SSP Notes PDF

SIGNALS & STOCHASTIC PROCESS DEPT.
OF ECE
SIGNALS AND STOCHASTIC PROCESS

II Year B. Tech. ECE – I Sem
Prepared by
B. Vamsi Krishna Mrs. V. Poornima M. Sruthi

Assistant Professor Assistant Professor Assistant Professor
Department of Electronics & Communication Engineering
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SIGNALS & STOCHASTIC PROCESS DEPT.OF ECE
EC304ES: SIGNALS AND STOCHASTIC PROCESS

B.Tech. II Year I Sem. LTPC
3103
UNIT – I
Signal Analysis: Analogy between Vectors and Signals, Orthogonal Signal Space, Signal
approximation using Orthogonal functions, Mean Square Error, Closed or complete set of
orthogonal functions, Orthogonality in Complex functions, Exponential and Sinusoidal signals,
Concepts of Impulse function, Unit Step function, Signum function.
Signal Transmission through Linear Systems: Linear System, Impulse response, Response of a
Linear System, Linear Time Invariant (LTI) System, Linear Time Variant (LTV) System,
Transfer function of a LTI system, Filter characteristics of Linear Systems, Distortion less
transmission through a system, Signal bandwidth, System bandwidth, Ideal LPF, HPF and
BPF characteristics, Causality and Paley-Wiener criterion for physical realization, relationship
between Bandwidth and Rise time. Concept of convolution in Time domain and Frequency
domain, Graphical representation of Convolution, Convolution property of Fourier Transforms
UNIT – II
Fourier series, Transforms, and Sampling: Fourier series: Representation of Fourier
series, Continuous time periodic signals, Properties of Fourier Series, Dirichlet’s conditions,
Trigonometric Fourier Series and Exponential Fourier Series, Complex Fourier spectrum.
Fourier Transforms: Deriving Fourier Transform from Fourier series, Fourier Transform of
arbitrary signal, Fourier Transform of standard signals, Fourier Transform of Periodic
Signals, Properties of Fourier Transform, Fourier Transforms involving Impulse function and
Signum function.
Sampling: Sampling theorem – Graphical and analytical proof for Band Limited Signals,
Reconstruction of signal from its samples, Effect of under sampling – Aliasing.
UNIT – III
Laplace Transforms and Z–Transforms: Laplace Transforms: Review of Laplace
Transforms (L.T), Partial fraction expansion, Inverse Laplace Transform, Concept of Region
of Convergence (ROC) for Laplace Transforms, Constraints on ROC for various classes of
signals, Properties of L.T, Relation between L.T and F.T of a signal, Laplace Transform of
certain signals using waveform synthesis.
Z–Transforms: Fundamental difference between Continuous and Discrete time signals,
Discrete time signal representation using Complex exponential and Sinusoidal components,
Periodicity of Discrete time signal using complex exponential signal, Concept of ZTransform
of a Discrete Sequence, Distinction between Laplace, Fourier and Z Transforms, Region
of Convergence in Z-Transform, Constraints on ROC for various classes of signals,
Inverse Z-transform, Properties of Z-transforms.
UNIT – IV
Random Processes – Temporal Characteristics: The Random Process Concept, classification
of Processes, Deterministic and Nondeterministic Processes, Distribution and Density Functions,
concept of Stationarity and Statistical Independence. First-Order Stationary Processes, Second-
Order and Wide-Sense Stationarity, (N-Order) and Strict- Sense Stationarity, Time Averages and
Ergodicity, Autocorrelation Function and Its Properties, Cross-Correlation Function and Its
Properties, Covariance Functions, Gaussian Random Processes, Poisson Random Process.

Random Signal, Mean and Mean-squared Value of System Response, autocorrelation Function of
Response, Cross-Correlation Functions of Input and Output.
UNIT- V:
Random Processes – Spectral Characteristics: The Power Spectrum: Properties, Relationship
between Power Spectrum and Autocorrelation Function, The Cross-Power Density Spectrum,
Properties, Relationship between Cross-Power Spectrum and Cross- Correlation Function.
Spectral Characteristics of System Response: Power Density Spectrum of Response, Cross-Power
Density Spectrums of Input and Output.
TEXT BOOKS:
1. Signals, Systems & Communications - B.P. Lathi, 2013, BSP.
2. Signal and systems principles and applications, shaila dinakar Apten, Cambridez
university press, 2016.
3. Probability, Random Variables & Random Signal Principles - Peyton Z. Peebles, MC
GRAW HILL EDUCATION, 4th Edition, 2001
REFERENCE BOOKS:
1. Signals and Systems - A.V. Oppenheim, A.S. Willsky and S.H. Nawab, 2 Ed.
2. Signals and Signals – Iyer and K. Satya Prasad, Cengage Learning
COURSE OBJECTIVES:
1. This gives the basics of Signals and Systems required for all Electrical Engineering
related courses.
2. This gives concepts of Signals and Systems and its analysis using different transform
techniques.
3. This gives basic understanding of random process which is essential for random
signals and systems encountered in Communications and Signal Processing areas.
COURSE OUTCOMES: Upon completing his course, the student will be able to
1. Represent any arbitrary analog or Digital time domain signal in frequency domain.
2. Understand the importance of sampling, sampling theorem and its effects.
3. Understand the characteristics of linear time invariant systems.
4. Determine the conditions for distortion less transmission through a system.
5. Understand the concepts of Random Process and its Characteristics.
6. Understand the response of linear time Invariant system for a Random Processes.

INDEX
Unit-I Part I: Signal Analysis Page No

1.1 Introduction 05
1.2 Classification of the Signals 05
1.3 Analogy between Vectors and signal 09
1.4 Orthogonal signal space 10
1.5 Signal approximation using Orthogonal functions 10
1.6 Mean square Error 11
1.7 Closed and complete set of orthogonal functions 12
1.8 Orthogonality in complex functions 12
1.9 Standard Signals 13
Part II: Signal Transmission through Linear Systems

1.10 System 16
1.11 Classification of Systems 16
1.12 Transfer Function of an LTI System 18
1.13 Filter characteristics of Linear Systems 19
1.14 Distortion less transmission through a system 19
1.15 Signal bandwidth & System bandwidth 20
1.16 Ideal LPF, HPF and BPF characteristics 21
1.17 Causality and Paley-Wiener Criterion 22
1.18 Relationship between bandwidth and rise time 23
1.19 Convolution in time and frequency domain 24
1.20 Graphical representation of convolution 25
1.21 Convolution properties of Fourier transform 27
Unit- II Fourier series, Fourier Transforms, and Sampling

2.1 Fourier series representation of Periodic signals 30
2.2 Dirichlet’s conditions 30
2.3 Trigonometric Fourier series 30
2.4 Exponential Fourier series 31
2.5 Properties of Continuous time Fourier series 35
2.6 Complex Fourier Spectrum 37
2.7 Fourier Transform 38
2.8 Derivation from FS- FT 38
2.9 Solved Problems 39
2.10 Fourier Transform of Standard Signals 43
2.11 Fourier Transform of Periodic Signal 50
2.12 Properties of the Fourier Transform 53
2.13 Sampling Theorem 55
2.14 Aliasing 60
2.15 Nyquist Rate & Nyquist Interval 60
Unit- III Part I: Laplace Transforms

3.1 Introduction 62
3.2 Relation between Laplace and Fourier transform 63
3.3 Region of Convergence (ROC) of LT 67
3.4 Properties of ROC 69
3.5 Properties of Laplace Transform 70

3.6 Inverse Laplace Transform 75

3.7 Partial Fraction Expansion 75
3.8 Laplace transform using Waveform Synthesis 77
PART - II Z–Transforms
3.9 Introduction 79
3.10 Relation between ZT and DTFT 79
3.11 Relation between Laplace, Fourier and z- transforms 80
3.12 Problems 80
3.13 Region of Convergence (ROC) of Z-Transforms 83
3.14 Properties of ROC 84
3.15 Properties of Z-Transform 87
3.16 Inverse Z-Transform 89
UNIT – IV Random Processes – Temporal Characteristics

4.1 Introduction 98
4.2 Classification of Random Processes 98
4.3 Deterministic and Non-deterministic processes 100
4.4 Distribution Function and Density function 101
4.5 Independence and Stationary Random Process 102
4.6 Time Averages and Ergodicity 105
4.7 Autocorrelation Function and its Properties 105
4.8 Properties of Cross Correlation Function 107
4.9 Covariance functions for random processes 108
4.10 Gaussian Random Process 109
4.11 System Response 109
UNIT – V Random Process - Spectral Characteristics

5.1 Introduction 111
5.2 Power Spectral Density of a random Process 111
5.3 Properties of power density spectrum 114
5.4 Cross Power density spectrum 115
5.5 Properties of cross power density spectrum 117
5.6 Problems 118
5.7 Spectral characteristics of system response 120
5.8 Problems 122
Question Bank

UNIT - I
Signal Analysis & Transmission through Linear Systems
1.1. Introduction:
In typical applications of science and engineering, we have to process signals, using
systems. While the applications can be varied large communication systems to control systems but
the basic analysis and design tools are the same. In a signals and systems course, we study these
tools: convolution, Fourier analysis, z-transform, and Laplace transform.
The use of these tools in the analysis of linear time-invariant (LTI) systems with
deterministic signals. For most practical systems, input and output signals are continuous and these
signals can be processed using continuous systems. However, due to advances in digital systems
technology and numerical algorithms, it is advantageous to process continuous signals using
digital systems by converting the input signal into a digital signal. Therefore, the study of both
continuous and digital systems is required. As most practical systems are digital and the concepts
are relatively easier to understand, we describe discrete signals and systems, immediately followed
by the corresponding description of continuous signals and systems.
1.2. Classification of the Signals:

The Signals can be classified into several categories depending upon the criteria and for
its classification. Broadly the signals are classified into the following categories
1. Continuous, Discrete and Digital Signals
2. Periodic and Aperiodic Signals
3. Even and Odd Signals
4. Power and Energy Signals
5. Deterministic and Random signals
1.2.1 Continuous-time and Discrete-time Signals:
Continuous-Time (CT) Signals: They may be defined as continuous in time and continuous in
amplitude as shown in Figure 1.1. Ex: Speech, audio signals etc..
Discrete Time (DT) Signals: Discretized in time and Continuous in amplitude. They may also be
defined as sampled version of continuous time signals. Ex: Rail track signals.
Digital Signals: Discretized in time and quantized in amplitude. They may also be defined as
quantized version of discrete signals.
1.2.2 Periodic Signals

