
both as functions oftime.
A discrete-time signal is a sequence of values of interest, where the integer index can be
thought of as a time index, and the values in the sequence represent some physical quantity of
interest.
A signal was defined as a mapping from a set of the independent variable (domain) to the
set of the dependent variable (co-domain). A system is also a mapping, but across signals, or
across mappings. That is, the domain set and the co-domain set for a system are both sets of
signals, and corresponding to each signal in the domain set, there exists a unique signal in the co-
domain set.
System description
The system description specifies the transformation of the input signal to the output
signal. In certain cases, a system has a closed form description. E.g. the continuous-time system
with description y (t) = x(t) + x(t-1); where x(t) is the input signal and y(t) is the output signal.
1.2 Continuous-time and discrete-timesystems
Physically, a system is an interconnection of components, devices, etc., such as a
computer or an aircraft or a powerplant.
Conceptually, a system can be viewed as a black box which takes in an input signal x(t)
(or x[n]) and as a result generates an output signal y(t) (or(y[n]).
A system is continuous-time (discrete-time) when its I/O signals are continuous-time
(discrete-time).
1.3 ElementarySignals:
The elementary signals are used for analysis of systems. Such signals are,
Step
Impulse
Ramp
Exponential
Sinusoidal
5
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1.3.1 Unit stepsignal:
Unit Step Sequence: The unit step signal has amplitude of 1 for positive value and
amplitude of 0 for negative value of independentvariable.
It have two different parameter such as CT unit step signal u(t) and DT unit step signal
u(n).
The mathematical representation of CT unit step signal u(t) is givenby,
1.3.2 RampSignal:
The amplitude of every sample is linearly increased with the positive value of
independentvariable.
Mathematical representation of CT unit ramp signal is givenby,
1.3.3 Unit impulsefunction:
Amplitude of unit impulse approaches 1 as the width approaches zero and it has zero
value at all othervalues.
The mathematical representation of unit impulse signal for CT is givenby,
It is used to determine the impulse response ofsystem.
1.3.4 Sinusoidalsignal:
A continuous time sinusoidal signal is givenby,
Where, A– amplitude - phase angle inradians
1.3.5 Exponentialsignal:
It is exponentially growing or decayingsignal.
Mathematical representation for CT exponential signalis,
6
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1.4 Classification of CT and DTsignals:
Periodic and non-periodicSignals
A periodic function is one which has been repeating an exact pattern for an infinite period of
time and will continue to repeat that exact pattern for aninfinite time. That is, a periodic function
x(t) is one for which
x (t) = x(t+nT)
for any integer value of n, where T >0 is the period of the function and − ∞ < t <∞ . The signal
repeats itself every T sec. Of course, it also repeats every 2 T,3T and nT. Therefore, 2T, 3T and
nT are all periods of the function because the function repeats over any of those intervals. The
minimum positive interval over which a function repeats itself is called the fundamental period
T0.T0 is the smallest value that satisfies the condition x ( t ) = x ( t+T0). The fundamental
frequency f 0 of a periodic function is the reciprocal of the fundamental period f 0=1/T0. It is
measured in Hertz and is the number of cycles (periods) per second. The fundamental angular
frequency ω0 measured in radians per second is ω0=2πT0= 2πf0. A signal that does not satisfy
the condition in (2.1) is said to be a periodic or non-periodic.
Deterministic and RandomSignals
Deterministic Signals are signals who are completely defined for any instant of time, there is no
uncertainty with respect to their value at any point of time. They can also be described
mathematically, at least approximately. Let a function be defined as
A random signal is one whose values cannot be predicted exactly and cannot be described by
any exact mathematical function, they can be approximately described.
Energy and PowerSignals:
Consider v(t) to be the voltage across a resistor R producing a current i(t). The
7
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
instantaneous power p(t) per ohm is defined as, Total energy E and average power P on a per-
ohm basis are
For an arbitrary continuous-time signal x (t), the normalized energy content E of x(t) is
defined as,
The normalized average power P of x (t) is defined as,
Similarly, for a discrete-time signal x[n], the normalized energy content E of x[n] is defined as,
The normalized average power P of x[n] is defined as,
1.5 CT Systems and DTSystems:
A system is defined as a physical device which contains set of elements or functional
blocks and that generates a response or output signal for a given input.
1.6 Classification ofsystem:
The systems are classified as,
Static & dynamicsystem
Time invariant and variantsystem
Linear and non linearsystem
Causal and non causalsystem
Stable and unstablesystem
1.6.1 Static and dynamic system:
Static system is said to be a memorylesssystem.
The output does not depend the past or futureinput.
It only depends the present input for anoutput.
8
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Eg, y(n) = x(n)
Dynamic system is said to be as system withmemory.
Its output depend the past values of input for anoutput.
Eg. Y(n) = x(n) + x(n - 1)
This static and dynamic systems are otherwise called as memoryless and system with
memory.
1.6.2 Systems with and withoutmemory:
A system is called memory less if the output at any time t (or n) depends only on the
input at time t (or n); in other words, independent of the input at times before of after t
(or n). Examples of memory lesssystems:
Examples of systems with memory:
1.6.3 Time invariant and time variantsystem:
If the time shifts in the input signals results in corresponding time shift in the output,
then the system is called as time invariant.
The input and output characteristics do not change withtime.
For a continuous timesystem,
f[x(t1 – t2)] = y(t1 – t2)
For a discrete time system,
F[x(n - k)] = y(n - k)
If the above relation does not satisfy, then the system is said to be a time variantsystem.
A system is called time-invariant if the way it responds to inputs does not change over
time:
Examples of time-invariant systems:
The RC circuit considered earlier provided the values of R or C areconstant.
Examples of time-varying systems:
The RC circuit considered earlier if the values of R or C change overtime.
9
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Most physical systems are slowly time-varying due to aging, etc. Hence, they can be
considered time-invariant for certain time periods in which its behavior does not
changesignificantly.
1.6.4 Linear and non linearsystem:
A system is said to be linear if it satisfies the superpositionprinciple.
Superposition principle states that the response to a weighted sum of input signal be
equal to the weighted sum of the output corresponding to each of the individualinput
signal
The continuous system is linearif,
F[a1x1(t) + a2x2(t)] = a1y1(t) + a2y2(t)
The discrete system is linearif,
F[a1x1(n) + a2x2(n)] = a1y1(n) + a2y2(n)
Otherwise the system is non linear.
A system is called linear if its I/O behavior satisfies the additivity and homogeneity
properties:
for any complex constant a.
Equivalently, a system is called linear if its I/O behavior satisfies the superposition
property:
where any complex constants a and b.
1.6.5 Causal and non causalsystem:
A causal system is one whose output depends upon the present and past inputvalues.
If the system depends the future input values, the system is said to be non causal.
Eg. for causalsystem.
Y(t) = x(t) + x(t - 1)
Y(n) = x(n) + x(n - 3)
Eg. For non causal system,
Y(t) = x(t+3) + x2(t)
Y(n) = x(2n)
10
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
A system is called causal or non-anticipative if the output at any time t (or n) depends
only on the input at times t or before t (or n or before n); in other words, independent of
the input at times after t (or n). All memory less systems are causal. Physical systems
where the time is the independent variable arecausal.
Non-causal systems may arise in applications where the independent variable is not
the time such as in the image processingapplications.
Examples of causal systems:
Examples of non-causal systems:
1.6.6 Stable and unstablesystem:
When every bounded input produces bounded output then the system is called as
stable system or bounded input bounded output (BIBOstable).
Otherwise the system is unstable.
A system is called stable if it produces bounded outputs for all boundedinputs.
Stability in a physical system generally results from the presence of mechanisms that
dissipate energy, such as the resistors in a circuit, friction in a mechanical system,etc.
Sample Problems:
Problems:
Determine whether the following system is linear or not.
dy
3ty(t) t 2x(t)
dt
For an input x1(t) and for the corresponding output y1(t) the differential equation can be written as,
dy1(t)
3ty(t)t2 x(t) --------(1)
1 1
dt
For an input x2(t) and for the corresponding output y2(t) the differential equation can be written as,
dy2 (t)
3ty (t) t 2x (t) --------(2)
2 2
dt
(1) a (2) b and rearrranging,
d[ay1 (t) by2 (t)]
3t[ay (t) by (t)] t 2[ax (t) bx (t)]
1 2 1 2
dt
d
[ay (t) by (t)] 3t [ay (t) by (t)] t 2[ax (t) bx (t)]
1 2 1 2 1 2
dt weighted sum of output weighted sum of output weighted sum of input
11
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
From the above equation, we can note that the weighted sum of inputs to the system produces an
output which is also equal to weighted sum of outputs corresponding to each of the individual
inputs.
Therefore, the system is linear.
1
Determine whether the system y(n) 2x(n) is linear or not
x(n 1)
1
y(n) T[x(n)]2x(n)
x(n 1)
For an input x1(n),
1
y(n)T[x(n)]2x(n) ----------------(1)
1 1 1
x1 (n 1)
For an input x2(n),
1
y (n) T[x (n)] 2x(n) ------------------(2)
2 2 2
x2 (n 1)
Weighted sum of outputs is given by
a b
ay (n) by (n) 2ax (n) 2bx(n) -------------(3)
1 2 1 2
x1 (n1) x2(n1)
Output due to weighted sum of inputs is
1
y(n)T[ax(n)bx(n)]2[ax(n)bx(n)] --------(4)
3 1 2 1 2 [ax(n1)bx(n1)]
1 2
y3 (n) ay1(n) by2 (n)
Therefore, the system is non-linear.
Determine whether the following systems are time-invariant or not.
(1) y(t) tx(t)
y(t) T[x(t)] tx(t)
The output due to delayed input is
y(t,T ) T[x(t T )] tx(t T )
If the output is delayed by T, we get
y(t T ) (t T )x(t T )
y(t,T ) y(t T )
Therefore, the system is time-variant
(2) y(n) x(2n)
y(n) T[x(n)] x(2n)
If the input is delayed by k units of time then the output is ,
y(n, k) T[x(n k)] x(2n k)
Outputdelayedbykunitsoftimeis,
y(nk)x[2(nk)]x[2n2k]
y(n, k ) y(n k )
Therefore, the system is time –variant.
12
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Problem
Find whether the signal xt 2cos(10t 1) sin(4t 1) is periodic or not.
Time period of 2cos10t 1is T 2sec
1 10 5
2
Time period of sin(4t 1) is T sec
4 2
2
T 2
The ratio of two periods is 1 5
T2 5
2
The ratio of two periods is a rational number.
Therefore, the sum of two signals are periodic and the period is given by
T 2T2 5T1 2 5 sec
2 5
ii) Find thesummation e
n
2n
(n2)
(n 2) 1 for n 2
0 for n 2
e2n(n2)e2n(n2)|
n
n2 e4
iii) Explain the properties of unit impulsefunction.
unit impulse function
0.8
( t)
0.6
0.4
0.2
0
-4 -3 -2 -1 0 1 2 3 4
t
The impulse function has zero amplitude everywhere except at t = 0.
(t)1 t 0
0 t 0
At t=0, the amplitude is infinity such that the area under the curve is equal to one.
(t)dt 1
13
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
iv) Find the fundamental period T of the continuous timesignal xt 20cos 10t
6
xt20cos 10t
6
0 10
2 2 1 sec
T
0 10 5
Derive the relationship between unit step and delta function.
( t ) i
u(t)
ntegrate
u (t ) d (t)
ifferentiate
(t ) unit impulse function
u (t ) unit step function
u(t) 1 t 0
0 t 0
du(t)
1 t0
dt
0 t0 (t)
Sketch the following signals:
(1) xtrt
(2) xtrt 2
14
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
(3) xt2rt where r(t) is a ramp signal.
.Determine whether the following systems are time invariant or not.
i) Y(t) =tx(t)
ii) Y(n) =x(2n)
Solution:
i) Y(t) =tx(t)
Y(t) = T[x(t)] = tx(t)
The output due to delayed input is,
Y(t,T) = T[x(t - T)] = tx(t - t)
If the output is delayed by T, we get
Y(t -T) = (t - T) x( t - T)
The system does not satisfy the condition, y(t,T) = y(t – T).
Then the system is time invariant.
ii) Y(n) = x(2n)
Y(n) =x(2n)
Y(n) = T[x(n)] = x(2n)
If the input is delayed by K units of time then the output is,
Y(n,k) = T[x(n-k)] = x(2n-k)
The output delayed by k units of time is,
Y(n-k) = x[2(n-k)]
Therefore, y(n,k) is not equal to y(n-k). Then the system is time variant.
15
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1.
16
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2.
17
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
18
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
19
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
20
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
UNIT – II
ANALYSIS OF CONTINUOUS TIME SIGNALS
2.1 Fourier Series representation of CT periodicsignals
Any periodic function of time x(t) can be represented by an infinite series called the Fourier Series.