A CT signal x(t) is said to be periodic if it satisfies the following condition
x (t) = x (t + T0) (1.1)
The smallest positive value of T0 that satisfies the periodicity condition Eq.(1.1), is referred as
the fundamental period of x(t). The reciprocal of fundamental period of a signal is fundamental
frequency f0.
Likewise, a DT signal x[n] is said to be periodic if it satisfies
x(n) = x (n + N0) (1.2)

Figure 1.1: Description of Continuous, Discrete and Digital Signals

The smallest positive value of N0 that satis es the periodicity condition Eq.(1.2) is referred
to as the fundamental period of x [n].
Note: All periodic signals are ever lasting signals i.e. they start at -1 and end at +1 as shown in
below Figure 1.2.
Figure 1.2: A typical periodic signal

Ex.1.1 Consider a periodic signal is a sinusoidal function represented as x (t) = A sin (10t + 20)
The time period of the signal T0 is 10
Ex.1.2 CT tangent wave: x (t) = tan (10t) is a periodic signal with period T =10
Note: Amplitude and phase difference will not affect the time period.
i.e. 2 sin (3t), 4 sin (3t), 4 sin (3t + 32) will have the same time period
1.2.3 Even and Odd Signals
Any signal can be called even signal if it satisfies x(t) = x(-t) or x(n) = x(-n). Similarly any signal
can be called odd signal if it not satisfies x(t) = x(-t) or x(n) = x(-n). Figure 1.2, shows an
example of an even and odd signal whereas Figure 1.3 shows neither even nor odd signal.

Any signal X(t) can be expressed in terms of even component Xe(t) and odd component X0(t).
X(t) =Xe(t)+Xo(t), Xe(t)+=(X(t) + X(-t)) / 2, Xo(t)+=(X(t) - X(-t)) / 2
1.2.4 Energy and Power signals

A signal x(t) (or) x(n) is called an energy signal if total energy has a non - zero finite value
i.e. 0 < Ex < 1 and Pavg = 0
A signal is called a power signal if it has non - zero nite power i.e. 0 < Px < 1 and E = 1.
A signal can't be both an energy and power signal simultaneously. The term instantaneous power
is reserved for the true rate of change of energy in a system. All periodic signals are power
signals and all finite durations signals are energy signals.
1.2.5 Deterministic and Random signal

A deterministic signal is a signal in which each value of the signal is fixed and can
be determined by a mathematical expression, rule, or table. Because of this the future values of the
signal can be calculated from past values with complete confidence. On the other hand, a random
signal has lot of uncertainty about its behaviour. The future values of a random signal cannot be
accurately predicted and can usually only be guessed based on the averages of sets of signals.

1.3. Analogy between Vectors and Signals
Vector
A vector contains magnitude and direction. The name of the vector is denoted by bold face type
and their magnitude is denoted by light face type.
Example: V is a vector with magnitude V. Consider two vectors V1 and V2 as shown in the
following diagram. Let the component of V1 along with V2 is given by C12V2. The component of
a vector V1 along with the vector V2 can obtained by taking a perpendicular from the end of V1
to the vector V2 as shown in diagram:
The vector V1 can be expressed in terms of vector V2
V1= C12V2 + Ve
Where Ve is the error vector.
But this is not the only way of expressing vector V1 in terms of V2. The alternate possibilities
are: V1=C1V2+Ve1 & V2=C2V2+Ve2

1.4
1.5 Signal approximation using Orthogonal functions

1.6.

1.7
1.8

1.9. Standard Signals



PART II - Signal Transmission through Linear Systems

1.10. System:
1.11. Classification of Systems: Systems are classified into the following categories:
 Linear and Non-linear Systems

 Time Variant and Time Invariant Systems
 Liner Time variant and Liner Time invariant systems
 Static and Dynamic Systems
 Causal and Non-causal Systems
 Stable and Unstable System
Linear and Non-linear Systems
A system is said to be linear when it satisfies superposition and homogeneity principles.

Consider two systems with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then,
according to the superposition and homogeneity principles,
T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

∴ T [a1 x1(t) + a2 x2(t)] = a1 y1(t) + a2 y2(t)
From the above expression, is clear that response of overall system is equal to response
of individual system.
Example: x (t) = x2(t)
Solution: y1 (t) = T[x1(t)] = x12(t)
y2 (t) = T[x2(t)] = x22(t)
T [a1 x1(t) + a2 x2(t)] = [ a1 x1(t) + a2 x2(t)]2
Which is not equal to a1 y1(t) + a2 y2(t). Hence the system is said to be non linear.
Time Variant and Time Invariant Systems

A system is said to be time variant if its input and output characteristics vary with time.
Otherwise, the system is considered as time invariant.
The condition for time invariant system is: y (n , t) = y(n-t)
The condition for time variant system is: y (n , t) ≠ y(n-t)
Where y (n , t) = T[x(n-t)] = input change
y (n-t) = output change

Example: y(n) = x(-n)

y(n, t) = T[x(n-t)] = x(-n-t)
y(n-t) = x(-(n-t)) = x(-n + t)
∴ y(n, t) ≠ y(n-t). Hence, the system is time variant.
Liner Time variant (LTV) and Liner Time Invariant (LTI) Systems
If a system is both liner and time variant, then it is called liner time variant (LTV) system.
If a system is both liner and time Invariant then it is called liner time invariant (LTI) system.
Static and Dynamic Systems
Static system is memory-less whereas dynamic system is a memory system.
Example 1: y(t) = 2 x(t)
For present value t=0, the system output is y(0) = 2x(0).
Here, the output is only dependent upon present input. Hence it is memory less or static.
Example 2: y(t) = 2 x(t) + 3 x(t-3)
For present value t=0, the system output is y(0) = 2x(0) + 3x(-3).
Here x(-3) is past value for the present input for which the system requires memory to get this
output. Hence, the system is a dynamic system.
Causal and Non-Causal Systems

A system is said to be causal if its output depends upon present and past inputs, and does not
depend upon future input.
For non-causal system, the output depends upon future inputs also.
Example 1: y(n) = 2 x(t) + 3 x(t-3)
For present value t=1, the system output is y(1) = 2x(1) + 3x(-2).
Here, the system output only depends upon present and past inputs. Hence, the system is causal.
Example 2: y(n) = 2 x(t) + 3 x(t-3) + 6x(t + 3)
For present value t=1, the system output is y(1) = 2x(1) + 3x(-2) + 6x(4) Here, the system output
depends upon future input. Hence the system is non-causal system.
Stable and Unstable Systems

The system is said to be stable only when the output is bounded for bounded input. For a
bounded input, if the output is unbounded in the system then it is said to be unstable.
Note: For a bounded signal, amplitude is finite.
Example 1: y (t) = x2(t)
Let the input is u(t) (unit step bounded input) then the output y(t) = u2(t) = u(t) = bounded
output. Hence, the system is stable.
Example 2: y (t) = ∫x(t)dt
Let the input is u (t) (unit step bounded input) then the output y(t) = ∫u(t)dt
= ramp signal (unbounded because amplitude of ramp is not finite, it goes to infinite when
t → infinite). Hence, the system is unstable.

1.12. Transfer Function of an LTI System:

The transfer function of a continuous-time LTI system may be defined using
Fourier transform or Laplace transform. The transfer function is defined only under zero initial
conditions.
A continuous time system is shown in fig:
i/p signal The o/p signal

x(t) h(t)or H(s)or H(ω) y(t)
The i/p spectrum The o/p spectrum
X(s) or X(ω) Y(s) or Y(ω)
Fig: A system
The transfer function of a LTI system H(ω) is defined as the ratio of the Fourier
transform of the output signal to the Fourier Transform of the input signal when the initial
conditions are zero.
𝐻(𝜔) = (𝜔)/ (𝜔)
H(ω) is a complex quantity having magnitude and phase.
𝐻(𝜔) = |𝐻(𝜔)| 𝜃(𝜔)
The transfer function in frequency domain H(ω) is also called frequency response of the system.
The frequency response is amplitude response plus phase response.
|H(𝜔)|= Amplitude response of the system.
θ(ω)=⌊𝐻(𝜔)= Phase response of the system.
We can say that H(ω) is a frequency domain representation of a system.
Since Y(ω)=H(ω)X(ω)
| (𝜔)| = |𝐻(𝜔)|| (𝜔)|
⌊ (𝜔) = ⌊𝐻(𝜔) + ⌊ (𝜔)
H(ω) has conjugate symmetry property.
H(-ω)=H*(ω)
i.e. H(−𝜔)| = |𝐻(𝜔)| and
⌊𝐻(ω)=-⌊𝐻(𝜔)
The impulse response h(t) of a system is the inverse Fourier transform of its transfer function
H(ω).
H(ω)=F[h(t)] & h(t)=𝐹−1 [H(ω)]

1.13. Filter characteristics of Linear Systems:

A filter is a frequency selective network. It allows transmission of signals of certain
frequencies with no attenuation or with very little attenuation, and it rejects or heavily attenuates
signals of all other frequencies. Filters are usually classified according to their frequency response
characteristics as low pass filter (LPF), high- pass filter (HPF), band-pass filter (BPF) and band-
elimination or band stop or band reject filter (BEF, BSF, BRF).
The system modifies the spectral density function of the input. The system acts as a kind
of filter for various frequency components. Some frequency components are boosted in strength,
i.e. they are amplified. Some frequency components are weakened in strength, i.e. they are
attenuated and some may remain unaffected. Similarly, each frequency component suffers a
different amount of phase shift in the process of transmission. The system, therefore, modifies the
spectral density function of the input according to its filter characteristics. The modification is
carried out according to the transfer function H(s) or H(ω), which represents the response of the
system to various frequency components. H(ω) acts as a weighting function or spectral shaping
function to the different frequency components in the input signal. An LTI system, acts as a filter.
A filter is a basically a frequency selective network.
(i) Some LTI systems allow the transmission of only low frequency components and stop
all high frequency components. They are called low – pass filters (LPFs).
(ii) Some LTI systems allow the transmission of only high frequency components and stop
all low frequency components. They are called high – pass filters (HPFs).
(iii) Some LTI systems allow the transmission of only a particular band of frequencies and
stop all other frequency components. They are called band pass filters (BPFs).
(iv) Some LTI systems reject the transmission of only a particular band of frequencies and
allow all other frequency components. They are called band-rejection filters (BRFs).
The band of frequency that is allowed by the filter is called pass-band. The band of
frequency that is severely attenuated and not allowed to pass through the filter is called stop-band
or rejection-band. An LTI system may be characterized by its pass-band, stopband and half power
band width.
1.14. Distortion less transmission through a system