(i) Trigonometric FourierSeries
x(t)a 0a ncosn 0t bnsinn o t
n1 n1
a0 , an , bn are Fourier Coefficients
1
0 2 f where f fundamental frequency &T
f
T
1
a0
T x(t)dt
0
T
an T2 x(t)cos(n0t)dt
0
2T
bn
T x(t)sin(n t)dt
0
0
(ii) CosineSeries
x(t)a0 a ncosn 0 t bnsinn o t
n1 n1
a0 , an , bn are FourierCoefficients
x(t)a0 a ncosn 0 tbnsinn 0 t
n1
x(t)A0Ancosn 0 tn
n1
where A0 a0
An an2b2 n
1bn
n tan
an
note:Trigonometric identity
1
A
Asin C B cosC A2B2 cosC tan B
(iii) Exponential FourierSeries
x(t) C
n
n
e jn0t
21
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1
C
T
jn0t
n x(t)e dt
T
2.2 Properties of CT FourierSeries
Using exponential Fourier series, a periodic signal x(t) with period T can be expressed as
x(t) C
n
n
e jn0t
1
C
T
jn0t
n x(t)e dt
T
22
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1. Linearity
If x1 (t) and x2(t) are two periodic signals with period T with Fourier series coefficients
Cn1 and Cn2 then Fourier series coefficient of linear combination of x1 (t) and x2 (t) is given by
FS Ax1(t)Bx 2(t)AC n1BC n2
Proof:
1 Ax (t)Bx
Fourier seriescoefficientofAx1(t)Bx2 (t)is= 2 (t) e
jn 0t
1 dt
1 1 T
t
Ax(t)e jn0 dt T
Bx(t)e jn0tdtAC BC
TT TT
1 2 n1 n2
2. TimeShifting
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient of the time shifted
signal x(t t0 ) is e jn t C n
0 0
1
x(t t )ejn0t dt
Proof:Fourier series coefficient of x(t t0 ) is =
TT 0
let t t0 m t m t0 &dt dm
1
x(m)ejn (mt )dm0 0
TT
1
ejn t x(m)e jn m dme jn t C
0 0 0 0 0
n
TT
3. TimeReversal
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient of the time reversed
signal x(t) is Cn
1
Proof: Fourier series coefficient of x(t) is =
TT
x(t)e jn0tdt
let t m t m &dt dm
1
x(m)e jn0(m)dmC
T
T
n
4. Timescaling
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient of the time scaled
1
signal x(t) is Cn /
1
Proof:Fourier series coefficient of x(t) is =
T
x(t)e jn dtT 0t
m dm
let t m t &dt
23
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1 1 jn0 m
dm
T Tx(m)e
n
1 1 jm0 1
dm
T Tx(m)e
Cn /
5. Conjugation
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient of the complex
conjugate of the signal x(t) is C n
Proof:
x(t) C
n
n
e jn0t taking complex conjugate on both sides,
x (t)
C e
n
n jn0t
let l n
C
jl0t
x (t) l e changing the variable
l
x(t) C
n
ejn0t
FS x (t) Cn
n
6. Differentiation
dx(t)
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient of the signal
dt
is jn0Cn
Proof:
x(t) C
n
n
e jn0t differentiatingw.r.t t on both sides,
dx(t)
dt Cejn
n
n
0t
jn 0
dx(t) dx(t)
jn0t
dt jn0Cn e
n
FS
dt
jn0 Cn
7. Integration
If the Fourier series coefficient of x(t) is Cn then the Fourier series coefficient ofthe signal
t
Cn
x(t)dt is
jn0
Proof:
x(t) C
n
n
e jn0t integratingw.r.t t on both sides,
t
1
x(t)dt C n ejn t 0
jn0
n
24
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
t C
t
C
x(t)dt n
ejn t
0 FSx(t)dt n
jn
n 0 jn0
8. Multiplication
If x (t) and y(t) are two periodic signals with period T with Fourier series coefficients Cn and Dn
then Fourier series coefficient of product of x(t) and y(t) isgiven by C D
l
l (nl)
Proof:
x(t) Cn ejn0t x(t) C e l
jl0t
n l
1
Fourier series coefficient of x(t) y(t) is =
T
x(t)y(t)e jn dt 0t
T
1
y(t)e jn dt 0t
C e jl0t
l l
TT
1
= Cl y(t)ej(nl) tdt 0
l
TT
ClD(nl)
l
[Link]’s Relation for periodic signal
1
If the Fourier series coefficient of x(t) is Cn then,
T T
x(t) 2 dt C n
2
n
1 2
x(t)
T
dt average power in the signal
T
Proof:
1 1 (t)]dt 1
C jn t
T x(t) 2
dt T [x(t)x T x(t)
n e 0 dt
T T T n
1
x(t) jn t
T dt C n Cn C n
2
Cn e 0
n n
T n
25
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2.3 laplacetransform
The Unilateral Laplace Transform
The Unilateral Laplace Transform is applied to the signals that are causal. The Unilateral Laplace
Transform of a signal x(t) is defined by
Lx(t)X (s) x(t)estdt
0
Properties of Unilateral Laplace transform
1 Linearity
Lx1(t)X 1(s)
Lx2(t)X2(s)
LAx1(t)Bx2(t)A X 1(s)BX2(s)
Proof:
LAx1(t)Bx2 (t) Ax (t) Bx
1 2 (t)est dt
0
A 1x (t)e st st
dt B x 2 (t)e dt
0 0
AX1(s)BX2 (s)
2 Shifting in timedomain
Lx(t)X (s)
L x(tt) e st0X(s)
0
Proof:
Lx(t t0 ) x(tt )e 0
st
dt
0
let t t0 p t t0 p and dt dp
x(p)es(t0p)dp
0
e st0x(p)e sp dpe st0X(s)
0
3 Shifting in frequencydomain
L x (t) X (s)
Le x(t)X (s a)
at
Proof:
L eatx(t) eatx(t)estdt x(t)e(sa)tdt X (s a)
0 0
4 Differentiation in time
Lx(t)X (s)
26
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
dx(t)
L sX (s) x(0 )
dt
Proof:
L
dx(t) dx(t)
e st dt
dt 0 dt
let u est and dv dx(t)
dusestdt and v x(t) udv uv vdu
L
dx(t) st
e x(t)
x(t)se d t
st
dt
0
0
x(0 )sx(t)estdtsX(s)x(0
0
Note:
d2x(t) dx(0 )
L s[sX (s) x(0 )]
2
dt dt
dx(0)
s X
2
(s) sx(0 )
dt
5. Integration in time
Lx(t)X (s)
0
X (s)
x()d
t
Lx()d s
s
proof :
t 0 t
x()dx()dx()d
0
0
0 x()d
Lx()d
s
t t
L x()d x()de st dt
0 00
t
let u x()ddu x(t)dt
0
st est
dv e dt v
s
t
t (est)
est
x()destdtx()d
s 0 0 s
x(t)dt
00 0
X (s)
0
s
27
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
0
t X (s) x()d
Lx()d
s s
6. Scaling in Time
Lx(t)X1(s) s
Lx(at) X
a a
Proof:
Lx(at)x(at)est dt
0
p dp
let at p t and dt
a a
s 1 s
Lx(at) x( p)e a dp
p
X
0 a a a
7. Differentiation in s-domain (frequencydifferentiation)
Lx(t)X (s)
dX (s)
Ltx(t)
ds
Proof:
X (s) x(t)estdt
0
de st
dX(s)
st
x(t) dt tx(t)e dt
ds 0
ds 0
dX (s)
Ltx(t)
ds
Note:
n
d n X (s)
L ( t) x(t)
dsn
8. Integration in s-domain (frequencyintegration)
Lx(t)X (s)
x(t)
t s
L X (s)ds
Proof:
28
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
X (s) x(t)est dt
0
1 stdt
X (s)ds x(t){e st ds}dt x(t)e
s 0 s 0t
x(t)
L
s
X (s)ds
t
9. initial and Final valuetheorem
Lx(t)X (s)
Initial Value theorem:
lt x(t) lt s X (s)
t0 s
dx(t)
Proof: L sX (s) x(0 )
dt
dx(t)
dt e st dt sX (s) x(0 )
0
dx(t)
e stdt 0 lt sX (s) x(0 )
lt
s
dt s
0
lt sX (s) x(0)
s
Final Value theorem:
lt x(t) lt s X (s)
t s0
dx(t)
e st dt sX (s) x(0 )
Proof:
dt
0
dx(t)
e st dt lt [sX (s) x(0 )]
lt
s0
dt s0
0
dx(t)
0
dt dt lt [sX (s) x(0 )]
s0
lt x(t) x(0) lt sX (s)x(0)
t s0
lt x(t) lt sX (s)
t s0
10. Convolution in Time
Lx1(t)X 1(s)
Lx2(t)X2(s)
Lx1(t)x2(t)X 1(s)X2(s)
By definition , x1(t)x2(t)
x ()x (t )d
1 2
Proof:
Lx1(t)x2 (t) x1 ()x2 (t )destdt
let t p t p &dt dp
29
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Lx1(t)x2 (t) x1 ()x2 ( p)des (p ) dp
x (1)esd ( p)espdp X 1 (s) X 2 (s)
x
2
11. Convolution in frequency(multiplication)
Lx(t)X (s)
Lg(t)G(s)
1
Lx(t)g(t) X (s) G(s)
2j
Note:
by definition
(i) X(s)G(s)
X(u)G(su)du
1
ds by definition
X (s)e
st
(ii) L 1 X (s) x(t)
2j
Proof:
1
Inverse Laplace Transform of X (s) G(s)is
2j
1 1
2j 2j
X(u)G(su)duestds
let s u p s u p and ds dp
1 1
X(u)G(p)due (up)tdp
1 1
2j 2j dp x(t)g(t)
X G(p)e
du pt
2j (u)eut
2j
2.4 Continuous Time Fourier Transform
X ( j) Fx(t) and x(t) F 1X ( j)
Xj for all
x(t)ejtdt
1
x(t) for all t
2 X(j)ejtd
Properties of CT Fourier
[Link]
F x1(t)X 1(j)
F x2(t)X2(j)
F Ax1(t)Bx2(t)A X 1(j)BX2(j)
Proof:
30
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
F Ax1(t)Bx2 (t) Ax (t)Bx
1 2
(t)e jtdt
A x1 (t)ejtdt B (t)ejt dt
x
2
AX1(j)BX2(j)
2. Shifting in timedomain
Fx(t)X ( j)
F x(tt)0ejt0X(j)
Proof:
Fx(t t0 ) x(t t )e 0
jt
dt
let t t0 p t t0 p and dt dp
x( p)ej(t0 p ) dp
e jt0x(p)e jp dpe jt0X(j)
3. Shifting in frequencydomain
Fx(t)X ( j)
F e j0tx(t) X(jj) 0
Proof:
F
e j0tx(t) e j0t
x(t)e jt
dt x(t)e (j 0j)t
dtX(jj) 0
4. Scaling in Time
Fx(t)X1( j)
j
F x(at) X
a a
Proof:
Fx(at)
x(at)ejt dt
p dp
let at p t and dt
a a
j
p dp
1 j
Fx(at) x( p)e a X
a a a
5. Time
reversalFx(t)X
( j) Fx(t)X
(j)
31
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Proof:
Fx(t) X (j)
x(t)ejtdt x(t)ejtdt x(t)e
(j)t
6. Differentiation in time
Fx(t)X ( j)
dx(t)
F jX ( j)
dt
Proof:
1
x(t) X(j)ejtd for all t
2
d
dx(t) 1 X(j) [e jt]d
dt 2 dt
dx(t) 1
jtddt
jX(j)e
2
dx(t)
F jX ( j)
dt
note:
dnx(t) n
F n (j) X ( j)
dt
7. Differentiation in frequency
Fx(t)X ( j)
dX ( j)
Ftx(t) j
d
Proof:
Xj
x(t) ejtdt
dX (j) d (ejt )
x(t) dt
d d
jt x(t)ejt dt
dX ( j)
Ftx(t) j
d
8. Duality
Fx(t)X ( j)
FX (t)2x (j)
Proof:
1
x(t) X(j)ejtd for all t
2
32
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2x(t) X ( j) e jt d
2x(t)X(j)ejtd FX ( j)
changing t toj
2x(j)X(t)e jtd FX (t)
9. Convolution in Time
F x1(t)X 1(j)
F x2(t)X2(j)
F x1(t)x2(t)X 1(j)X2(j)
By definition , x1(t)x2(t)
x ()x (t )d
1 2
Proof:
F x1(t)x2 (t) x1 ()x2 (t )dejt dt
let t p t p &dt dp
F x1(t)x2 (t) x1 ()x2 ( p)dej(p ) dp
x (1 )e (p)ejp dpX
d x ( j)X 2 ( j)
j
2 1
10. Convolution in frequency(multiplication)
Fx(t)X ( j)
Fg(t)G( j)
1
F x(t)g(t) X ( j) G( j)
2
Note:
by definition
(i) X(j)G(j)
X(u)G(ju)du
1
X (j) x(t)
X ( j)e jt dby definition
(ii) F 1
2
Proof:
1
Inverse Fourier Transform of X ( j) G( j)is
2
1 1 X(u)G(ju)duejtd
22
let s u p s u p and ds dp
33
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1 1
2j
X(u)G(p)due (up)tdp
1 1
2j dp x(t)g(t)
X (u)eut du 2j G(p)e
pt
2j
Parseval’s Theorem
Fx(t)X ( j)
2dt 1
E x(t)
2
X ( j d
2
Proof:
x(t) 2dt x(t)x(t)dt
1 1
x(t) X(j)ejtd x (t) X ( j) ejtd
2 2
x(t)x (t)dt x(t) 21 X ( j)e jtddt
1 X(j)
x(t)ejtdtd
2
1 X(j)
x(t)ejtdtd
2
1 1
X ( j) X ( j)d
2
2
2
X ( j) d
34
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Continuous Time Fourier Transform:
Viewed periodic functions in terms of frequency components ((Fourier series) as well as
ordinary functions of time
Viewed LTI systems in terms of what they do to frequency components (frequency
response)
Viewed LTI systems in terms of what they do to time--domain signals (convolution with
impulse response)
► View aperiodic functions in terms of frequency components
viaaFouriertransform
The Fourier expansion coefficient ( in OWN) of a periodic signal is
is
and the Fourier expansion of the signal is:
which can also be written as:
where is defined as
When theperiodof approaches infinity , the periodic signal
becomes a non-periodic signal and the following willresult:
Interval between two neighboring frequency components becomes zero:
35
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Discrete frequency becomes continuous frequency:
Summation of the Fourier expansion in equation (a) becomes an integral:
the second equal sign is due to the general fact:
Time integral over in equation (b) becomes over the entire time axis:
In summary, when the signalisnon-periodic , the Fourier
expansion becomes Fourier transform. The forward transform (analysis)is:
and the inverse transform (synthesis) is:
Notethat isdenoted by in OWN.