The change of shape of the signal when it is transmitted through a system is called
distortion. Transmission of a signal through a system is said to be distortion less if the output is an
exact replica of the input signal. This replica may have different magnitude and also it may have
different time delay. Mathematically, We can say that a signal x(t) is transmitted without distortion
if the output
y(t) = kx(t-td)
Where k is a constant representing the change in amplitude and td is delay time.
Taking Fourier transform on both sides of the equation for y(t) and using the shifting property, we
have
Y ω = k e-jωtd

Therefore inverse Fourier transform, the corresponding impulse response must be

h(t)=kδ(t-td)
The magnitude of the transfer function is given by |H(𝜔)|=k for all w
The phase shift is given by θ (ω) = ⌊(𝜔)= -ωtd
and it varies linearly with frequency given by θ (ω) = n -ωtd ( n integral )
So for distortion less transmission of a signal through a system, the magnitude

|H(𝜔)|should be a constant, i.e. all the frequency components of the input signal must undergo the
same amount of amplification or attenuation, i.e. the system bandwidth is infinite and the phase
spectrum should be proportional to frequency as shown in above figure. But, in practice, no system
can have infinite bandwidth and hence distortion less conditions are never met exactly.
1.15. Signal bandwidth & System bandwidth:

Signal Bandwidth:
The spectral components of a signal extend from -∞ ∞. Any practical signal has finite
energy. As a result, the spectral components approach zero as ω tends to ∞. Therefore, we neglect
the spectral components which have negligible energy and select only a band of frequency
components which have most of the signal energy is known as the bandwidth of the signal.
Normally, the band is selected such that it contains 95% of total energy depending on the precision.
Fig. Frequency response

System Bandwidth:
For distortion less transmission, we need a system with infinite bandwidth. Due
to physical limitations, it is impossible to construct a system with infinite bandwidth. Actually a
satisfactory distortion less transmission can be achieved by a system with finite, but fairly large
band widths, if the magnitude H ω is constant over this band. The bandwidth of a system is
defined as the range of frequencies over which the magnitude H ω remain within 1/√2 times
(within 3 dB) of its value at mid band . The bandwidth of a system H ω plot is shown in above
figu e is ω2-ω1 whe e ω2 is called the upper cut off frequency or upper 3 dB frequency or upper
half powe f e uency and ω1 is called lower cut off frequency or lower 3dB frequency or lower
half frequency. The band limited signals can be transmitted without distortion, if the system
bandwidth is atleast equal to the signal bandwidth.
1.16. Ideal LPF, HPF and BPF characteristics

An ideal filter has very sharp cutoff characteristics, and it passes signals of certain
specified band of frequencies exactly and totally rejects signals of frequencies outside this band.
Its phase spectrum is linear.
Fig. Ideal filter characteristics
Ideal LPF
An ideal low-pass filter transmits, without any distortion, all of the signals of
frequencies
below a certain frequency ⍵c radians per second. The signals of frequencies above ⍵c
radians/second are completely attenuated. ⍵c is called the cutoff frequency. The corresponding
phase function for distortion less transmission is -⍵td. the transfer function of an ideal LPF is given
by
H(⍵) = 1, ⍵ ˂⍵c

= 0, ⍵ >⍵c
The frequency response characteristics of an ideal LPF are shown in figure (a).
It is a gate function.
Ideal HPF
An ideal high-pass filter transmits, without any distortion, all of the signals of
frequencies above a certain frequency ⍵c radians per second and attenuates completely the signals
of frequencies below ⍵c radians per second, where ⍵c is called the cutoff frequency. The
corresponding phase function for distortion less transmission is -⍵td. the transfer function of an
ideal LPF is given by
H ⍵ = , ⍵ ˂⍵c
= , ⍵ >⍵c
The frequency response characteristics of an ideal HPF are shown in figure (b).
Ideal BPF
An ideal band-pass filter transmits, without any distortion, all of the signals of
frequencies within a certain frequency band ⍵2-⍵1 radians per second and attenuates completely
the signals of frequencies outside this band. (⍵2-⍵1) is the bandwidth of the band-pass filter. The
corresponding phase function for distortion less transmission is -⍵td.
An ideal BPF is given by H(⍵) = 1, ⍵1 ˂⍵˂ ⍵2
= 0, ⍵˂ ⍵1 and ⍵> ⍵2
The frequency response characteristics of an ideal BPF are shown in figure (c).
Ideal BRF
An ideal band-rejection filter rejects totally all of the signals of frequencies within a
certain frequency band ⍵2-⍵1 radians per second and transmits without any distortion all signals
of frequencies outside this band. (⍵2-⍵1) is the rejection band. The corresponding phase function
for distortion less transmission is -⍵td.
An ideal BRF is given by H(⍵) = 0, ⍵1 ˂⍵˂ ⍵2
= 1, ⍵˂ ⍵1 and ⍵> ⍵2
The frequency response characteristics of an ideal BRF are shown in figure (d).
All ideal filters are non-causal systems. Hence none of them is physically realizable.
1.17. Causality and Paley-Wiener Criterion For Physical Realization:

A system is said to be causal if it does not produce an output before the input is applied.
For an LTI system to be causal, the condition to be satisfied is its impulse response must be zero
for t < 0, i.e. h(t)=0 for t<0. Physical realizability implies that it is physically possible to construct
that system in real time. A physically realizable system cannot have a response before the input is
applied. This is known as causality condition. It means the unit impulse response h(t) of a
physically realizable system must be causal. This is the time domain criterion of physical
realizability. In the frequency domain, this criterion implies that a necessary and sufficient
condition for a magnitude function H(ω) to be physically realizable is:

The magnitude function H(ω) must, however, be square integrable before the Paley-wiener
criterion valid, that is,
1.18. Relationship between bandwidth and rise time:

Risetime is an easily measured parameter that provides considerable insight into the
potential pitfalls in performing a measurement or designing a circuit. Risetime is defined as the
time it takes for a signal to rise (or fall for falltime) from 10% to 90% of its final value.
We know that the transfer function of an ideal LPF is given by
⍵c is called the cutoff frequency.
The impulse response h(t) of the LPF is obtained by taking the inverse Fourier transform of the
transfer function H(⍵).
The impulse response of the ideal LPF is shown in below figure. The impulse response
has a peak value at t=td. This value ⍵c/𝜋 is proportional to cutoff frequency ⍵c. The width of the
main lobe is 2𝜋/⍵c. As ⍵c⟶∞, the LPF becomes an all pass filter. As td⟶0, the output response
peak ⟶∞, that is, the output response approaches input.
If the impulse response is known, the step response can be obtained by convolution.
The step response is given by
We have

1.19. Convolution in time and frequency domain
Convolution of signals may be done either in time domain or frequency domain. So

there
are following two theorems of convolution associated with Fourier transforms:
1.Time convolution theorem
2.Frequency convolution theorem
Time convolution theorem
The time convolution theorem states that convolution in time domain is equivalent to
multiplication of their spectra in frequency domain.
Mathematically, if x1(t)↔X1(⍵), x2(t)↔X2(⍵) then, x1(t) * x2(t)↔ X1(⍵)X2(⍵).

Frequency convolution theorem

The frequency convolution theorem states that the multiplication of two functions in
time domain is equivalent to convolution of their spectra in frequency domain.
Mathematically, if x1(t)↔X1(⍵), x2(t)↔X2(⍵), then, x (t) x (t) ↔ 1/2𝜋 [ X (⍵) * X (⍵)]
Interchanging the order of integration, we get
1.20 Graphical representation of convolution

The convolution of two signals can be performed using graphical method. The procedure is:
1. For the given signals x(t) and h(t), replace the independent variable t by a dummy
variable 𝜏 and plot the graph for x(𝜏) and h(𝜏).
2. Keep the function x(𝜏) fixed. visualize the function h(𝜏) as a rigid wire frame and rotate
(or invert) this frame about the vertical axis (𝜏 = 0) to obtain h(-𝜏).
3. Shift the frame along the 𝜏-axis by t sec. the shifted frame now represents h(t-𝜏).
4. Plot the graph for x(𝜏) and h(t-𝜏) on the same axis beginning with very large negative
time shift t.
5. For a particular value of t=a , integration of the product x(𝜏)h(t-𝜏) represents the area
under the product curve (common area). this common area represents the convolution of
x(t) and h(t) for a shift of t=a.
6. Increase the time shift t and take the new interval whenever the function either x(𝜏) or
h(t-𝜏) changes. the value of t at which the change occurs defines the end of the current
interval and the beginning of a new interval. calculate y(t) using step5.
7. The value of convolution obtained at different values of t ( both positive and negative
values) may be plotted on a graph to get the combined convolution.