Comparing Fourier coefficient of a periodic signal with with Fourier spectrum of a
non-periodicsignal
36
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
we see that thedimension of is different fromthatof :
If represents the energy contained in the kth frequency component of a periodic
signal , then represents the energy density of a non-periodic signal
distributed along the frequency axis. We can only speak of the energy contained ina
particular frequencyband :
2.5 InverseTransforms
If we have the full sequence of Fourier coefficients for a periodic signal, we can reconstruct it by
multiplying the complex sinusoids of frequency ω0k by the weights Xk and summing:
We can perform a similar reconstruction for aperiodic signals
These are called the inverse transforms.
37
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2.6 PROBLEMS:
Example0:
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:
38
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
The spectrum is
This is the sinc function withaparameter , as shown in thefigure.
Note that the height of the main peak is and it gets taller andnarroweras gets
larger. Also note
When approachesinfinity, forall , and the spectrumbecomes
Recall that the Fourier coefficient of is
which represents the energy contained in thesignal at (DC component atzero
frequency), andthespectrum is the energy density or distribution which is infinity at
zerofrequency.
The integral in the above transform is an important formula to be used frequently later:
which can also be written as
39
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Switching and in the equation above, we alsohave
representing a superposition of an infinite number of cosine functions of all frequencies,
which cancel each other any where along the time axisexcept 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 containedat frequency .
Continuous Time Fourier Transform-PROBLEMS
X ( j) Fx(t) and x(t) F 1X ( j)
Xj for all
x(t)ejtdt
1
x(t) for all t
2 X(j)ejtd
40
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1. Find the inverse Fourier transformof (i) () (ii) (0 )
1
(i) x(t) X ( j) e jt d
2
1 1
() ejtd
2 2
()1 for 0
Since,
0 for 0
1
F 1()
2
F 12() 1
F1 2()
1
(ii) x(t) X ( j) e jt d
2
1 (0)ejtd
e j0t
2 2
Since, ( 0 )1 for 0
0 for 0
e j0t
F1 (0 )
2
F1 2()e
0
j 0t
F ej t 2(0 )
0
Fourier transform of standard signals:
a) x(t) cost0
e j0t ej0t
cost0
2
F e j
0t
F e j0t 1
X j
2 2 [2(0)2(0)]
[(0 ) (0 )] [(0 ) (0 )]
b) x(t) sin t0
41
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
e j0t
j0t
e
sin t
0
F ej0t2jF e j0t 1
X j [2(0 ) 2(0 )]
2j2j
j [(0 ) (0 )] j[(0 ) (0 )]
c) x(t) sgnt
The given function x(t) sgnt is known as signum function and is defined as,
sgn(t) 1if t 0
0 if t 0
1if t 0
The function is not absolutely integrable. The Fourier transform of x(t) sgntis
obtained by considering thefunction lt e sgnt
a|t|
a0
Xj lt e a|t| sgn(t)e jt dt
a0
0
lta0 eat e jt dt eat e jt dt
0
1
lt 1 lt 2 j
a j
a j a0 a2 2
a0
2j 2
j
d) x(t) ut
u(t) 1 for t 0
0 for t 0
42
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1 1
This signal can be expressedas u t sgn(t)
2 2
F1 2()
2
Fsgn(t)
j
Fu(t)
1
21 2
2 2 j
1
Fu(t)
j
e) Rectangular pulse
t
Rectangular pulse ofwidth extending from to and amplitude1.
2 2
t
x(t)
x(t) 1 to
2 2
0otherwise
2
Xjx(t)ejtdte jtdt
2
1 j
2 sin
e
jt 2 j
e 2
e 2 2
j j
2
43
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2 sin
sin 2
2
sinc
2
2
sin x
Note: sin c(x)
x
t
f)Triangularpulse
Triangular pulse of width extending from to
2
2
and amplitude 1.
t
x(t)
2t
x(t) 1 0 t
2
1
2t
t 0 2
Xjx(t)ejtdt
0 2
2t 2t
jt
(1 )e dt
jtdt
(1
0
)e
2
e jt 0 2 t
e
0
jt e jt e
jt 2
2 t
e e
jt jt 2
2 j ( j)2
j j ( j) 0 j
2 2
0
44
II YEAR/THIRD SEMESTER
EC3354-SIGNALS AND SYSTEM
j j j 2
j
1 e 2 2 1 e 2 e e
2 1
j
j (j)2 2 (j) (j)2 j j
j j
2 e 2 e 2 1
2 j
( j)2 ( j)2
2 j
j
2 e 2
e 2
(j)2 (j)2 ( j)2
2 j
j j
j 2
2 e 2 e
2 2 e 4 e 4
(j)2 ( j)2 ( j)2 j
2
2
j j 2 sin
2 e 4 e 4 8 4
j 16
4
sinc2
2 4
45
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2.5 LaplaceTransform
► Lapalcetransform isageneralization oftheFourier transform in the sensethat it allows ―complex
frequency‖ whereasFourieranalysis can on lyhandle ―real frequency‖.LikeFourier transform,Lapalce
transform allows us to analyzea―linearcircuit‖problem, no matterhow complicated the circuit is, in the
frequency domain in stead of in he timedomain.
►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.
46
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
47
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
48
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
49
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
ROC for Laplace Transform
Laplace transform of x(t) is X (s) x(t)est dt
The range of values of ‗s‘ for which the integral in the equation converges is referred to as the region
of convergence(ROC).
2.8 Properties of ROC for LaplaceTransform
1. ROC of X(s) consists of strips parallel to the jaxis in the s-plane.
2. For rational Laplace Transform, the ROC does not contain anypoles.
N(s)
X (s) general formof rational LT
D(s)
Roots of D(s) 0 are poles
3. If x(t) is of finite duration and if there is at least one value of s for which the LT converges, then the
ROC is the entires-plane.
4. If x(t) is right sided and if the line Res0is in the ROC , then all values of s forwhich
Res0will also be in the ROC.
5. If x(t) is left sided and if the line Res0 is in the ROC , then all values of s for which
Res0will also be in the ROC.
6. If x(t) is two sided and if the line Res0 is in the ROC , then the ROC will consist of a strip in
the s-plane which includes the line Res0.
50
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
UNIT III
LINEAR TIME INVARIANT –CONTINUOUS TIME SYSTEMS
3.1 System:
A system is an operation that transforms input signal x into output signal y.
3.2 LTISystems
• 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 physicalsystems
• They can be modeledsystematically
– Non-LTI systems typically have no general mathema tical procedure to obtainsolution
Differential equation:
• This is a linear first order differential equatio n with constant coefficients (assuming a and bare
constants)
The general nth order linear DE with constant equations is
Linear constant-coefficient differential equations In RC circuit
51
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
- To introduce some of the important ideas concerning systems specified by linearconstant-
coefficient differential equations ,let us consider a first-order differentialequations:
3.3 Block diagramrepresentations
Block diagram representations of first-order systems described by differential and
difference equations
3.4 ImpulseResponse
This impulse response signal can be used to infer properties about the system‘s structure (LHS of
difference equation or unforced solution). The system impulse response, h(t) completely characterises a
linear, time invariant system
52
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3.5 Properties of System ImpulseResponse
Stable
A system is stable if the impulse response is absolutely summable
Causal
A system is causal if h(t)=0 when t<0
Finite/infinite impulse response
The system has a finite impulse response and hence no dynamics in y(t) if there exists T>0, such
that: h(t)=0 when t>T
3.6 ConvolutionIntegral
• An approach (available tool or operation) to desc ribe the input-output relationship for LTISystems
• In a LTIsystem
– d(t) h(t)
– Remember h(t) isT[d(t)]
– Unitimpulsefunction the impulseresponse
• It is possible to use h(t) to solve for any input -outputrelationship
• Any input can be expressed using the unit impulsefunction
3.6.1 Convolution Integral –Properties
53
IIYEAR/THIRD SEMESTER EC6303-SIGNALS AND SYSTEM
54
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3.7 BLOCK DIAGRAM REPRESENTATION-STRUCTUREREALIZATION
IIR Systems are represented in four different
ways Direct Form Structures Form I and Form II
Cascade Form Structure
Parallel Form Structure
Lattice and Lattice-Ladder structure.
DIRECT FORM STRUCTURE FOR IIR SYSTEMS
IIR systems can be described by a generalized equations as
N M
y(n)=-∑ ak y(n–k)+∑ bk x(n–k)
k=1 k=0
Z transform is given as
M N
H(z) = ∑ bk z–k / 1+ ∑ akz –k
K=0 k=1
M N
Here H1(z) = ∑ bk z–k And H2(z) = 1+ ∑ akz–k
K=0 k=0
Overall IIR system can be realized as cascade of two function H1(z) and H2(z). Here H1(z) represents zeros
of H(z) and H2(z) represents all poles of H(z).
1. Direct form I realization of H(z) can be obtained by cascading the realization of H1(z)
which is all zero system first and then H2(z) which is all pole system.
· There are M+N-1 unit delay blocks. One unit delay block requires one memory location.
Hence direct form structure requires M+N-1 memorylocations.
3. Direct Form I realization requires M+N+1 number of multiplications and M+N number
of additions and M+N+1 number of memory locations.
55
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
DIRECT FORM -I
DIRECT FORM - II
1. Direct form realization of H(z) can be obtained by cascading the realization of H1(z)
which is all pole system and H2(z) which is all zero system.
· Two delay elements of all pole and all zero system can be merged into single delayelement.
· Direct Form II structure has reduced memory requirement compared to Direct form I
structure. Hence it is called canonicform.
· The direct form II requires same number of multiplications(M+N+1) andadditions
(M+N) as that of direct formI.
56
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
CASCADE FORM STRUCTURE FOR IIR SYSTEMS
In cascade form, stages are cascaded (connected) in series. The output of one system is input to
another. Thus total K number of stages are cascaded. The total system function
'H' is given by
H= H1(z).H2(z)… .............................Hk(z) (1)
H=Y1(z)/X1(z).Y2(z)/X2(z). ......................Yk(z)/Xk(z) (2)
k
H(z)=π Hk(z) (3)
k=1
57
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
x(n)=x1(n) y1(n)=x2(n) y2(n)=x3(n) yk(n)=y(n)
H1(z) H2(z) Hk(z)
FIG - CASCADE FORM REALIZATION OF IIR SYSTEM
Each H1(z), H2(z)… etc is a second order section and it is realized by direct form 2.
PARALLEL FORM STRUCTURE FOR IIR SYSTEMS
58
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3.8 PROBLEMS
1. Determine the output response of RC Low pass network shown in figure due toinput
t
x(t) te RC
by convolution.
R
X(t) C Y(t)
For the circuit shown in figure,
1 (1)
x(t) Ri(t) i(t)dt
C
1 (2)
C
y(t) i(t)dt
Taking LT of the equations and rearranging,
I(s)
X (s) RI (s) (1)
sC
I (s)
Y (s) (2)
sC
X(s)sRCY(s)Y(s)
Y (s)
H (s)
X (s)
1
1
H (s) RC
(sRC 1) (s 1 )
RC
t
x(t) te RC
1
X(s)
1 2
(s )
RC
59
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
y(t) x(t)h(t)
1
Y(s)X(s)H(s) RC
1 3
(s )
RC
t
1 2
y(t)L1Y(s)
RC
te
2RC
2. The Input and Output of a causal LTI system are related by the differentialequation
d 2y(t) dy(t)
6 8y(t)2x(t). Using Fourier Transform
dt2 dt
(i) Find the Impulse response of thesystem
(ii) Find the response of the systemif x(t) e3tu(t)
SOLUTION:
(i) Impulse response of thesystem:
d 2y(t) dy(t)
6 8y(t)2x(t) ---------------(1)
dt 2 dt
Applying Fourier Transform to the given Differential Eq.
j2Yj j6Yj 8Yj2X jYj
(j26j8)2X j
2X j
Yj ----------------------(2)
( j2 6 j8)
Here x(t) tand X ( j) 1
using partial fraction expansion
A B
Y( j)
j 2 j 4
Set j = -2 then A(j+2)+B(j+4)2, gives B 1
Set j = -4 then A( j+2)+B(j+4)2, gives A 1
1 1
Y( j)
j 2 j 4
By taking inverse FT , y(t) e4tu(t) e2tu(t)
1
(ii) If x(t) e3tu(t) then X ( j)
j 3
2
Yj(j2j68)
j 3
2
Yj
(j3)(j26j8)
using partial fraction expansion
A B C
Y( j)
j 2 j 4 j 3
Solving for A,B & C, we get A=1 , B=1 and C=-2
60
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Then by taking Inverse FT, weget
y(t) e2tu(t) e4tu(t) 2e3tu(t)
3. Realize the system described by the differential equation
d3y(t) d2y(t) dy(t) d2x(t) dx(t)
4 7 8y(t)5 4 7x(t) in Direct FormII.
dt 3 dt 2 dt dt 2 dt
SOLUTION:
Apply Laplace transform for the given differential equation with assuming zero initial conditions.
s3YS 4s2YS 7sYS 8YS5s2XS 4sXS 7XS(1)
Y(s) 5s24s 7
H(s)
X(s) s34s27s 8
5s24s7 Y (s) W(s)
s34s27s8 X(s) W(s)
W(s) 1
let (1)
X(s) s34s27s 8
Y (s)
let 5s2 4s 7 (2)
W (s)
from (1)
(s34s27s8)W(s)X(s) (3)
from (2)
Y(s)(5s24s7)W(s) (4)
from (3)
s3W (s) 4s2W (s) 7sW (s) 8W (s) X (s)
s3W(s)X(s)4s2W(s)7sW(s)8W(s) (5)
from (4)
Y(s)5s2W(s)4sW(s)7W(s) (6)
Equations (5) and (6) are used for implementing the system.
61
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4. Find the convolution of the followingsignals.
x (t) eatu(t) ; x (t) ebtu(t)
1 2
SOLUTION:
L[x1(t) x2(t)]=X1(s)X2(s)
1
x1(t)=e-atu(t) ; X1(s)=L[e-atu(t)]=
s a
1
x1(t)=e-btu(t) ;X2(s)=L[e-btu(t)]=
s b
Y(s) = X (s)X (s) A B
1 2 sa s b
s a 1
A= | s a ,then A
(s a)(s b) (b a)
s b 1
A= | s b ,then B
(s a)(s b) (a b)
1 1
(b a) (b a)
Y(s) =
s a s b
y(t) = e
1 at
ebtu(t)
b-a
[Link] the convolution of the followingsignals:
x(t) e2tu(t)
h(t) u(t 2).
SOLUTION:
L[x1(t) x2(t)]=X1(s)X2(s)
1
x1(t)=e-2tu(t) and X1(s)=L[e-2tu(t)] =
s2
e2S , 1
x2(t)=L[u(t+2)]= [Using time shiftingproperty,L[u(t)]= ]
s s
e2S
L[x1(t) x2(t)]=
ss 2
e2S A B
Y(s) =
ss 2 s s 2
1 1
A s | s0 ; A
s(s 2) 2
1 1
B(s2) | s 2 ; B
s(s 2) 2
62
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1/ 2 1/ 2
Y1 (s) =
s (s 2)
1 1
Y(s) = e2S( )
2s 2(s 2)
e2S e2S
Y(s) =
2s 2(s 2)
1 1
y(t) = u(t 2)- e-2( t+2 ) u(t+2)
2 2
6.i) Using Laplace transform , find the impulse response of an LTI system
dyt
2yt xt
d2
yt
described by thedifferentialequation
dt dt
SOLUTION:
Apply Laplace transform for the given differential equation with assuming zero initial conditions .
s2YS sYS 2YSXSYS
s2s2
YS
XS
HS 2
1
X S s s 2
1
(s1)(s2)
A B
S 2 (S 1)
1 1 1 1
3 (S 2) 3 S 1
1 1
ht e2t u(t) etu(t)
3 3
ii) Explain the properties of Convolutionintegral.
(1) Commutativeproperty
x1(t) x2 (t) x2 (t) x1(t)
(2) Associative property
x1 (t)x2(t)x3(t)x1 (t) x2(t)x3(t)
(3) Distributiveproperty
x1 (t) x2(t)x3(t) x1 (t)x2(t)x1 (t)x3(t)
(4) shift property
ifx1(t) x2 (t) z(t) then
x1(t)x2(tT)z(tT)
(5) Convolution with animpulse
x(t) (t) x(t)
(6) Convolution with unitstep
63
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
t
x(t)u(t)
x()d
t t0
And x(t)u(tt0)
x()d
(7) Width property
Let the duration of x1(t) beT1
Let the durationof x2 (t) beT2
Then the duration of the signal obtained by convolving x1(t) and x2 (t) is T1 T2
8.i) Solve the differential equation:
dyt
yt 4 dyt 5yt 5xt with y(0) 1and
d2
| 2 andxt ut
dt dt dt 0
SOLUTION:
To solve the given differential equation we can use LT method.
Applying LT to the given differential Eq. With given initial conditions.
2
dy(0 )
sYS sy(0 ) 4 sYs y(0 ) 5Y s 5X s
dt
s2Y S s 2 4sYs1 5Y s 5X s
5
s2Y S 4sY (s) 5Y s s 6
s
5
(s2 4s 5)Y (s) s 6
s
5 s 6
Y (s) ----------(1)
s(s2 4s 5) (s2 4s 5) (s2 4s 5)
Applying Partial fraction to the first term in Y(s)
5 A Bs C
s(s 4s 5) s (s 4s 5)
2 2
5 A(s2 4s 5) (Bs C)s
s(s24s5) s(s2 4s 5)
5 A((s 2)2 1) (Bs C)s
s(s24s5) s(s2 4s 5)
By solving this Eq. We can get the values of A, B and C
A=1, B=-1 and C=-4
By substituting the values in Eq. (1) and rearranging the terms we have
1 s 4 6 s
Y (s)
s ((s 2) 1) ((s 2) 1)
2 2
((s 2) 1) ((s 2)2 1)
2
1
2
s ((s 2)21)
By taking inverse LT we can get y(t)
y(t) u(t) 2e2tsint
64
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
dx(t)
d2
dyt 3yt
2xt .
ii) The system is described by the input output relation y t dt dt
dt
Find the system transfer function, frequency response and impulse response.
System Transfer function:
Applying Laplace Transform to the given differential equation with zero initial conditions,
s2Y S sYS 3Y S sX s 2X s
(s 2)
Y s Xs
(s2s 3)
Y s (s2)
H(s)systemtranferfunction
Xs (s2s 3)
Frequence response of the System :
Applying FOURIER Transform to the given differential equation.
j2Yj jYj 3YjjXj 2X jYj(j
2j3)(j2)Xj
( j2)
Yj X j
( j2j 3)
Yj ( j2)
H(j) Frequencyresponse
Xj ( j2j3)
Impulse response of the system:
s2Y S sYS 3Y S sX s 2X s
(s 2)
Y s Xs
(s2s 3)
Y s (s2)
H(s)
Xs (s2s 3)
Impulse response of the systemh(t)
h(t) inverse LT of H(s),
Y s (s2)
H(s)
Xs (s2s 3)
9. Draw the direct form I and II implementations of the system describedby
dyt
5 yt 3xt
dt
Applying LT to the given differential Eq.
sY(s) 5Y s 3X (s)
(s 5)Y s3X (s)
Y s 3
H(s)
X(s) (s 5)
Direct form I
65
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3
Y s 3 3 5s
X (s) (s 5) s 1 1 5
s s
5 3
Y(s) Y(s) X(s) (1)
s s
3
let X (s) W (s) (2)
s
from (1) and (2)
5
Y(s)W(s) Y(s)(3)
s
Using (2) and (3) the system is implemented asfollows:
Direct form II
Ys 3 Ys Ws
H (s)
X(s) (s5) X(s) W(s)
W s 1
let, (1)
X(s) (s 5)
Y s
let, 3 (2)
W (s)
from (1)
sW (s) 5W (s) X (s)
sW(s)X(s)5W(s)(3)
from (2)
Y (s)3W(s) (4)
Using (3) and (4) the system is implemented as follows:
66
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
10. The input and the output of a causal CTI system are related by thedifferential
d
d 2yt
equation 6 yt 8y(t) 2x t .Find impulse response of the
dt dt
system. 2
SOLUTION:
Apply Laplace transform for the given differential equation with assuming zero initial conditions .
s2YS 6sYS 8YS2X S
Y S 2
HS 2
XS s 6s8
x(t) (t) and X (s) 1
2
HS 2
s 6s 8 X (s)
A B 2
H (s)
(s2) (s4) (s 2)(s 4)
2
A(s2) | s2 A 1
(s 4)(s 2)
2
B(s4) | s 4 B 1
(s 4)(s 2)
1 1
H (s)
s2 s 4
Taking inverse Laplace transform to get impulse response
h(t) e2tu(t) e4t u(t)
3.9 Convolution integralexamples
To find the output of the system with impulse response
to the input
we will use the convolution integral
Because the input function has three distinct regions t<0, 0<t<1 and 1<t, we will need to split up the
integral into three parts.
Section 1: t<0
67
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
For t<0 the argument of the impulse function (t-λ) is always negative. Since h(t-λ)=0 for (t-λ)<0,
the result of the integral is zero for t<0.
This situation is depicted graphically below (t=-0.2):
Section 1: t<0
For t<0 the argument of the impulse function (t-λ) is always negative. Since h(t-λ)=0 for (t-λ)<0,
the result of the integral is zero for t<0.
This situation is depicted graphically below (t=-0.2):
The result for the first part of our solution is the integral of the yellow line (which is always zero),
Section 2: 0<t<1
For 0<t<1 we need to evaluate the integral only from λ=0 to λ=t, since f(λ)=0 when λ<0, and h(t-
λ)=0 when (t-λ)<0 (or, equivalently t<λ). So the integral becomes, in effect:
This situation is depicted graphically below (t=0.5):
68
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
We can now evaluate the integral of the yellow line:
Thus, the result for the second part of the solution is
Section 3: 1<t
For 1<t we need to evaluate the integral only from λ=0 to λ=1, since f(λ)=0 when λ<0 and when
λ>1. So the integral becomes, in effect:
This situation is depicted graphically below (t=1.2):
We can now evaluate the integral:
69
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Thus, the result for the third part of the solution is:
The Complete Answer
We can get the results for all time by combining the solutions from the three parts.
This result is shown below. Click on image to go to an animation of the procedure.
70
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Continuous-time convolution
Here is a convolution integral example employing semi-infinite extent signals. Consider the convolution
of x(t) = u(t) (a unit step function) and
(a real exponential decay starting from t = 0). The figure provides a plot of the waveforms.
The output support interval is
You need two cases (steps) to form the analytical solution valid over the entire time axis.
Case 1: Using Figure b, you can clearly see that for t < 0, it follows thaty(t) =0.
Case 2: Again looking at Figure b, you see that for t ≥ 0, some overlap always occurs betweenthe
two signals of the integrand. The convolution integral outputis
Putting the two pieces together, the analytical solution for y(t)
71
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
UNIT IV
ANALYSIS OF DISCRETE TIME SIGNALS
4.1 Samplingtheory
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 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 imageplane
► Filter: blends multiple samplestogether
► Samplingtheory
Sampling Theorem: bandlimited signal can be reconstructed exactly if it is sampled at a rate
atleast twice the maximum frequencycomponent in it."
72
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Consider a signal g(t) that is bandlimited.
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
A continuous time signal can be processed by processing its samples through a discrete time
system. For reconstructing the continuous time signal from its discrete time samples without any error,
the signal should be sampled at a sufficient rate that is determined by the sampling theorem.