The above figure shows the plots of x( ) and h(t- ) together on the same time
axis. Here the signal h(t- ) is sketched for t<-3. x( ) and h(t- ) do not overlap. Therefore, the
product x( ) h(t- ) is equal to zero. y(t)=0 (for t<-3)
Plots of (a) x(𝜏), and (b)h(t-𝜏)when there is no overlap

Now, increase the time shift t until the signal h(t- ) intersects x( ). Figure below shows the
situation for t>-3. Here x( ) and h(t- ) overlapped.[ This overlapping continuous for all values for
t>-3 up to t=∞ because x( ) exists for all values of >0]. But x( )=0 to =t+3.
Plot of x(𝜏), and h(t-𝜏) with overlap
1.21 Convolution properties of Fourier transform:

With two functions h(t) and g(t), and their corresponding Fourier transforms H(f) and
G(f), we can form two special combinations – The convolution, denoted f = g * h, defined by
Convolution: g*h is a function of time, and g*h = h*g. The convolution is one member of a
transform pair g*h ↔ G(f) H(f)
The Fourier transform of the convolution is the product of the two Fourier transforms. This is
the Convolution Theorem.
Problems:
Find the convolution of the signals using Fourier transform.
1
2
3
4

2)

3)
4)

UNIT – II
Fourier series, Fourier Transforms, and Sampling
2.1. Fourier series representation of Periodic signals
Introduction:
A signal is said to be a continuous time signal if it is available at all instants of time. A
real time naturally available signal is in the form of time domain. However, the analysis of a signal
is far more convenient in the frequency domain. These are three important classes of
transformation methods available for continuous-time systems. They are:
1. Fourier series 2. Fourier Transform 3. Laplace Transform
Out of these three methods, the Fourier series is applicable only to periodic signals, i.e. signals
which repeat periodically over -∞ < t < ∞. Not all periodic signals can be represented by Fourier
series. Fourier series is to project periodic signals onto a set of basic functions. The basis
functions are orthogonal and any periodic signal can be written as a weighted sum of these basis
functions.
Representation of Fourier Series
The representation of signals over a certain interval of time in terms of the linear
combination of orthogonal functions is called Fourier Series. The Fourier analysis is also
sometimes called the harmonic analysis. Fourier series is applicable only for periodic signals. It
cannot be applied to non-periodic signals. A periodic signal is one which repeats itself at regular
intervals of time, i.e. periodically over -∞ to ∞. Three important classes of Fourier series methods
are available.
They are: 1. Trigonometric form 2. Cosine form 3. Exponential form
1) The function x(t) is absolutely integrable over one period, that is
2) The function x(t) has only a finite number of maxima and minima.
3) The function x(t) has a finite number of discontinuities.
2.3. Trigonometric Fourier series: A sinusoidal signal, x(t)=A sin ⍵0t is a periodic signal with
period T=2𝜋/⍵0. Also, the sum of two sinusoids is periodic provided that their frequencies are
integral multiples of a fundamental frequency ⍵0. We can show that a signal x(t), a sum of sine
and cosine functions whose frequencies are integral multiples of ⍵0, is a periodic signal.

The infinite series of sine and cosine terms of frequencies 0, ⍵0, 2⍵0, …… k⍵0 is known as
trigonometric form of Fourier series and can be written as:
where an and bn are constants; the coefficient a0 is called the dc component; a1cos⍵0t+b1sin⍵0t
the
first harmonic, a2cos2⍵0t+b2sin2⍵0t the second harmonic and ⍵0 + ⍵0 the nth
harmonic. The constant b0=0 because sin⍵0 = 0 for n=0.
2.4. Exponential Fourier series:

The exponential Fourier series is the most widely used form of Fourier series. In this,
the function x(t) is expressed as a weighted sum of the complex exponential functions. The
complex exponential form is more general and usually more convenient and more compact. So, it
is used almost exclusively, and it finds extensive application in communication theory.

Problems:

3) Obtain the exponential Fourier Series for the wave form shown in below figure

Solution: The periodic waveform shown in fig with a period T= 2π can be expressed as:
4) Find the exponential Fourier series for the full wave rectified sine wave given in below
figure.

2.5. Properties of continuous- time Fourier series

The Fourier series representation possesses a number of important properties
that are useful for various purposes during the transformation of signals from one form to other .
Some of the properties are listed below.
[x1(t) and x2(t)] are two periodic signals with period T and with Fourier
series coefficients Cn and Dn respectively.
1) Linearity property The linearity property states that, if x Cn and x Dn
then Ax1(t)+Bx +BDn

Proof: From the definition of Fourier series, we have
FS[Ax1(t)+Bx
𝐹
x(t - t0) − 𝜔0 0 C
n
=ACn+BDn
Ax1(t)+Bx +BDn
2) Time shifting property The time shifting property states that, if x(t) 𝐹 Cn then
x(t- Cn
x(t)=
x(t-
− 𝜔0
=FS-1[Cn ]

3) Time reversal property The time reversal property states that, if x(t) Cn
then x(- -n

x(t)=
x(- substituting n = -p in the right hand side, we get
x(-
substituting p = n , we get
x(- ]
x(- -n
x(-
4) Time scaling property The time scaling property states that, if x(t) Cn
then x(𝛼t) Cn with
ω0→𝛼ω0 Proof: From the definition of Fourier
series, we have
x(t)=
x( ]
where ω0→ ω0.
x(𝛼t)𝐹 Cn with fundamental frequency of 𝛼ω0
5) Time differential property: The time differential property states that, if x(t) 𝐹 Cn
then Cn
Proof: From the definition of Fourier series, we

have x(t)=
Differentiating both sides with
respect to t, we get
Cn
6) Time integration property: The time integration property states that, if x(t) n
then


x(t)=
Interchanging the order of integration and summation, we get
=
2.6. Complex Fourier Spectrum
The Fourier spectrum of a periodic signal x(t) is a plot of its Fourier coefficients
versus frequency ω. It is in two parts: (a) Amplitude spectrum and (b) phase spectrum. The plot
of the amplitude of Fourier coefficients verses frequency is known as the amplitude spectra,
and the plot of the phase of Fourier coefficients verses frequency is known as phase spectra.
The two plots together are known as Fourier frequency spectra of x(t).This type of
representation is also called frequency domain representation. The Fourier spectrum exists only
at discrete frequencies nωo, where n=0,1,2,….. Hence it is known as discrete spectrum or line
spectrum. The envelope of the spectrum depends only upon the pulse shape, but not upon the
period of repetition.
The below figure (a) represents the spectrum of a trigonometric Fourier series
extending from 0 to ∞, producing a one-sided spectrum as no negative frequencies exist here.
The figure (b) represents the spectrum of a complex exponential Fourier series extending from
- ∞ ∞, producing a two-sided spectrum. The amplitude spectrum of the exponential Fourier
series is symmetrical Fourier series is symmetrical about the vertical axis. This is true for all
periodic functions.
Fig: Complex frequency spectrum for (a) Trigonometric Fourier series and (b) complex
exponential Fourier series. If Cn is a general complex number, then
−𝜃
Cn = Cn & C-n = Cn & Cn = C-n

The magnitude spectrum is symmetrical about the vertical axis passing

through the origin, and the phase spectrum is antisymmetrical about the vertical axis passing
through the origin. So the magnitude spectrum exhibits even symmetry and phase spectrum
exhibits odd symmetry. When x(t) is real , then C- , the complex conjugate of Cn.
2.7. Fourier Transform
The Fourier transform is used to analyse aperiodic signals and can be use analyse
periodic signals also. So it over comes the limitations of Fourier series. Fourier transform is a
transformation technique which transforms signals from the continuous-time domain to the
corresponding frequency domain and vice versa, and which applies for both periodic as well as
aperiodic signals. Fourier transform can be developed by finding the Fourier series of a periodic
function and then tending T to infinity.
The Fourier transform is an extremely useful mathematical tool and is extensively

used in the analysis of linear time-invariant systems, cryptography, signal analysis, signal
processing, astronomy, etc. Several applications ranging from RADAR to spread spectrum
communication employ Fourier transform.
2.8. Derivation of the Fourier Transform of a non-periodic signal from the Fourier series
of a periodic signal
Let x(t) be a non-periodic function and, xT (t) be periodic with period T, and let their
relation is given by
x(t) = (t) The
Fourier series of a periodic signal x (t) is
jn t
Where and ω
Let nω0 = ω at T → . As T → , we have ω → 0 and the discrete Fourier spectrum

becomes continuos. Further, the summation becomes integral and xT(t) → x(t).
Thus, as T → ,
X(ω) =
Hence, X(ω) is called Fourier transform or the Fourier integral of x(t).

[n⍵0 = ω, T
x(t) =
T→∞
As T → becomes infinitesimally small and may be represented by dω.
Also the summation becomes integration.
Hence, x(t) is called the inverse Fourier transform of X(ω) .

The equations
X(ω) =
And
For
X(ω) and x(t) are known as Fourier transform pair and can be denoted as:
X(ω) = F
and x(t) =
The other notation that can be used to represent the Fourier transform pair is
x(t)
Magnitude and phase representation of Fourier transform

The magnitude and phase representation of the Fourier transform is the tool used to
analyse the transformed signal. In general, X( ) is a complex valued function of . Therefore,
X( ) can be written as: X( ) = XR( ) + jXI(⍵)
where XR( ) is real part of X( ) and XI(
2.9. Solved Problems:

Problem 1: Find the Fourier transform of x(t) = f(t-2)+f(t+2)
Solution:
Given x(t) = f(t-2)+f(t+2)
Using linearity property )], we have

F[x(t)] = F[f(t-2)+f(t+2)]
Using time shifting property [ i.e. x(t )] , we have

F[x(t)] = F[f(t)] − 2𝜔 + F[f(t)] 2𝜔 =
− 2𝜔 F(⍵)+ 2𝜔 F(⍵)
2𝜔 − 2𝜔
F(⍵)[ + ]
−
Problem 2:Find the Fourier transform of the signal ()
Solution:
Given x(t) = − ( )
We know that F[
− −
The signal x(t) = ( )is the time reversal of the signal (− ). Therefore, using time
reversal property [ i.e. x(- , we have
F[
∴
Problem 3: Find the Fourier transform of the signals cos ωot u(t)
Solution:
Given x(t) = cos ωot u(t)
i.e. u(t)
X( ) = F[cos ωot u(t)] = dt
With impulses of strength at ω = ωo and ω = −ωo
X(
Problem 4:Find the Fourier transform of the signals sin ωot u(t)
Solution: Given x(t) = sin ωot u(t)
i.e. u(t)

X( ) = F[sin ωot u(t)] = dt
With impulses of strength at ω = ωo and ω = −ωo
X(
Problem 5: Find the Fourier transform of the signals e−tsin5t u(t)