4.2 Aliasing
Aliasing is a phenomenon where the high frequency components of the sampled signal interfere
with each other because of inadequate sampling ωs < 2ωm. Aliasing
Aliasing leads to distortion in recovered signal. This is the reason why sampling frequency should be
atleast twice the bandwidth of the signal.
73
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
.SAMPLING THEOREM
It is the process of converting continuous time signal into a discrete time signal by taking samples of the
continuous time signal at discrete time instants.
X[n]= Xa(t) where t= nTs=n/Fs ….(1)
When sampling at a rate of fs samples/sec, if k is any positive or negative integer, we cannot distinguish
between the samples values of fa Hz and a sine wave of (fa+ kfs) Hz. Thus (fa + kfs) wave is alias or
image of fa wave.
Thus Sampling Theorem states that if the highest frequency in an analog signal is Fmax and the
signal is sampled at the rate fs > 2Fmax then x(t) can be exactly recovered from its sample values. This
sampling rate is called Nyquist rate of sampling. The imaging or aliasing starts after Fs/2 hence folding
frequency is fs/2. If the frequency is less than or equal to 1/2 it will be represented properly.
Example:
Case1: X1(t) = cos 2∏(10) t Fs=40 Hz i.e t= n/Fs
x1[n]= cos 2∏(n/4)= cos(∏/2)n
Case2: X1(t) = cos 2∏(50) t Fs=40 Hz i.e t=
n/Fs x1[n]= cos 2∏(5n/4)= cos 2∏( 1+¼)n
= cos
(∏/2)n
Thus the frequency 50 Hz, 90 Hz , 130 Hz … are alias of the frequency 10 Hz at the sampling rate of 40
samples/sec
.QUANTIZATION
The process of converting a discrete time continuous amplitude signal into a digital signal by
expressing each sample value as a finite number of digits is called quantization. The error
introducedin
representing the continuous values signal by a finite set of discrete value levels is called
quantization error or quantization noise.
Example: x[n] =5(0.9)nu(n) where 0 <n<∞ & fs= 1Hz
N [n] Xq [n] Rounding Xq [n] Truncating eq [n]
0 5 5.0 5.0 0
1 4.5 4.5 4.5 0
2 4.05 4.0 4.0 -0.05
3 3.645 3.6 3.6 -0.045
4 3.2805 3.2 3.3 0.0195
Quantization Step/Resolution : The difference between the two quantization levels is called
quantization step. It is given by Δ = XMax – xMin / L-1 where L indicates Number of quantizationlevels.
74
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
.CODING/ENCODING
Each quantization level is assigned a unique binary code. In the encoding operation, the
quantization
sample value is converted to the binary equivalent of that quantization level. If 16 quantization levels
are present, 4 bits are required. Thus bits required in the coder is the smallest integer greater than or
equal toLog2L. i.e b= Log2L
Thus Sampling frequency is calculated as fs=Bit rate /
b.
.ANTI-ALIASING FILTER
When processing the analog signal using DSP system, it is sampled at some rate depending upon
the
bandwidth. For example if speech signal is to be processed the frequencies upon 3khz can be
used. Hence the sampling rate of 6khz can be used. But the speech signal also contains some frequency
components more than 3khz. Hence a sampling rate of 6khz will introduce aliasing. Hence signal
should be band limited to avoidaliasing.
The signal can be band limited by passing it through a filter (LPF) which blocks or attenuates
all the frequency components outside the specific bandwidth. Hence called as Anti aliasing filter or
pre- filter. (BlockDiagram).
.SAMPLE-AND-HOLD CIRCUIT:
The sampling of an analogue continuous-time signal is normally implemented using a device called
an
analogue-to-digitalconverter(A/D).Thecontinuous-timesignalisfirstpassedthroughadevicecalled a
sample-and-hold (S/H) whose function is to measure the input signal value at the clock instant and
hold
itfixedforatimeintervallongenoughfortheA/[Link]-to-digitalconversion
ispotentiallyaslowoperation,andavariationoftheinputvoltageduringtheconversionmaydisruptthe
[Link]/Hpreventssuchdisruptionbykeepingtheinputvoltageconstantduring the
conversion. This is schematically illustrated byFigure.
After a continuous-time signal has been through the A/D converter, the quantized output may differ from
the input value. Themaximum possible output value after the quantization process could be up to half the
quantization level q above or q below the ideal output value. This deviation from the ideal output value is
called the quantization error. In order to reduce this effect, we increases the number ofbits.
75
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.3 Sampling of Non-bandlimited Signal: Anti-aliasingFilter
Anti aliasing filter is a filter which is used before a signal sampler, to restrict the
bandwidth of a signal to approximately satisfy the sampling theorem.
The potential defectors are all the frequency components beyond fs/2 Hz.
We should have to eliminate these components from x(t) before sampling x(t).
As a result of this we lose only the components beyond the folding frequency fs/2 Hz.
These frequency components cannot reappear to corrupt the components with frequencies below the
folding frequency.
This suppression of higher frequencies can be accomplished by an ideal filter of bandwidth fs/2 Hz. This
filter is called the anti-aliasing filter.
The anti aliasing operation must be performed before the signal is sampled. The anti aliasing filter,
being an ideal filter isunrealizable.
In practice, we use a steep cutoff filter, which leaves a sharply attenuated residual spectrum beyond the
folding frequency fs/2.
76
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.4 DISCRETE TIME FOURIERTRANSFORM
In mathematics, the discrete-time Fourier transform (DTFT) is one of the specific
forms of Fourier analysis. As such, it transforms one function into another, which is called the frequency
domain representation, or simply the "DTFT", of the original function (which is often a function in the
time-domain). But the DTFT requires an input function that is discrete. Such inputs are often created by
sampling a continuous function, like a person's voice.
Given a discrete set of real or complex numbers: (integers),the
discrete-time Fourier transform (or DTFT) of is usually written:
77
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.5 Inversetransform
The following inverse transforms recover the discrete-time sequence:
The integrals span one full period of the DTFT, which means that the x[n] samples are also the
coefficients of a Fourier series expansion of the DTFT.
Infinite limits of integration change the transform into a continuous-time Fourier transform
[inverse], which produces a sequence of Dirac impulses. That is:
4.6 Properties
78
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.7 SYMMETRYPROPERTIES
The Fourier Transform can be decomposed into a real and imaginary part or into an even and odd
part.
79
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.8 Z-transforms
► Definition: The Z – transform of a discrete-time signal x(n) is defined as the powerseries:
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-
timesystems
Bilateral forward Z transform
80
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
ADVANTAGES OF Z TRANSFORM
1. The DFT can be determined by evaluating ztransform.
2. Z transform is widely used for analysis and synthesis of digitalfilter.
3. Z transform is used for linear filtering. z transform is also used for finding Linear convolution,
cross-correlation and auto-correlations of sequences.
4. In z transform user can characterize LTI system(stable/unstable,causal/anti- causal) and its
response to various signals by placements of pole and zeroplot.
ADVANTAGES OF ROC(REGION OF CONVERGENCE)
1. ROC is going to decide whether system is stable orunstable.
2. ROC decides the type of sequences causal oranti-causal.
3. ROC also decides finite or infinite durationsequences.
Z TRANSFORM PLOT
Imaginary Part of z
Im (z)
Z-Plane
|z|>a
|z|<a
Re (z) Real part of z
Fig show the plot of z transforms. The z transform has real and imaginary parts. Thus a plot of
imaginarypartversusrealpartiscalledcomplexz-plane.Theradiusofcircleis1calledasunitcircle.
This complex z plane is used to show ROC, poles and zeros. Complex variable z is also expressed
in polar form as Z= rejωwhere r is radius of circle is given by |z| and ω is the frequency of the
sequence in radians and given by∟z.
81
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
[Link] Time Domain Property z Transform ROC
Sequence
1 δ(n) (Unit sample) 1 complete z plane
2 δ(n-k) Time shifting z-k except z=0
3 δ(n+k) Time shifting zk exceptz=∞
4 u(n) (Unit step) 1/1- z-1 = z/z-1 |z| > 1
5 u(-n) Time reversal 1/1- z |z| < 1
6 -u(-n-1) Time reversal z/z- 1 |z| < 1
7 n u(n) (Unit ramp) Differentiation z-1 / (1- z-1)2 |z| > 1
8 an u(n) Scaling 1/1- (az-1) |z| > |a|
9 -an u(-n-1)(Left side 1/1- (az-1) |z| < |a|
exponentialsequence)
10 n an u(n) Differentiation a z-1 / (1- az-1)2 |z| > |a|
11 -n an u(-n-1) Differentiation a z-1 / (1- az-1)2 |z| < |a|
12 an for 0 < n < N-1 1- (a z-1)N/ 1- az-1 |az-1| <∞
except z=0
13 1 for 0<n<N-1 or Linearity 1- z-N/ 1- z-1 |z| > 1
u(n) – u(n-N) Shifting
14 cos(ω0n) u(n) 1- z-1cosω0 |z| > 1
1- 2z-1cosω0+z-2
15 sin(ω0n) u(n) z-1sinω0 |z| > 1
1- 2z-1cosω0+z-2
16 an cos(ω0n) u(n) Time scaling 1- (z/a)-1cosω0 |z| > |a|
1- 2(z/a)-1cosω0+(z/a)-2
17 an sin(ω0n) u(n) Time scaling (z/a)-1sinω0 |z| > |a|
1- 2(z/a)-1cosω0+(z/a)-2
4.9-Properties Of Z transform
Z- Transform of a signal x(n) is defined by
X (z) x(n)z
nn
Properties of z- transform
1. Linearity
x(n)
1 z X 1 (z)
z X2(z)
x 2(n)
82
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2 AX(z)B
Ax(n)Bx(n) 1 X(z)
z
1 1
Proof:
z- transform of Ax1 (n) Bx2 (n) Ax (n)Bx 1 2
(n)z n
n
Ax (n)
1 z
n
Bx 2 (n)z nA x (n)
1 z
nB
x 2 (n)z n
n n n n
AX1 (z) BX1 (z)
2. Shifting in timedomain
x(n) z X(z)
x(nk) z zkX(z)
Proof:
z- transformofx(nk)x(nk)zn
n
let n k m
n m k
x(nk)znx(m)z(mk)zkx(m)z(m)zkX (z)
n m m
[Link] reversal
x(n)z X(z)
x(n) z X(z1)
Proof:
z- transform of x(n)x(n)zn
n
let n m
n m
x(n)z nx(m)z (m)x(m)(z 1)mX (z 1
n m m
[Link] in z-domain
(multiplication by an exponential sequence)
x(n)z X(z)
z
a nx(n)z X a
Proof:
a x(n)z
z- transform of a n x(n) n n
n
z
x(n)(a
z) X (a z) X a
1 n 1
n
83
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
5. Differentiation in z-domain
x(n) z X(z)
dX(z)
nx(n) z z
dz
Proof:
z- transform of x(n) X (z) x(n) z
n
n
differentiating the z-transform, we have,
dX (z)
x(n)(n)zn1
dz n
z 1
(nx(n))z n
n
dX (z)
z (nx(n))zn
dz n
[Link]
x(n)
1 z X(z)1
z
x(n)
2 X(z) 2
x(n)*x(n)
1 2 z X(z)X(z)
1 2
Proof:
By definition,
x1(n)*x2(n)x1(k)x 2 (nk)
k
Z – transform of x1 (n) * x(n)
2 [x (n) * x(n)]z
1 2
n
n
nk
x1(k )x2 (n k)z n
let n k p
n p k
1 x (k)x 2( p)z ( pk )
x2 ( p)z x (k)z
p k
pk
1
X 2 (z) X1 (z)
p k
[Link]
x(n)
1 z X 1 (z)
x(n)
2 z X(z)2
xx
z X(z)X(z
12 1 2
1 )
Proof:
By definition,
84
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Correlation between two signals x1(n)&x2(n) is given by
x x x (n)
12 1
* x(n)
2
Z – transform of x1(n)*x2 (n) [x1(n)*x2 (n)]z n
n
X (z)
1 X (z
2
1)
Using convolution and time reversal property of z- transform
Note:
Convolution property:
2 X(z)X(z)
x(n)*x(n) z
1 1 2
Time reversal property:
x(n)z X(z1 )
[Link]
x(n) z X(z)
x (n)z X(z )
Proof:
x (n)z
n
z- transform of x (n)
n
x(n)zX(z) X(z)
n
n
[Link] value theorem
For causal signal x(n)
x(0) lt X (z)
z
Where, x(n)z X(z)
Proof:
For causal signal x(n)
X(z)x(n)znn0
X (z) x(0) x(1)z 1x(2)z 2x(3)z 3........
lt X (z) x(0)
z
10Final value theorem
For causal signal, x(n)
x(n) z X(z)
If poles of X (z) are within the unit circle in z-plane, then
x() lim(z 1) X (z)
z1
Proof:
85
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
k
Z{x(n1)x(n)}lim
n0
x(n 1) x(n) zn k
k
zX(z)zx(0)X (z)lim x(n 1) x(n) z
n
k
n0
k
(z1)X (z)zx(0)lim x(n 1) x(n) z
n
k
n0
lim{(z 1) X (z) zx(0)} limlimx(n 1) x(n)z n
k
z1 z1 k
n0
limlimx(1)x(0)x(2)x(1)x(3)x(2)x(k1)x(k)
z1 k
x() x(0)
lim(z 1) X (z) x(0) x() x(0)
z1
x() lim(z 1) X (z)
z1
4.10-Properties of DISCRETE TIME FOURIER TRANSFORM (DTFT)
FT X(ej)
DT
x(n)
x(n)ejn
X ej Analysisequation
n
Xejejnd
1
x(n)
2 synthesis equation
2
Properties of DTFT:
1. Linearity:
DT
x(n)
1
FT X e j 1
FT Xej
DT
x(n)
2 2
Ax1 (n) B x2 (n)A X
DTFT
1 ejBX e j 2
Where A , B are constants.