Solution:
Given x(t) = e−tsin 5t u(t)
x(t) = u(t)
X( ) = F[e−t sin 5t u(t)]
)u(t)] e−jωt dt
[neglecting impulses]
Problem 6: Find the Fourier transform of the signals e−2tcos 5t u(t)
Solution:
Given x(t) = e−2tcos 5t u(t)
x(t) = u(t)
X( ) = F[e−2t cos 5t u(t)]
)u(t)] e−jωt dt

Problem 7: Find the Fourier transform of the signals e− t sin 5 t for all t
Solution:
Given x(t) = e− t sin 5 t for all t
i.e. x(t) =
i.e. x(t) = − 5 (− ) + 5 ()
∴ X( )u(t)] e−jωt dt
Problem 8: Find the Fourier transform of the signals eat u(-t)

Solution:
Given x(t) = eat u(-t)
X( dt
dt = dt =
Problem 9: Find the Fourier transform of the signals teat u(t)
Solution:
Given x(t) = teat u(t)

X( dt
dt = dt = =

2.10. Fourier Transform of Standard Signals

Introduction:
• The Fourier transform of a finite duration signal can be found using the formula
This is called as analysis equation

• The inverse Fourier transform is given by
This is called as synthesis equation

Both these equations form the Fourier transform pair.
Existence of Fourier Transform:
The Fourier Transform does not exist for all aperiodic functions. The condition for a function
x(t) to have Fourier Transform, called Dirichlet conditions are:
1. is absolutely integrable over the interval -∞ to +∞,that is
2. has a finite number of discontinuities in every finite time interval. Further, each of
these discontinuities must be finite.
3. has a finite number of maxima and minima in every finite time interval.
1. Impulse Function
Given δ ,
Then
= e−jωt t=0 = 1
Hence , the Fourier Transform of a unit impulse function is unity.

w
w w
The impulse function with its magnitude and phase spectra are shown in below figure:

Similarly,
2. Single Sided Real exponential function 𝐞−𝐚 ( )

Given
Then
or
Now,
Figure shows the single-sided exponential function with its magnitude and phase spectra.

3.Double sided real exponential function 𝐞−𝐚

Given
And
A Two sided exponential function and its amplitude and phase spectra are shown in figures
below:
4. Constant Amplitude (1)

Let ∞ ≤t≤ ∞
The waveform of a constant function is shown in below figure .Let us consider a small section of
constant function, say, of duration .If we extend the small duration to infinity, we will get back
the original function.Therefore

Where
By definition, the Fourier transform of x(t) is:
X( ) = F[x(t)] = F
dt =
Using the sampling property of the delta function , we get
X(
5. Signum function (sgn(t))

The signum function is denoted by sgn(t) and is defined by
sgn(t) =
This function is not absolutely integrable. So we cannot directly find its Fourier transform.
Therefore, let us consider the function e−a t sgn(t) and substitute the limit a 0 to obtain the
above sgn(t)
Given x(t) = sgn(t) = sgn(t) =

X( ) = F[sgn(t)] = dt

and
Figure below shows the signum function and its magnitude and phase spectra
6. Unit step function u(t)

The unit step function is defined by
u(t)
since the unit step function is not absolutely integrable, we cannot directly find its Fourier
transform. So express the unit step function in terms of signum function as: u(t) =
x(t)= u(t) =
X( ) = F[u(t)] = F
We know that F[1] = 2𝜋𝛿(𝜔) and F[sgn(t)] =
F[u(t)]=
u(t)
∴ X(⍵) =∞ at ⍵=0 and is equal to 0 at ⍵=−∞ and ⍵=∞

7. Rectangular pulse ( Gate pulse) or rect

Consider a rectangular pulse as shown in below figure. This is called a unit gate function and is
defined as
x(t) = rect
Then X( ) = F[ x(t)] = F
∴ F , that is
rect
Figure shows the spectra of the gate function
8. Triangular Pulse
Consider the triangular pulse as shown in below figure. It is defined as:

i.e. as x(t) =
Then X( ) = F[ x(t)] = F
Or
Figure shows the amplitude spectrum of a triangular pulse.

2.11 Fourier Transform of Periodic Signal

The periodic functions can be analysed using Fourier series and that non-periodic function can
be analysed using Fourier transform. But we can find the Fourier transform of a periodic
function also. This means that the Fourier transform can be used as a universal mathematical tool
in the analysis of both non-periodic and periodic waveforms over the entire interval. Fourier
transform of periodic functions may be found using the concept of impulse function.
We know that using Fourier series , any periodic signal can be represented as a sum of complex
exponentials. Therefore, we can represent a periodic signal using the Fourier integral. Let us
consider a periodic signal x(t) with period T. Then, we can express x(t) in terms of exponential
Fourier series as:
x(t) =
The Fourier transform of x(t) is:
X( ) = F[x(t)] = F
Using the frequency shifting theorem, we have
= =s
X(
Where 𝐶 are the Fourier coefficients associated with x(t) and are given by
Thus, the Fourier transform of a periodic function consists of a train of equally spaced impulses.
These impulses are located at the harmonic frequencies of the signal and the strength of each
impulse is given as 2 .

Solved Problems:
Problem 1:Find the Fourier transform of the signals e3tu(t)
Solution:
Given x(t) = e3tu(t)
The given signal is not absolutely integrable.
That is .
Therefore, Fourier transform of x(t) = e3tu(t) does not exist.
Problem 2: Find the Fourier transform of the signals cosωotu(t)

Solution:
Given x(t) = cosωot u(t)
i.e. u(t)
X( ) = F[cosωot u(t)] = dt
With impulses of strength at ω=ωo and ω=−ωo
X(
Problem 3: Find the Fourier transform of the signals sinωot u(t)
Solution:
Given x(t) = sinωot u(t)
i.e. u(t)

X( ) = F[sinωot u(t)] = dt
With impulses of strength at ω=ωo and ω=−ωo
X(
Problem 4: Find the Fourier transform of the signals e−tsin5t u(t)

Solution:
Given x(t) = e−tsin5t u(t)
x(t) = u(t)
X( ) = F[e−t sin5t u(t)]
)u(t)] e−jωt dt
Problem 5: Find the Fourier transform of the signals e−2tcos5t u(t)
Solution:
Given x(t) = e−2tcos5t u(t)
x(t) = u(t)
X( ) = F[e−2t cos5t u(t)]

)u(t)] e−jωt dt
2.12. Properties of the Fourier Transform
These properties provides significant amount of insight into the transform

and into the relationship between the time-domain and frequency domain descriptions
of a signal. Many of these properties are useful in reducing the complexity Fourier
transforms or inverse transforms.
Linearity
If x ( t) f X (jw)
y (t) f Y (jw)
Then
a x(t) + b y(t) f (aX (jw) + Y b (jw))
Time Shifting
If x (t) I ( X (jw)
Then
x (t - t0) - jwt 0
( fX (jw)e
To establish this property, consider x(t)=1/2π∫-∞ X(jw)e-jwtO dw
Replacing t by t-to in this equation, we obtain

x(t-to)=1/2π∫-∞ X(jw)e-jw(t-to)dw
x(t)=1/2π∫-∞e-jwto X(jw)ejwt dw
Recognizing this as the synthesis equation for x(t-to) ,we conclude that
F{x(t-to)}= e-jwto X(jw)

Conjugation and Conjugate symmetry
The conjugation property states that if
x (t) fX(jw) Then
x* (t) f X*(-jw) ……………………………………………………(i)
This property follows from the evaluation of the complex conjugate
-jwt
X*(-jw) = [ x(t) dt ]*
∞
= ∫-∞ x*(t)ejwt dt.
Replacing w by –w, we see that
∞
X*(-jw) = ∫-∞ x*(t)ejwt dt. ………………………………………….(ii)
The conjugate property allows us to show that if x(t) is real ,then X(jw) has conjugate
symmetry: that is
X(-jw)=X*(jw)
x(t) real]
…………………………………..(iii)
If x(t) is real so that x*(t) = x(t), we have ,from eq.(ii)

X*(-jw)= ∫-∞ x*(t)ejwt dt = X(jw). Follows by replacing w by –w
Differentiation and Integration

If x (t) f X(jw)
then differentiating both sides of the Fourier transform synthesis equation we have
∞
dx(t)/dt=1/2π∫-∞jwX(jw)e-jwto dw Therefore,
dx(t)/dt f jwX (jw)
This important property replaces the operation of the differentiation in time domain
with that of multiplication by jw in the frequency domain similarly integration should involve
division by jwin frequency domain.

t
)+ π X(0 ) δ w )
∫-∞ x(t)dt f ( 1/ jw(X( jw)
)
The impulse term on the right-hand side above equation reflects the dc or average value that
can result from integration.
Time and Frequency Scaling

If x (t) fX(jw)
Then x(at) f1/|a|X(jw/a) ………………………………(v)
Where a is real constant. This property follows directly from the

definition of the Fourier transform If a = -1 we have,
x (t) f X ( -jw)
-
2.13. Sampling Theorem

Statement of the sampling theorem
 A band limited signal of finite energy , which has no frequency components higher than W
hertz , is completely described by specifying the values of the signal at instants of time
separated by 1/2W seconds and
 A band limited signal of finite energy, which has no frequency components higher than W
hertz, may be completely recovered from the knowledge of its samples taken at the rate of
2W samples per second.
The first part of above statement tells about sampling of the signal and second part tells about
reconstruction of the signal. Above statement can be combined and stated alternately as follows:
A continuous time signal can be completely represented into samples and recovered back if the
sampling frequency is twice of the highest frequency content of the signal i.e., fs≥2W Here fs is
the sampling frequency and W is the higher frequency content.
Proof of sampling theorem

There are two parts:
I) Representation of x(t) in terms of its samples
II) Reconstruction of x(t) from its samples
PART I: Representation of x(t) in its samples x(nTs)

Step 1: Define xδ(t)
Step 2 : Fourier transform of xδ(t) i.e. Xδ(f)

Step 3: Relation between X(f) and Xδ(f)

Step 4 : Relation between x(t) and x(nTs)
Step 1: Define xδ(t)
The sampled signal xδ(t) is given as ,
----------- (1)
Here, observe that xδ(t) is the product of x(t) and impulse train δ(t) as shown in figure.
In the above equation δ(t-nTs) indicates the samples placed at ±Ts,±2Ts,±3Ts… and so on
Step 2: Fourier transform of xδ(t)i.e. Xδ(f)

Taking FT of equation (1)
---------- (2)
We know that FT of product in time domain becomes convolution in frequency domain i.e.,
------------ (3)

Conclusions:
Figure 1.10 Spectrum of Original Signal & Sampled Signal
Step 3: Relation between X(f) and Xδ(f)

Important assumption
Let us assume that fs=2W , then as per above diagram
---------- (4)
Step 3: Relation between x(t) and x(nTs): From DTFT,
------------ (5)

Putting above expression in equation (4)
------------ (6)
Conclusions:
 Here x(t) is represented completely in terms of x(nTs)
 Above equation holds for fs=2W.This means if the samples are taken at the
rate of 2W or higher, x(t) is completely represented by its samples.
 First part of the sampling theorem is proved by above two conclusions.
II) Reconstruction of x(t)from its samples
Step 1 : Take inverse Fourier transform of X(f) which is in terms of Xδ(f)
Step 2 : Show that x(t) is obtained back with the help of interpolation function.
Step 1 : Take inverse Fourier transform of equation (6) becomes ,

Step 2: Let us interpret the above equation and exapanding we get,

Figure 1.11 Sampled Version of Signal & Reconstruction of x(t) from its samples
Conclusions:
The samples x(nTs(nTs)are weighted by sinc functions.
The sinc function is the interpolating function above figure shows, how x(t) is interpolated.
Step 3: Reconstruction of x(t) by low pass filter
When the interpolated signal of equation (6) is passed through the low pass filter of bandwidth -
W≤f≤W , then the reconstructed waveform shown in figure is obtained. The individual sinc
functions are interpolated to get smooth x(t).
2.14. Aliasing
When high frequency interferes with low frequency and appears as low frequency, then the
phenomenon is called aliasing.
Figure 1.12 Effects of under sampling or aliasing

Effects of aliasing:
 Since high and low frequencies interfere with each other, distortion is generated.
 The data is lost and it cannot be recovered.
Different ways to avoid aliasing:
Aliasing can be avoided by two methods
 Sampling rate fs≥2W
 Strictly band limit the signal to ‘W’
Figure 1.13 Methods to avoid aliasing
2.15. Nyquist Rate & Nyquist Interval

UNIT – III
Laplace Transforms and Z–Transforms
3.1 Introduction:
Laplace Transform is a powerful tool for analysis and design of Continuous Time
signals and systems. The Laplace Transform differs from Fourier Transform because it covers a
broader class of CT signals and systems which may or may not be stable. Most of the LTI Systems
act in time domain but they are more clearly described in the frequency domain instead. It is
important to understand Fourier analysis in solving many problems involving signals and LTI
systems. Now, we shall deal with signals and systems which do not have a Fourier transform.
We found that continuous-time Fourier transform is a tool to represent signals as linear
combinations of complex exponentials. The exponentials are of the form est with = 𝜔and 𝜔 is
an eigen function of the LTI system. Also, we note that the Fourier Transform only exists for
signals which can absolutely integrated and have a finite energy. This observation leads to
generalization of continuous-time Fourier transform by considering a broader class of signals
using the powerful tool of "Laplace transform".
Bilateral Laplace Transform

The Laplace transform of a general signal x(t) is defined as
It is a function of complex variable„s‟ and is written as =𝜎+ 𝜔, with and , the real and
imaginary parts, respectively.
The transform relationship between x(t) and X(s) is indicated as
Existence of Laplace Transform

In general,
The ROC consists of those values of „s‟ (i.e., those points in the s-plane) for which X(s)
converges i.e., value of s for which
Since =𝜎+ 𝜔 the condition for existence is
Thus, ROC of the Laplace transform of an x(t) consists of all values of s for which x(t) e- t is
absolutely integrable.

3.2. Relation between Laplace and Fourier transform
When corresponds to the Fourier transform of

x(t), i.e., . The Laplace transform also bears a straight forward
relationship to the Fourier transform when the complex variable „s‟ is not purely imaginary. To
see this relationship, consider X(s) with
or
−𝜎
The real exponential may be decaying or growing in time, depending on being positive or
negative.
Unilateral Laplace Transform This Transform have considerable value in analyzing causal
systems and particularly, systems specified by linear constant coefficient differential equations
with nonzero initial conditions( i.e., systems that are not initially at rest)
The Unilateral Laplace transform of a continuous time signal x(t) is defined as
Problem 1: Find the Laplace transform of ( )= − ( )

Solution:
The Fourier transform X(jω) converges for a>0 and is given by
Now, the Laplace transform is
with or equivalently, since = 𝜎+ 𝜔 and 𝜎 = { },
, Re{s} > -a
That is ,Re{s} > -a.

For example a=0, x(t) is the unit step with Laplace transform , Re{s} > 0
Problem 2: Find the Laplace transform of

Solution: The Laplace transform is
This result converges only when Re{s+a} <0, or Re{s}<-a
That is ,Re{s} < -a
Comparing the two results in the above two problems, we see that Laplace transform is
identical for both the signals. But the range of values of „s‟ for which the transform converges is
different i.e., Region of Convergence (ROC) is different for the above said signals. Note: same
X(s) may correspond to different x(t) depending on ROC
Problem 3: Find the Laplace transform of ( )=𝛿( ). What is the region of convergence?
Solution: The Laplace transform of a general signal x(t) is defined as
From the property of impulse function

As we know that area under impulse function is unity.

Since , which is a constant, Laplace transform converges for all values of s i.e., ROC
is entire s-plane
Problem 4: What is the Laplace transform of What is its Region of
Convergence?
Solution: The Laplace transform of a general signal x(t) is defined as
From Euler‟s relation
− ( +1)
Problem 5: Find the unilateral Laplace transform of ( )= ( +1)
Solution: The Unilateral Laplace transform is
ROC: Re{s}>-a
Problem 6: Determine the Laplace transform of the ramp function.

Solution: The unit ramp function is given as
or ( )= ()
The Laplace transform of a general signal x(t) is defined as

With ROC: Re{s}>0
−
Problem 7: A damped sine wave is given by ( )= (𝜔 ). Find the LT of this signal
Solution: With the help of Euler‟s Identity,
Applying the Laplace transform
ROC : Re{s} > -a
Problem 8: Find the Transfer function of a system with impulse response
Solution: Transfer function is obtained by applying Laplace Transform to the impulse response
h(t)
Problem 9: Find the Laplace transform of

Solution: With the help of Euler‟s Identity,
Therefore,
Problem 10: Find the Unilateral Laplace transform of

Solution:
The Unilateral Laplace transform of a general signal x(t) is defined as

3.3 Region of Convergence (ROC) of LT

The Laplace transform of a continuous signal x(t) is given by
The Laplace transform has two parts which are, the expression and Region of Convergence
respectively.
Whether the Laplace transform X(s) of a signal x(t) exists or not depends on the complex
variable „s‟ as well as the signal itself. All complex values of „s‟ for which the integral in the
definition converges form a region of convergence (ROC) in the s-plane
The concept of ROC can be understood easily by finding Laplace transform of two functions
given below:
a)
with =𝜎+ 𝜔 the integral converges only when , i.e.,
therefore . The ROC is shown in figure below.
b)

with =𝜎+ 𝜔 the integral converges only when , i.e.,

therefore . The ROC is shown in figure below.
The plane in which ROC is shown is known as s-plane.As =𝜎+ 𝜔, s-plane consists of real and
imaginary axes. The region towards left side of the imaginary axis is called Left Half Plane and
towards right is called Right Half Plane.
Zeros and Poles of the Laplace Transform
Laplace transforms in the above examples are rational, i.e., they can be written as a ratio of
polynomials of variable „s‟in the general form
• N(s) is the numerator polynomial of order M withszk,(k=1,2,…,M) roots

• D(s) is the denominator polynomial of order N with spk(k=1,2,…,N) roots
Roots of numerator polynomial are called zeros and the roots of denominator polynomial are
called poles. Poles in s-plane are indicated with „x‟ and zeros with‟o‟. The representation of
X(s) through its poles and zeros in the s-plane is referred to as the pole-zero plot of X(s).
In general, we assume the order of the numerator polynomial is always lower than that of the
denominator polynomial, i.e., M<N. If this is not the case, we can always expand X(s) into
multiple terms so that M<N is true for each of terms.

3.4. Properties of ROC:

Property 1: The ROC of X(s) consists of strips parallel to the jω-axis in the s-plane.
Property 2: For rational Laplace transforms, the ROC does not contain any poles but it is
bounded by poles or extends to infinity.
Property 3: If x(t) is of finite duration and is absolutely integrable, then the ROC is the entire s-
plane.
Property 4: If x(t) is right sided, and if the line Re{s}= o is in the ROC, then all values of s for
which Re{s}> o will also be in the ROC.
Property 5: If x(t) is left sided, and if the line Re{s}= o is in the ROC, then all values of s for
which Re{s}< o will also be in the ROC.
Property 6: If x(t) is two sided, and if the line Re{s}= o is in the ROC, then the ROC consist of
a strip in the s-plane that includes the line Re{s}= o.
Property 7: Ifx(t) is right sided and its Laplace transform X(s) is rational, then the ROC in s-
plane is right of the rightmost pole.
Property 8: If x(t) is left sided and its Laplace transform X(s) is rational, then the ROC in
s-plane is left of the leftmost pole.
Solved Problems:
Problem 1: Find the Laplace transform of x(t) = [2e-2t+3e-3t]u(t).Also indicate locations of poles
and zeros and Plot Region of Convergence.
Solution:The signal given x(t) = [2e-2t+3e-3t]u(t) is a right-sided signal and its Laplace
transform is with ROC: Re{s}>-2 ∩ Re{s}>-3 = Re{s}>-2
ROC is right of the right most pole (Property 7) and the plot is shown below
Problem 2: Find the Laplace transform of x(t) = [2e2t+3e3t]u(-t).Also indicate locations of poles
and zeros and Plot Region of Convergence.