Proof:
DTFT of Ax1 (n) Bx2 (n) Ax (n)Bx
1 2
(n) e jn
n
Ax(n)
1 Bx
e jn 2 (n) e jnA x(n) 1 e jnB x 2 (n)e j n
n n n n
A X (e
1
j) BX (e1 j)
2. Periodicity
DTFT is periodic with period 2 π
86
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
x(n) FTX e j
DT
where ‗k‘ is an integer.
Xej(2k) Xe j
proof : Xe j(2k)
x(n) ej (2 k )n
n
x(n)e jn x(n)e j n Xe j
n
e j 2 k n
n
sin ce e j 2 k n 1
n int eger
3. Time shiftingproperty
proof :
DTFT of x(nk)is x(nk)ejn
DT
x(n) FT Xe j
n
x(nk) FT e j kXeje
DT n k p n
k p
x(p)e
j(k p)
ej k x(p)e j p
ej k Xe j
p p
4. Time Reversalproperty
FT Xejx(n)
DT
x(n)
FT Xe j
DT
proof :
DTFT of x(n)isx(n)ejn
let n p
n p
x(p)e j(p )
p x(p)(e
j
)p Xe j
p
5. Conjugation
FT Xej
DT
x(n)
FT Xe j
DT
x (n)
87
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
proof :
DTFTof x(n)isx(n)ejn
DTFTof x (n)isx (n)ejnn
x(n)ejn
n
x(n)(e jn) X e j
n
6. Frequency shifting
DT
x(n)
FT X ej
ej0 n FT Xej(
DT
x(n) 0)
proof :
DTFT of x(n) is x(n)ejn
n
DTFTofej0 nx(n)is ej0 nx(n)e jn
n
x(n)ej( )nX ej( )
n
0
0
[Link] in frequency
FTX ej
DT
x(n)
n x(n) j
DTFT
dX ej
d
proof :
DTFTofx(n)isx(n)ejnXej
n
dXej
j nx(n)ejn
d
dX e j
n
jn
j
d nx(n)e
n
dXej
DTFT of nx(n) j
d
88
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
8. Convolution in time
x(n) DT
FT X(ej)
1 1
x(n) DT
FT X(ej)
2 2
x1 (n)* x2 (n)
DT FT X (ej)X
1 2 (e
j)
Proof:
By definition,
x1 (n) * x2 (n) x1 (k )x2 (n k )
k
DTFT of x1(n)*x2 (n) [x1(n)*x2 (n)]ej n
n
x1(k)x2 (n k )ejn
nk
let n k p n
p k
x1 (k)x2 ( p)ej( pk )
pk
x (k)e
1
k
jk
x 2 (p)ejpX 1 (ej)X 2 (e j)
p
9. Convolution in frequency (Multiplication)
x(n) DT
FT X(ej)
1 1
DT
x(n) FT X(ej)
2 2
1
x1 (n)x2 (n) [X1 (ej)X
DTFT
(e j)]
2
2
Proof:
x(n)e
X ej j n Analysisequation
n
Xe e d
1 j j n
x(n) synthesisequation
22
X 1 e je j n d
1
2 2
x1 (n)
X e e
1
x2 (n) j j n
d
2
2
2
X e e d
1
let x2 (n) j j n
changing var iable to
2
2
2
89
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
DTFT of x1 (n)x2 (n) [x1(n)x2 (n)]ej n
1
x (n)
n
X e e j n de j n
j
1
2 2
n 2
X 2e
1
j
x1 (n)e j () n d
22 n
1 22 1X
X 2e jX 1 (ej())d ejX e j
2
2 1
10 .Parseval’s relation for DT signals
x(n) FT X(ej)
DT
1
n
x(n) 2
22
X (e j) 2d
Proof:
X e x(n) e
j jn
DTFT of x(n)
n
Xe e d
1
x(n) j j n
22
x (n)
1
X ej e j n d
22
x(n)
n
2
x(n)x(n)
n
X e e
1 j jn
d
n
x(n)
2
2
1
X
e j
{ x(n)e jn }d
22 n
X e X e jd X (e j) d
1 j 1 2
22 22
RELATIONSHIP BETWEEN FOURIER TRANSFORM AND Z TRANSFORM.
There is a close relationship between Z transform and Fourier transform. If we replace the complex
variable z by e –jω, then z transform is reduced to Fourier transform.
Z transform of sequence x(n) is given by
∞
X(z) = ∑ x (n)z –n (Definition ofz-Transform)
n=-∞
Fourier transform of sequence x(n) is given by
90
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
∞
X(ω) = ∑ x (n)e–jωn (Definition of FourierTransform)
n=-∞
Complex variable z is expressed in polar form as Z= rejωwhere r= |z| and ω is ∟z. Thus we can be
written as
∞
X(z) = ∑ [ x (n) r–n] e–jωn
n=-∞
∞
X(z) z=ejw = ∑ x (n) e–jωn
n=-∞
X(z) z=ejw=x(ω) at |z| = unitcircle.
Thus, X(z) can be interpreted as Fourier Transform of signal sequence (x(n) r–n). Here r–n grows with n if
r<1 and decays with n if r>1. X(z) converges for |r|= 1. hence Fourier transform may be viewed as Z
transform of the sequence evaluated on unit circle. Thus The relationship between DFT and Z transform
is given by
X(z) z=j2∏kn
e =x(k)
The frequency ω=0 is along the positive Re(z) axis and the frequency ∏/2 is
along the positive Im(z)
axis. Frequency ∏ is along the negative Re(z) axis and 3∏/2 is along the
negative Im(z) axis.
91
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Frequency scale on unit circle X(z)= X(ω) on unit circle
Z transform properties
92
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
93
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4.11 INVERSE Z TRANSFORM (IZT)
The signal can be converted from time domain into z domain with the help of z transform (ZT). Similar
way the signal can be converted from z domain to time domain with the help of
inverse z transform(IZT). The inverse z transform can be obtained by using two
different methods.
1) Partial fraction expansion Method (PFE) / Application of residuetheorem
2) Power series expansion Method(PSE)
1. PARTIAL FRACTION EXPANSIONMETHOD
The above equation can be written in partial fraction expansion form and find the coefficient AK and
take IZT.
SOLVE USING PARTIAL FRACTION EXPANSION METHOD (PFE)
94
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
S.N Function (ZT) Time domain sequence Comment
o
1 an u(n) for |z| > a causal sequence
1 -an u(-n-1) for |z| <a anti-causal sequence
1-
az-
1
1 (-1)n u(n) for |z| > 1 causal sequence
2 1
+ -(-1)n u(-n-1) for |z| < a anti-causal sequence
z-
1
-2(3)n u(-n-1) + (0.5)n u(n) stable system
3- for 0.5<|z|<3
4z 2(3)n u(n) + (0.5)n u(n) causal system
-1
3 for |z|>3
1- 3.5z- -2(3)n u(-n-1) - (0.5)n u(-n-1) for |z|<0.5 anti-causal system
1+1.5z-2
-2(1) n u(-n-1) + (0.5)n u(n) stable system
for 0.5<|z|<1
4 1 2(1)n u(n) + (0.5)n u(n) causal system
for |z|>1
1- 1.5z- -2(1)n u(-n-1) - (0.5)n u(-n-1) for |z|<0.5 anti-causal system
1+0.5z-2
5 1+2 z-1+ z-2 2δ(n)+8(1)n u(n)- 9(0.5)n u(n) causal system
for |z|>1
1-3/2 z-1+0.5z-2
1+ z-1 (1/2-j3/2) (1/2+j1/2)n u(n)+ causal system
6 (1/2+j3/2) (1/2+j1/2)n u(n)
1- z-1 + 0.5z-2
1 –(0.5) z-1 4(-1/2)n u(n) – 3 (-1/4)n u(n) for |z|>1/2 causal system
7
1-3/4 z-1+1/8 z-2
1- 1/2z-1 (-1/2)n u(n) for |z|>1/2 causal system
8
1- 1/4z-2
95
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
z+1 δ(n)+ u(n) – 2(1/3)n u(n) causal system
9 for |z|>1
3z2 - 4z + 1
96
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
5z 5(2n-1) causal system
10 for |z|>2
(z-1) (z-2)
z3 4-(n+3)(1/2)n causal system
11 for |z|>1
(z-1) (z-1/2)2
RESIDUE THEOREM METHOD
In this method, first find G(z)= zn-1 X(Z) and find the residue of G(z) at various poles of X(z).
S. No Function (ZT) Time domain Sequence
1 z For causal sequence (a)n u(n)
z–a
2 z (2n -1 ) u(n)
(z–1)(z-2)
3 z2 + z (2n+1) u(n)
(z – 1)2
4 z3 4 – (n+3)(0.5)n u(n)
(z-1) (z–0.5)2
3. POWER-SERIES EXPANSION METHOD
The z transform of a discrete time signal x(n) is given as
∞
X(z) = ∑ x (n) z –n (1)
n=-∞
Expanding the above terms we have
x(z) = …..+x(-2)Z2+ x(-1)Z+ x(0)+ x(1) Z-1 + x(2)Z2+….. (2)
This is the expansion of z transform in power series form. Thus sequence x(n)is
given as x(n) ={…..,x(-2),x(-1),x(0),x(1),x(2),…................}.
Power series can be obtained directly or by long division method.
97
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
SOLVE USING ―POWER SERIES EXPANSION― METHOD
[Link] Function (ZT) Time domain Sequence
1 z For causal sequence an u(n)
For Anti-causal sequence -an u(-n-1)
z-a
2 1 {1,3/2,7/4,15,8,….........} For |z| >1
{… 14,6,2,0,0} For |z| < 0.5
1- 1.5 z-1+0.5z-2
3 z2+z {0,1,4,9,…..} For |z| > 3
z3 -3z2+3z -1
4 z2(1-0.5z-1)(1+z-1) (1-z-1) X(n) ={1,-0.5,-1,0.5}
5 log(1+az-1)
(-1)n+1an/n for n≥1 and |z|>|a|
98
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Sample Problem:
1. Obtain the z transformof,
2. Find the inverse z transformof,
99
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
UNIT V
LINEAR TIME INVARIANT DISCRETE TIME SYSTEMS
5.1 Introduction
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 differentialequations.
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).
5.2 BLOCK DIAGRAMREPRESENTATION
Block diagram representation of
LTI systems with rational system function can be represented as constant-coefficient difference
equation
• The implementation of difference equations requires delayed values ofthe
– input
– output
10
0
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
– intermediateresults
• The requirement of delayed elements implies need forstorage
• We also need meansof
– addition
– multiplication
Direct Form I
General form of difference equation
Alternative equivalent form
100
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Cascadeform
General form for cascade implementation
Parallel form
► Represent system function using partial fractionexpansion
5.3 CONVOLUTIO NSUM
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,
101
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
For the case of discrete-time convolution, here are two convolution sum examples. The first employs finite
extent sequences (signals) and the second employs semi-infinite extent signals. You encounter both types of
sequences in problem solving, but finite extent sequences are the usual starting point when you‘re first
working with the convolution sum.
Two finite length sequences
Consider the convolution sum of the two sequences x[n] and h[n], shown here, along with the convolution
sum setup.
When convolving finite duration sequences, you can do the analytical solution almost by inspection or
perhaps by using a table (even a spreadsheet) to organize the sequence values for each value of n, which
produces a nonzero overlap between h[k] and x[n – k].
The support interval for the output follows the rule given for the continuous-time domain. The output y[n]
starts at the sum of the two input sequence starting points and ends at the sum of input sequence ending
points. For the problem at hand this corresponds to y[n] starting at [0 + –1] = –1 and ending at [3 + 1] = 4.
Looking at Figure b, you can see that as n increases from n < –1, first overlap occurs when n = –1. The last
point of overlap occurs when n – 3 = 1 or n = 4. You can set up a spreadsheet table to evaluate the six sum-
of-products related to the output support interval.
102
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
One finite and one semi-infinite sequence
As a second example of working with the convolution consider a finite duration pulse sequence of 2M + 1
points convolved with the semi-infinite exponential sequence an u[n] (a real exponential decay starting
from n = 0). A plot of the waveforms is given here.
With the help of Figure b, you have three cases to consider in the evaluation of the convolution for all values
of n. The support interval for the convolution is
Here are the steps for each case:
Case 1: From Figure b, you see that for n + M < 0 or n < –M no overlap occurs between thetwo
sequences of the sum, so y[n] = 0.
103
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Case 2: Partial overlap between the two sequences occurs when n + M ≥ 0 and n – M ≤ 0 or –
M ≤ n ≤ M. The sum limits start at k = 0 and end at k =n + M. Using the finite geometric seriessum
formula, the convolution sum evaluatesto
Case 3: Full overlap occurs when n – M > 0 or n >M. The sum limits under this case run from k = n–
M to k = n + M. Again, using the finite geometric series sum formula, the convolution sum evaluates to
Putting the pieces together, the complete analytical solution for this problem is
Graphically understanding convolution
Convolution can be seen as a graphical process:
1. Plot x[m]x[m] with dependent variablemm
2. Plot h[−m]h[−m] with dependent variable mm (hh reflected aroundm=0m=0).
3. Plot h[n−m]h[n−m] with dependent variable mm (nn can shift h[n−m]h[n−m] from −∞−∞ (all the
way to the left) to ∞∞ (all the way to theright).
4. For each shift (i.e. nn), compute y[n]=∑∞m=−∞x[m]h[n−m] y[n]=∑m=−∞∞x[m]h[n−m](i.e.,
multiply x[m]h[n−m]x[m]h[n−m] and then sum the result ).
104
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
1.
105
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
106
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
107
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
LTI systems are causal if
h[n] = 0 n < 0
5.4 LTI System analysis usingDTFT
5.5 LTI SYSTEMS ANALYSIS USINGZ-TRANSFORM
108
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
POLE –ZERO PLOT
1. X(z) is a rational function, that is a ratio of two polynomials in z-1 orz.
The roots of the denominator or the value of z for which X(z) becomes infinite, defines locations
of the poles. The roots of the numerator or the value of z for which X(z) becomes zero, defines
locations of the zeros.
2. ROC dos not contain any poles of X(z). This is because x(z) becomes infiniteat
the locations of the poles. Only poles affect the causality and stability of the system.
3. CASUALTY CRITERIA FOR LSISYSTEM
LSI system is causal if and only if the ROC the system function is exterior to
the circle. i. e |z| > r. This is the condition for causality of the LSI system in terms of z transform.
(The condition for LSI system to be causal is h(n) = 0 ….. n<0 )
4. STABILITY CRITERIA FOR LSISYSTEM
Bounded input x(n) produces bounded output y(n) in the LSI system only if
∞
∑ |h(n)| <∞
n=-∞
With this condition satisfied, the system will be stable. The above equation states that the LSI
system is stable if its unit sample response is absolutely summable. This is necessary and
sufficient condition for the stability of LSI system.
H(z) = ∑ h (n)z–n Z-Transform.……(1)
n=-∞
109
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Taking magnitude of both the sides
∞
|H(z)|= ∑ h(n)z–n ...................................................................................(2)
n=-∞
Magnitudes of overall sum is less than the sum of magnitudes of individual sums.
∞
|H(z)|≤ ∑ h(n) z-n
n=-∞
|H(z)|≤ ∑ |h(n)| |z-n| ….(3)
n=-∞
5. If H(z) is evaluated on the unit circle |z-n|=|z|=1.
Hence LSI system is stable if and only if the ROC the system function includes the unit circle. i.e
r < 1. This is the condition for stability of the LSI system in terms of z transform. Thus
For stable system |z| < 1
For unstable system |z| > 1
Marginally stable system |z| = 1
Im(z)
z-Plane
110
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Poles inside unit circle gives stable system. Poles outside unit circle gives
unstable system. Poles on unit circle give marginally stable system.
6. A causal and stable system must have a system function that convergesfor
|z| > r < 1.
STANDARD INVERSE TRANSFORMS
S. No Function (ZT) Causal Sequence Anti-causal sequence
|z| > |a| |z| <|a|
1 z n
(a) u(n) n
-(a) u(-n-1)
z–a
2 z u(n) u(-n-1)
z–1
3 z2 (n+1)an -(n+1)an
(z – a)2
4 zk 1/(k-1)! (n+1) (n+2)……an -1/(k-1)! (n+1) (n+2)………an
(z – a)k
5 1 δ(n) δ(n)
6 Zk δ(n+k) δ(n+k)
7 Z-k δ(n-k) δ(n-k)
[Link] z Transform (Bilateral) One sided z Transform (Unilateral)
1 z transform is an infinite power series One sided z transform summation index varies from 0 to
because summation index varies from ∞ ∞. Thus One sided z transform are given by
to -∞. Thus Z transform are given by ∞
∞
X(z) = ∑ x (n) z –n
X(z) = ∑ x (n) z –n
n=-∞ n=0
2 z transform is applicable for relaxed One sided z transform is applicable for those systems
systems (having zero initial condition). which are described by differential equations with non zero
initial conditions.
3 z transform is also applicable for non- One sided z transform is applicable for causal systems
causal systems. only.
4 ROC of x(z) is exterior or interior to ROC of x(z) is always exterior to circle hence need not to
circle hence need to specify with z be specified.
transform of signals.
111
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
5.6Sample Problems:
Unilateral z-transform
Unilateral z-transform is used in the analysis of non-relaxed DTsystems.
Non-relaxed DT systems are systems with non zero initialconditions.
Unilateral Z- Transform of a signal x(n) is definedby
X(z)x(n)znn0
time shifting property
x(n)zX(z)
Case 1:
z k k
x(n)z X(z)
n
x(nk)z n1
X(z)x(n)znn0
Proof:
Unilateralz-transformofx(nk)x(nk)zn
n0
let n k m n
m k
x(nk)znx(m)z(mk)zkx(m)zm
n0 mk mk
1
x(m)z x(m)z ( m)
k (m)
z
mk m0
k
z k x(m)z (m) x(m)z ( m)
m1 m0
k
zk x(n)z(n) x(m)z (m)
n1 m0
k x(n)znX (z)
zk
n1
Note:
Unilateral z- transform of y(n 3) is,
112
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3 y(n)znY (z)
z3
n1
z 3Y (z) y(1)z 2y(2)z 1y(3)
Case2:
z k k 1 n
x(nk)z X(z)x(n)z
n0
X(z)x(n)znn0
Proof:
Unilateralz-transformofx(nk)x(nk)zn
n0
let n k m
n m k
x(nk)
n0
zn x(m)
mk
z(mk) zk x(m)zm
mk
k 1
z k x(m)z (m) x(m)z (m)
m0 m0
k 1
x(n)z
n
z k X(z)
n0
Discrete time systems has one more type ofclassification.
1. RecursiveSystems
2. Non-RecursiveSystems
S. No Recursive Systems Non-Recursive systems
1 In Recursive systems, the output depends upon past, In Non-Recursive systems, the
present, future value of inputs as well as past output. output depends only upon past,
present or future values of inputs.
2 Recursive Systems has feedback from output to No Feedback.
input.
3 Examples y(n) = x(n) + y(n-2) Y(n) = x(n) + x(n-1)
Example:
Find the step response of the system described by the difference equation
y(n)y(n1)2y(n2)x(n1)2x(n2) given the initialconditions
y(1) 0.5 ; y(2) 0.25
Solution:
y(n) y(n 1) 2 y(n 2) x(n 1) 2x(n 2)
Taking unilateral z- transform
113
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Y (z) z 1Y (z) y(1) 2 z 2Y (z) z 1y(1) y(2) z 1X (z) x(1)
2 z 2X (z) z 1x(1) x(2)
Y (z) z 1Y (z) 0.5 2 z 2Y (z) z 1(0.5) 0.25 z 1X (z) 2 z 2X (z)
sin ce x(n) u(n) &x(1) 0; x(2) 0
Y (z) 2z 2Y (z) z 1Y (z) z 1 z 1X (z) 2 z 2X (z)
Y (z)1 2z 2z 1z 1X (z)z 1 2z 2
Y (z) z 2z X (z) z 21
2
z2 z2 z
z
sin ce x(n) u(n) &X (z)
z1
Y (z) z 2z
2
z
z 21
z2 (z 1) z2 z
z(z2) z z(z 2) z(z 1) (z
Y (z)
(z 2)(z 1)(z 1) (z2)(z1) 2)(z 1)(z 1)
2z 2z
(z 2)(z 1)(z 1)
2z 1 A
Y (z)
B C
z (z2)(z1)(z1) (z 2) (z 1) (z 1)2
1 1
A ; B ; C 1
3 3
1 1
z z
3 3 z
Y(z)
(z2) (z1) (z 1)2
y(n) inverseztransformofY(z)
1 1
y(n) u(n) (2)nu(n)nu(n)
3 3
Example:
Find the difference equation description for the system with transfer function
5Z 2
H(Z)
Z 2 3Z 2
Ans:
5Z2 Y(Z) 5Z 12Z2
H(Z)
Z 2 3Z 2 X (Z) 1 3Z 1 2Z 2
(1 3Z 1 2Z 2 )Y (Z ) (5Z 1 2Z 2 ) X (Z )
Taking inverse z transform,
y(n) 3y(n 1) 2 y(n 2) 5x(n 1) 2x(n 2)
114
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
System analysis using DTFT
For LTI DT system
y(n) x(n) h(n)
y(n) output
x(n) input
h(n) impulseresponse
In frequency domain,
YejXejHej
YejDTFTofy(n)
X e jDTFT of x(n)
HejDTFTofh(n)
Yej
H e j frequency Re sposeof the system
Xej
Frequency response of the system
Plot of magnitudeof Hej Vs frequency is magnitude response of the system
Plot of phase angleof Hej Vs frequency is phase response of the system
Example:
Determine and sketch the magnitude response of the system described by the difference
1
equation y(n) x(n)x(n1)x(n2)3
Y e j X e j e jX e j e j 2X e j
1
Yej 1
3
j
j
e
j 2
e j
He 1 e 1 2 cos()
Xe j 3 3
magnitudeof H e 1 2 cos()
j 1
3
o Compute magnitudeof H e jfor values of frequency ranging from to
o Plot of magnitudeof Hej Vs frequencyis magnitude response of the system
Example:
Consider a system consisting of the cascade of two LTI systems with frequency responses
2 e j j 1
j
H1 (e ) 1 and H2 (e ) . Find the difference equation
1 1
1 e j 1 e j e2 j
2 2 4
describing the overall system.
Frequency response of the overall system is H (e j) H1 (e j)H 2 (e j)
115
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
j 2 ej Y (e j)
H (e )
1 X(ej)
1 e j3
8
1
Y(ej) e j3Y(ej)2X(ej)e jX(e)8
taking inverseDTFT ,
1
y(n) y(n 3) 2x(n) x(n 1)
8
Example:
Determine the frequency response of the system
1 1
described by the difference equation, y(n) y(n 1) y(n 2) x(n 1) .
6 6
taking DTFT , and rearranging,
j Y(ej) e j
frequency responseH (e )
X(ej) 1 1 ej1 ej 2
6 6
1. Consider the system described by the differenceequation.
116
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
2. Compute the impulse response of the system describedby,
117
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4. Obtain the structures realization of LTI system
118
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
119
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Find the convolution of x(n)=[1,1,1,1,2,2,2,2] with h(n)=[3,3,0,0,0,0,3,3] by using
matrix method.
Solution: By using matrix method, N=8
120
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
REALIZATION
A linear time invariant discrete-time systems characterized by the general linear constant coefficient
difference equation
There are various forms to implement above equation either in hardware or in software. For each set
of equation, we can construct a block diagram consisting of an interconnection of delay elements,
multipliers and adders. We referred to such a block diagram as a realization of the system.
The computational complexity refers to the number of operations (like multiplication, addition etc.,)
required to compute an output value of the system
121
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Tapped Delay Line (TDL) or Transversal System (TS)
Direct form is one of the simplest structures. The direct realization can be obtained from the non-
recursive difference equation given in equation
The structure has M-1 memory location for storing M-1 previous inputs. The structure has M
complex multiplication and M-1 additions. The output is the weighted linear combination of M-1
past input and the weighted current value of the input.
Direct Form (N = odd)
When the FIR system has linear phase, the unit sample response of the system satisfies either the
symmetry or asymmetry condition
h(n) = ± h(M – 1 – n) 7
for such system, the number of multiplication is reduced from M to M/2 for even M and M to (M-
1)/2 for odd M.