Solution:The signal given x(t) = [2e2t+3e3t]u(-t) is a left-sided signal and its Laplace transform is
with ROC: Re{s}<2 ∩ Re{s}<3 = Re{s}<2
ROC is left of the left most pole (Property 8) and the plot is shown below
Problem 3: Find the Laplace transform of x(t) = e-tu(t)+e2tu(-t).Also indicate locations of

poles and zeros and Plot Region of Convergence.
Solution:The signal given x(t) = e-tu(t)+e2tu(-t) is a two-sided signal and its Laplace transform is
with ROC: Re{s}>-1 ∩ Re{s}<2 = -1<Re{s}<2
ROC is strip in the s-plane and lies to the right of the pole at s=-1 and to the left of the pole at
s=2. Theplot is shown below
3.5. Properties of Laplace Transform
1. Linearity of the Laplace Transform Statement:
If with a region of convergence denoted as R1 and
with a region of convergence denoted as R2
then , with ROC containing 1∩ 2
Proof:

Consider the linear combination of two signals x1(t) and x2(t) as z(t)=ax1(t)+bx2(t). Now, take the
Laplace transform of z(t) as
2. Time Shifting: If with ROC= R
then with ROC= R

Proof:
Let t- =p
3. Shifting in s-Domain: If with ROC= R then with ROC=

R+Re{so}
Proof:
4. Time Scaling: If with ROC= R then with ROC= R1=aR

Proof:
To prove this we have to consider two cases: a (real) is positive and a is negative. Case 1: For
a>0:
Using the substitution of λ=at; dt=adλ

Case 2: For a< 0 :
Using the substitution of λ=at; dt=adλ
Combining the two cases, we get with ROC= R1=aR
5. Convolution Property: If with ROC = R1 and with ROC = R2

then , with ROC containing 1∩ 2
Proof:
Interchanging the order of integrations
(Since from Time

shifting property)
6. Differentiation in the Time Domain: If with ROC= then with

ROC containing R
Proof: Inverse Laplace transform is given by
Differentiating above on both sides with respect to „t‟

Comparing both equations is the Laplace transform of .
7. Integration in the Time Domain: If with ROC= R
then with ROC containing
Proof: This can be derived using convolution property as
8. The Initial and Final Value Theorem:
If x(t) and are Laplace transformable, and under the specific constraints that x(t)=0 for t<0
containing no impulses at the origin,one can directly calculate, from the Laplace transform, the
initial value x(0+), i.e., x(t) as t approaches zero from positive values of t. Specifically the initial
-value theorem states that
Also, if x(t)=0 for t<0 and, in addition, x(t) has a finite limit as t→∞, then the final value
theorem says that
Proof:
To prove these theorems, we need to consider the Unilateral Laplace transform of
Unilateral Laplace transform of
Initial value theorem

From the above discussion, we know that
Applying the lim →∞ ℎ

Final value theorem we know that
Applying the lim →0 ℎ
Solved Problems:
Problem 1: Find the Laplace transform and ROC of
Solution:
We know that
Signal is now delayed by 2 units to get

Therefore, applying time shifting property
Problem 2: Given , find f(0)
Solution:
Consider
From the initial value theorem, we know that

3.6. Inverse Laplace Transform

Inverse Laplace transform maps a function in s-domain back to the time domain. One
application is to convert a system response to an input signal from s-domain back to the time
domain. The Laplace transform converts the differential equations that describe system behaviour
to a polynomial. Also the convolution operation which describes the system action on the input
signals is converted to a multiplication operation. These two properties make it much easier to do
systems analysis in the s-domain. Inverse Laplace transform is performed using Partial Fraction
Expansion that split up a complicated fraction into forms that are in the Laplace Transform table.
The Laplace transform of a continuous signal x(t) is given by
Since s= +jω
We can recover x(t) from its Laplace transform evaluated for a set of values of s= +jω
in the ROC, with fixed and ω varying from -∞ to +∞. Recovering s(t) from X(s) is done by
changing the variable of integration in the above equation from ω to s and using the fact that is
constant, so that ds=jdω.
The contour of integration in above equation is a straight line in the s-plane corresponding to all
points s satisfying Re{s}= . This line is parallel to the jω-axis. Therefore, we can choose any
value of such that converges.
3.7. Partial Fraction Expansion
As we know that the rational form of X(s) can be expanded into partial fractions, Inverse
Laplace transform can be taken according to location of poles and ROC of X(s). The roots of
denominator polynomial, i.e., poles can be simple and real, complex or multiple.
We know that X(s) is expanded in partial fractions as
Here the roots s0,s1,s2,...sn can be real, complex or multiple. Then the values of
k0,k1,k2,...kn constants are calculated accordingly.
In order to find the appropriate time domain function, ROC should be indicated for the s-
domain function. Otherwise we may have multiple time-domain functions based on different
possible ROCs.

Example for Real roots:

Problem 1:Find out the partial fraction expansion and hence Inverse Laplace transform of the
function , ROC: Re{s} > 3
Solution: The function can be written as,
The constants calculated are A=1/3, B=-1/5, C=13/15
From the given ROC: Re{s}>3, the resultant signal x(t) should be right sided.
Therefore,
Example for Complex roots:
Problem 2: Obtain right sided time domain signal for the function
Solution: We can write the given function as
The constants can be calculated as A=1/8, B=0.874, C=0.5
Therefore,
Finally
i.e.,
Example for Multiple roots:

Problem 3: Find out the inverse Laplace transform of , ROC:Re{s}<-1
Solution: We can write the given function as
The constants can be calculated as A=3, B=2, C=2, D=-2

Therefore,
From the given ROC: Re{s}<-1, the resultant signal x(t) should be left sided.
Finally,

From the result of Laplace transform
i.e.
3.8. Laplace transform using Waveform Synthesis

In waveform synthesis, the unit step function u(t) and other functions serve as
building blocks in constructing other waveforms. Once the waveforms are synthesized in the
form of other functions, Laplace transform is found and simplified.
For example, we may describe a pulse waveform in terms of unit step functions. A pulse of unit
amplitude from t=a to t=b can be formed by taking the difference between the two step
functions
i.e., x(t) = u(t -a) - u(t -b)
Hence
Examples:
Solved Problems:
Problem 1: Find the Inverse Laplace transform of ROC: Re{s}>3
Solution:
We know that
Writing X(s) in the form of partial fraction expansion
The constants can be calculated as A=4, B=-1

Therefore,
From the given ROC: Re{s}>3, the resultant signal x(t) should be right sided.
i.e.,

Problem 2: Find the Inverse Laplace transform of ROC: Re{s}<-2
Solution:
We know that
Therefore,
From the given ROC: Re{s}<-2, the resultant signal x(t) should be left sided.
i.e.,
Problem 3: Find the Inverse Laplace transform of ROC: -2<Re{s}<-1
Solution: We know that
Therefore,
From the given ROC: -2<Re{s}<-1, the two derived conditions are Re{s}<-1 which suits for
and Re{s}>-2 which suits for . Therefore, the resultant signal x(t) should be two-
sided.
i.e.,

PART - II
Z–Transforms
3.9 Introduction:
Digital signals are discrete in both time (the independent variable) and amplitude (the
dependent variable). Signals that are discrete in time but continuous in amplitude are referred to
as discrete-time signals.
Z Transform is a powerful tool for analysis and design of Discrete Time signals and
systems. The Z Transform differs from Fourier Transform because it covers a broader class of DT
signals and systems which may or may not be stable. Fourier Transform only exists for signals
which can absolutely integrated and have a finite energy. Z-transforms is a generalization of
Discrete-time Fourier transform by considering a broader class of signals.
Existence of z Transform
In general,
The ROC consists of those values of „z‟ (i.e., those points in the z-plane) for which X(z)
converges i.e., value of z for which
Since = 𝜔 the condition for existence is
−𝜔
Since =1
Therefore, the condition for which z-transform exists and converges is

Thus, ROC of the z transform of an x(n) consists of all values of z for which is
absolutely summable.
3.10. Relation between ZT and Discrete Time Fourier transform
When corresponds to the Discrete Time

Fourier transform (DTFT) of x(n), i.e., . Thez transform also bears a
straight forward relationship to the DTFT when the complex variable = . To see this
relationship, consider X(z) with

Unilateral Z Transform have considerable value in analyzing causal systems and particularly,
systems specified by linear constant coefficient difference equations with non-zero initial
conditions( i.e., systems that are not initially at rest).
The Unilateral z- transform of a discrete time signal x(n) is defined as
3.11. Relation between Laplace, Fourier and z- transforms

Let x(t) be a continuous signal sampled with a sampling time of T units. Call this sampled signal
as xs(t). We represent this sampled signal by
Applying the Laplace transform to xs(t) results
Interchanging the order of integration and summation
For uniform sampling x(kT)≡x(k)

Then
Comparing this with the z-transform formula
We get a relation that z=esT
3.12. Problems
Problem 1. Finding the z-transform of
a)

For convergence of X(z), we require that . Consequently, the region of

−1
convergence is that range of values of z for which <1, or equivalently, > and is
shown in figure below
Then
b)
This result converges only when or equivalently, |z| < |a|. The ROC is shown below
If we consider the signals anu(n) and -anu(-n-1), we note that although the signals are
differing, their z Transforms are identical which is .Thus, we conclude that to distinguish z-
Transforms uniquely their ROC's must be specified.
Problem 2: Find the z-transform of ( )=𝛿( ). What is the region of convergence?

Solution:
The z transform of a general signal x( ) is defined as

Since
Since , which is a constant, the result of z transform converges for all values of „z‟,
i.e., ROC is entire z-plane
Problem 3: Determine the z-transform of
Solution:
Therefore,
Problem 4: Find the z-transform of

Solution:
We know that
AlsoWe know that
Therefore,
ROC: 2<|z|<3
Problem 5: Find the two sided z-transform of the signal
Solution:
Here x(n) can be written as
Therefore, the z-transform of x(n) is

ROC: 1/3 < |z| < 2
3.13. Region of Convergence (ROC) of Z-Transforms

The Z- transform of a discrete signal x(n) is given by
The Z-transform has two parts which are, the expression and Region of Convergence
respectively.
Whether the Z-transform X(z) of a signal x(n) exists or not depends on the complex variable „z‟
as well as the signal itself. All complex values of „z=rejω‟ for which the summation in the
definition converges form a region of convergence (ROC) in the z-plane. A circle with r=1 is
called unit circle and the complex variable in z-plane is represented as shown below.
Description :
The concept of ROC can be understood easily by finding z transform of two functions given
below:
a)
For convergence of X(z), we require that . Consequently, the region of

−1
convergence is that range of values of z for which <1, or equivalently, > and is
shown in figure below

Then
b)
This result converges only when or equivalently, |z| < |a|. The ROC is shown below
3.14. Properties of ROC
Property 1: The ROC of X(z) consists of a ring in the z-plane centered about the origin.
Property 2: If the z-transform X(z) of x(n) is rational, then the ROC does not contain any poles
but is bounded by poles or extend to infinity.
Property 3: If x(n) is of finite duration, then the ROC is the entire z-plane, except possibly z=0
and / or z=∞
Property 4: If x(nt) is a right sided sequence, and if the circle |z|=ro is in the ROC, then all finite
values of z for which |z|>ro will also be in the ROC.