122
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
123
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
124
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
PROBLEMS: ON CONVOLUTION
125
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
126
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
127
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
128
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
PROBLEMS
1. The input and output of an LTI system are relatedby
Note that without further information such as the initial condition, this equation does not
uniquelyspecify when is given. Taking z-transform of this equation and using the
time shifting property, weget
and the transfer function can be obtained
Note that the causality and stability of the system is not provided by this equation, unlessthe
ROCof this is specified. Consider these two possible ROCs:
If ROCis , it is outsidethepole and includes the unit circle. The
system is causal andstable:
If ROCis , it is inside thepole and does not include the unit
circle. The system is anti-causal andunstable:
129
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
First order system
The first order discrete system is described by
Theimpulse response can be found by solving thefollowing
to be
Alternatively, we can take z-transform of the DE and get
and the transfer function of the system (assumed causal)
has azeroat and apole and its ROC istheregion outside
[Link] , then theunitcircle can be included in the ROC, the Fourier
transform exists and the system is stable. The impulse response (unit sample response) of the
systemis
Although seems to have a form different from the typical impulse response incontinuous
case , they are essentially thesame as can be rewrittenas
130
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
where
Letting in , we get the frequency response function of thesystem
where and are two vectors in z-plane definedas
Foranyfrequency represented byapoint on the unit circle, the
magnitude and phase angle of the frequency response function can be represented in the z-plane
as
and
which can be evaluated graphically in the z-plane asthe frequency changes in the
range . Ifweassume , then when , the denominator reaches
itsminimumof , and is maximizedto be ; andwhen ,the
denominator reaches its maximumof , and is minimized to
be . Thephaseangle of is zero when or ,
and isnegativefor and positive for .
131
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
System Algebra and Block Diagram
Z-transform converts time-domain operations such as difference and convolution into algebraic
operations in z-domain. Moreover, the behavior of complex systems composed of a set of
interconnected LTI systems can also be easily analyzed in z-domain. Some simple
interconnections of LTI systems are listed below.
Parallel systems: If the system is composed of two LTI systems
with and connected in parallel, its impulse response is
or in s-domain
Serial or cascade system: If the system is composed of two LTIsystems
with and connected in series, its impulse responseis
or in s-domain
132
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Feedback system: If the system is composed of an LTIsystemwith in aforward
path and anotherLTIsystem in a feedback path,its output can be implicitly
found in timedomain
or in s-domain
While it is difficult to solve the equation in time domain to find an explicit expressionfor
output , it is easy to solve the algebraic equation in z-domain to
find
133
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
and the transfer function can be obtained
The feedback could be either positive or negative. For the latter, there will be anegative
sign infront of and of the feedback pathso
that and
Example 0: The transfer function of a first order LTI system
is
Comparingthis with the transfer function of the feedback system, we see that a first
order system can be represented as a feedbacksystemwith in the forward path,
and for theproduct of and (a delay elementwithinput and
output ) in the negative feedbackpath.
Example 1:
134
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
This equation can be rewritten as:
where
can be obtained the same way as in previous example. Once and are
available, we can easily obtain :
Example 2: Consider a second order system with transfer function
These three expressions of this correspond to three different block diagram
representations of the system. The last two expressions are, respectively, the cascade and the
parallel representations (same as the corresponding cases in Laplace transform), while the first
one is the direct representation, as shown below. We first consider a general 2nd order system
Weseethat is the linear combination of the delayed versions of itself and the
input which can be represented as a feedback system with two feedback paths
of and . In thisparticular system, and .
135
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Example 3: A second order system with transfer function
This system can be represented as a cascade of two systems
and
136
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Thefirstsystem can be implemented by two delay elements with proper feedback paths
as shown in the previous example, and the second system is a linearcombination
of , and , all of which are available along the feedback path of
the first system. The over all system can therefore be represented as shown. Obviously theblock
diagram of this example can be generalized to represent any system with a rational transfer
function
137
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
EXAMPLES OF DT CONVOLUTION
138
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
139
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
The u(n) comes from our first case above since s(n) = 0 for n < 0, and obviously the other part
comes from the expression found in the second case above.
140
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
3. In the convolution sum, the impulse response is written as h(n−k), meaning that in the k
domain, the impulse response is shifted by n and flipped around thatpoint
We can visualize the convolution operation as that shifted-and-flipped impulse response
sliding along the k axis from −∞ to ∞ as the summation occurs.
Whenever there is some non-zero overlap between this shifted-and-flipped impulse
response and the input signal, the system output will be non-zero (unless the non-zero
overlaps cancel each other).
141
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
142
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
143
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
4. Finding the impulse response of a diffeq system. Find the impulse response of thesystem
described by the followingdiffeq:
y[n] = 4 3 y[n − 1] − 7 12 y[n − 2] + 1 12 y[n − 3] + x[n] − x[n − 3] .
Step 0: Find the system function. (linearity, shift property)
144
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
145
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
146
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
Why Are Z Transforms Used?
You should know that Laplace transform methods are widely used for analysis in linear
systems. Laplace transform methods are used when a system is described by a linear differential
equation, with constant coefficients. However:
There are numerous systems that are described by difference equations - notdifferential
equations - and those systems are common and different from those described by
differentialequations.
Systems that satisfy difference equations include thingslike:
o Computer controlled systems - systems that take measurements with digital I/O
boards or GPIB instruments, calculate an output voltage and output thatvoltage
147
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
digitally. Frequently these systems run a program loop that executes in a fixed
interval of time.
Other systems that satisfy difference equations are those systems with Digital Filters -
which are found anywhere digital signal processing - digital filtering is done. That
includes:
o Digital signal transmission systems like the telephonesystem.
o Systems that process audio signals. For example, a CD contains digital signal
information, and when it is read off the CD, it is initially a digital signal thatcan
be processed with a digitalfilter.
At this point, there are an incredible number of systems we use every day that have digital
components which satisfy difference equations.
In continuous systems Laplace transforms play a unique role. They allow system and
circuit designers to analyze systems and predict performance, and to think in different terms -
like frequency responses - to help understand linear continuous systems. They are a very
powerful tool that shapes how engineers think about those systems. Z-transforms play the rolein
sampled systems that Laplace transforms play in continuoussystems.
In continuous systems, inputs and outputs are related by differential equations and
Laplace transform techniques are used to solve those differentialequations.
In sampled systems, inputs and outputs are related by difference equations and Z-
transform techniques are used to solve those differentialequations.
In continuous systems, Laplace transforms are used to represent systems withtransfer
functions, while in sampled systems, Z-transforms are used to represent systems with
transfer functions.
There are numerous sampled systems that look like the one shown below.
An analog signal is converted to a digital form in anA/D.
The digital signal is processedsomehow.
The processed digital signal is converted to an analog signal for use in the analogworld.
The processing can take many forms.
In a voice transmission situation, the processing might be to band-limit the signal and
filter noise from thesignal.
In a control situation, a measurement might be processed to calculate a signal to controla
system.
And there are many othersituations.
Goals
In sampled systems you will deal with sequences of samples, and you will need to learn Z-
transform techniques to deal with those signals. In this lesson many of your goals relate to basic
understanding and use of Z-transform techniques. In particular, work toward these goals.
Given a sequence of samples intime,
o Be able to calculate the Z-transform of the sequence for simplesequences.
Given a Z-transform,
148
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
o Be able to determine the poles and zeroes of theZ-transform.
o Be able to locate and plot the poles and zeroes in thez-plane.
Later you will need to learn about transfer functions in the realm of sampled systems. As you
move through this lesson, there are other things you should learn.
Given a Z-transform of a signal, and the pole locations,
o Be able to relate distance from the origin to decayrate.
o Be able to relate angle off the horizontal to the number of samples in a cycleof
signaloscillation.
What Is A ZTransform?
You will be dealing with sequences of sampled signals. Let us assume that we have a
sequence, yk. The subscript "k" indicates a sampled time interval and that ykis the value of y(t)
at the kthsampleinstant.
ykcould be generated from a sample of a time function. Forexample:
o yk= y(kT), where y(t) is a continuous time function, and T is the sampling
interval.
o We will focus on the index variable, k, rather than the exact time, kT, in allthat
we do in thislesson.
It's easy to get a sequence of this sort if a computer is running an A/D board, and measuring
some physical variable like temperature or pressure at some prescribed interval, T seconds. A
sampled sequence like this plays the same role that a continuous signal plays in a continuous
system. It carries information just like a continuous signal.
The Z transform, Y[z], of a sequence, ykis defined as:
We will use the following notation. A large "z" denotes the operation of taking a Z-transform
(i.e., performing the sum above) and the result is usually denoted with an upper-case version of
the variable used for the sampled time function, yk.
Z[yk] =Y[z]
The definition is simple. Take the sequence, and multiply each term in the sequence by a
negative power of z. Then sum all of the terms to infinity. That's it.
149
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
HINTS
Some important examples of systems include filters, encryption and control. •
Time domain vs. frequency domain: We are used to viewing signals in the time domain — that
is, as a function of time. however, there is another domain — namely, the frequency domain, that
is critically important in understanding, analyzing and transforming signals. In lecture, we
introduced the concept of frequency, and the frequency domain. . The process of sampling a
continuous-time signal changes the signal, and it is important to understand how exactly the
sampling process changes the information content of that signal. In due time, we will carefully
look at the implications of sampling, and how choosing an appropriate sampling rate is critically
important to preserving the information contained in a signal.
The frequency domain tells us the relative contribution of sinusoids of different frequencies to
the overall signal. Consider the figure below, which illustrates a simple example of the
transformation between the time and frequency domains. On the left, we plot the function :
150
II YEAR/THIRD SEMESTER EC3354-SIGNALS AND SYSTEM
ALL THE BEST
151
IIYEAR/THIRD SEMESTER EC6303-SIGNALSAND
152
You might also like
- Signals and Systems Overview and AnalysisNo ratings yetSignals and Systems Overview and Analysis18 pages
- Understanding Signals and Systems BasicsNo ratings yetUnderstanding Signals and Systems Basics9 pages
- Signals and Systems Question Bank 2022-23No ratings yetSignals and Systems Question Bank 2022-2330 pages
- Signals and Systems Classification GuideNo ratings yetSignals and Systems Classification Guide61 pages
- Signals and Systems: Key Questions & Answers100% (1)Signals and Systems: Key Questions & Answers13 pages
- Continuous-Time Signal Analysis in EC241No ratings yetContinuous-Time Signal Analysis in EC241119 pages
- Signals and Systems I Syllabus OverviewNo ratings yetSignals and Systems I Syllabus Overview94 pages
- Discrete Time Signals & DSP Course OverviewNo ratings yetDiscrete Time Signals & DSP Course Overview175 pages
- Signals & Systems For Dummies Cheat Sheet - For DummiesNo ratings yetSignals & Systems For Dummies Cheat Sheet - For Dummies8 pages
- Impulse Function Properties in LTI SystemsNo ratings yetImpulse Function Properties in LTI Systems79 pages
- Dynamics of Linear Systems Course GuideNo ratings yetDynamics of Linear Systems Course Guide14 pages
- Convolution and Fourier Transform in LTI SystemsNo ratings yetConvolution and Fourier Transform in LTI Systems16 pages
- Convolution in LTI Systems: x(t) & h(t)No ratings yetConvolution in LTI Systems: x(t) & h(t)16 pages
- Evolution of Computer Technology HistoryNo ratings yetEvolution of Computer Technology History15 pages
- 9.design and Development of A Solar Powered Lighting System Using Motion SenNo ratings yet9.design and Development of A Solar Powered Lighting System Using Motion Sen8 pages
- Vacon NX IO Boards User Manual DPD00884A UK PDF100% (2)Vacon NX IO Boards User Manual DPD00884A UK PDF84 pages
- Tidalflux 2300 F Tidalflux 2300 F Tidalflux 2300 F Tidalflux 2300 FNo ratings yetTidalflux 2300 F Tidalflux 2300 F Tidalflux 2300 F Tidalflux 2300 F24 pages
- Manoj HP: M.Tech Power Electronics ResumeNo ratings yetManoj HP: M.Tech Power Electronics Resume4 pages
- Core Components of Communication SystemsNo ratings yetCore Components of Communication Systems4 pages
- Brain Wave Controlled Robotic Arm For Paralytic and Physically Impaired PatientsNo ratings yetBrain Wave Controlled Robotic Arm For Paralytic and Physically Impaired Patients8 pages
- 1978 - F. R. Moore - An Introduction To The Mathematics of Digital Signal Processing - Part II - Sampling, Transforms, and Digital Filtering PDFNo ratings yet1978 - F. R. Moore - An Introduction To The Mathematics of Digital Signal Processing - Part II - Sampling, Transforms, and Digital Filtering PDF24 pages
- Adaptive Linear Equalizer in Wireless CommunicationNo ratings yetAdaptive Linear Equalizer in Wireless Communication5 pages
- Motor Protection and Control REM615: Product GuideNo ratings yetMotor Protection and Control REM615: Product Guide72 pages