Property 5: If x(n) is a left sided sequence, and if the circle|z|=ro is in the ROC, then all values
of z for which 0<|z|<ro will also be in the ROC.
Property 6: If x(n) is two sided, and if the circle |z|=ro is in the ROC, then the ROC will consist
of a ring in the z-plane that includes the circle |z|=ro.
Property 7: Ifthe z-transform X(z) of x(n) is rational, and if x(n) is right sided, then the ROC is
the region in the z-plane outside the outermost pole i.e., outside the circle of radius equal to the
largest magnitude of the poles of X(z).
Property 8:If the z-transform X(z) of x(n) is rational, and if x(n) is left sided, then the ROC is
the region in the z-plane inside the innermost pole i.e., inside the circle of radius equal to the
smallest magnitude of the poles of X(z) other than any at z=0 and extending inward to and
possibly including z=0.
Solved Problems:
Problem 1: Find the Z-transform and plot the ROC of
Solution:
Given signal is right sided
We know that
Therefore, {shown in figure a} and
{Shown in figure b}
For convergence of X(z), both sums must converge, which requires that the ROC should be
an intersection of and . i.e., {shown in figure c}
The pole zero plot and ROC are shown in the figure below

Problem 2:Draw the pole-zero plot and graph the frequency response for the system whose
transfer function is . Is the system both causal and stable?
Solution:
The transfer function can be factored into
The Pole-zero diagram is shown below
The magnitude and phase frequency responses of the system are obtained by substituting
z=ejωand varying the frequency variable ω over a range of 2 . The frequency response plots are
illustrated in Figure below
The system is both causal and stable because all the poles are inside the unit circle
Problem 3: For the following algebraic expression for the z-transform of a signal, determine the
number of zeros in the finite z-plane and the number of zeros at infinity
Solution:
The given z-transform may be written as
Clearly, X(z) has a zero at z=1/2. Since the order of the denominator polynomial exceeds the
order of the numerator polynomial by 1, X(z) has a zero at infinity. Therefore, X(z) has one zero
in the finite z-plane and one zero at infinity.

3.15. Properties of Z-Transform

The z transform of a discrete signal x(n) is given by
And inverse z transform is given by
1. Linearity:
If with ROC = R1 and with ROC = R2 then
, with ROC containing 1∩ 2
Proof:
Taking the z-transform
2. Time Shifting:
If with ROC= R
then with ROC= R, except for the possible addition or deletion of

the origin or infinity Proof:
Let n-m=p
3. Scaling in the z-Domain:

If with ROC= R
then with ROC= |zo|R where, |zo|R is the scaled version of R.
Proof:

4. Time Reversal:
If with ROC= R then
with ROC=
Proof:
Let -n=p
5. The Initial Value Theorem:

If x(n)=0, for n < 0 then initial value of x(n) i.e., Proof:
We know that Z{x(n)}= as x(n) is causal.
Expanding the summation
Applying the lim →∞ ฀
i.e.,
6. The Final Value Theorem:

If x(n) is causal and X(z) is the z-transform of x(n) and if all the poles of X(z) lie
strictly inside the unit circle except possibly for a first order pole at z=1 then
Proof:
Consider the z-transform of x(n)-x(n-1)

Also, the above can be written as
Applying the limit z→1 on both sides
LHS after applying the limit z→1 becomes
All terms cancel except x(N). Therefore,
7.Differentiation in the z-Domain:

If with ROC= R then
with ROC = R
Proof: z transform is given by
Differentiating above on both sides with respect to „z‟
Comparing both equations is the z transform of ( ).
3.16. Inverse Z-Transform

Inverse z-transform maps a function in z-domain back to the time domain. One
application is to convert a discrete system response to an input sequence from z-domain back to
the time domain. The z-transform converts the Linear Constant Coefficient Difference
Equations(LCCDE) that describe system behaviour to a polynomial. Also the convolution
operation which describes the system action on the input signals is converted to a multiplication
operation. These two properties make it much easier to do systems analysis in the z-domain.
Inverse z-transform is performed using Long Division Method(Power Series Expansion
method), Partial Fraction Expansion and Residue method (Contour Integral Method).

The z- transform of a discrete signal x(n) is given by
Since z=rejω
Long Division Method (Power Series Expansion)

Z-transform of the sequence x(n) is given as,
From above expansion of z-transform, the sequence x(n) can be obtained as,
The Power series expansion can be obtained directly or by long division method.
Example: Determine inverse z-transform of the following:
i) ii)

Partial Fraction Expansion
As we know that the rational form of X(z) can be expanded into partial fractions, Inverse
z-transform can be taken according to location of poles and ROC of X(z).
Following steps are to be performed for partial fraction expansions:
Step 1: Arrange the given X(z) as,
Step 2:
Where Ak for k=1, 2,…N are the constants to be found in partial fractions. Poles may be of
multiple order. The coefficients will be calculated accordingly.

Step 3: Above equation can be written as
Step 4: All the terms in above step are of the form . Depending upon ROC, following
standard pairs must be used.
with ROC: |z| >| |,i.e., causal response
with ROC: |z| <| |,i.e., non-causal response
Example: Determine the inverse z-transform of
For ROC i) |z| > 1 ii)|z| < 0.5 iii)0.5 < |z| < 1
Solution:
Given
Which can be written as
After finding the constants as A=2 and B=-1
i) For ROC |z| > 1 i.e., causal or right sided
ii) For ROC |z| < 0.5 i.e., non-causal or left sided

iii) For ROC 0.5 < |z| < 1 i.e., two sided
The ROC is a circular strip between z>0.5 and z<1

Residue Method (Contour Integral Method)
Cauchy integral theorem is used to calculate inverse z-transform. Following steps are to be
followed:
−1
Step 1: Define the function ( )= ( ) , which is rational and its denominator is expanded into
product of poles.
Here „m‟ is order of the pole

Step 2: For poles of order „m‟, the residue of can be calculated as,
Step 3:
i) Using residue theorem, calculate x(n) for poles inside the unit
circle, i.e.,
ii) For poles outside the contour of integration,
with n < 0
Example: Determine the inverse z-transform of ,
ROC : |z| > |a| using contour integration

Solution:
Step 1:

Step 2: Here the pole is at z=a and it has order m=2. Hence finding the residues at z=a
Step 3: The sequence x(n) is given as
Since ROC: >| |
Examples:
Solved Problems:
Problem 1: Find the inverse z-transform of , |z|>|a|
Solution:
log(1-p) is expanded with the help of power series. It is given as
Therefore,
for |z| > |a|
Hence
i.e.,
Problem 2: Find the inverse z-transform of using partial fraction method when
ROC is
(i) |z| <- 2

(ii) |z| > 3 (iii) -2 < |z| < 3

Solution: Given
Applying partial fractions
i) For ROC |z| < -2
ii) For ROC |z| > 3
iii) For ROC -2 < |z| <3
Problem 3: Determine the sequence whose z-transform is,
ROC : |z|> 1
Solution: To arrange X(z) in proper form suitable for partial fraction expansion
The highest power of denominator polynomial should be atleast one less than that of numerator
polynomial
Let us arrange X(z) as follows:
Now perform one step division so that order of numerator polynomial is reduced by one unit.
Now

Writing in partial fractions
Finding the constants A =8 and B=-9
As ROC is |z| > 1, representing the signal as causal signal which is right sided
Problem 4: An LTI system is characterized by the system function
Specify the ROC of H(z) and determine h(n) for the following conditions
a) The system is causal and unstable
b) The system is non-causal and stable
c) The system is non-causal and unstable
Solution: Given that
Using partial fraction expansion, we obtain
Finding the constants A =1 and B=2
The system has poles at z=0.5 and z=3

a) For the system to be causal and unstable, the ROC of H(z) is the region in the z-plane
outside the outermost pole and it must not include the unit circle. Therefore, the ROC is
the region |z| > 3

Hence
b) For the system to be non-causal and stable, the ROC of H(z) is the ring in the z-plane and
it must include the unit circle. Therefore, the ROC is the region, 0.5 < |z| < 3
Hence
c) For the system to be non-causal and unstable, the ROC of H(z) is the ring in the zplane
inside the inner most pole and it must not include the unit circle. Therefore, the ROC is
the region, |z| < 0.5
Hence

UNIT – IV
Random Processes – Temporal Characteristics
4.1. Introduction
A Random Variable ‘X’ is defined as a function of the possible outcome ‘s’ of an
experiment or whose value is unknown and possibly depends on a set of random events. It is
denoted by X(s). The Concept of Random Process is based on enlarging the random variable
concept to include time ‘t’ and is denoted by X(t,s) i.e., we assign a time function to every outcome
according to some rule. In short, it is represented as X(t). A random process clearly represents a
family or ensemble of time functions when t and s are variables. Each member time function is
called a sample function or ensemble member. Depending on time ‘t’ and outcome ’s’ fixed or
variable. A random process represents a single time function when t is a variable and s is fixed at
a specific value.
Note: A random process represents a random variable when t is fixed and s is a variable.
A random process represents a number when t and s are both fixed.
4.2. Classification of Random Processes
A Random Processes X(t) has been classified in to four types as listed below
depending on whether random variable X and time t is continuous or discrete.
1. Continuous Random Processes
If a random variable X is continuous and time t can have any of a continuum of
values, then X(t) is called as a continuous random process.
Example: Thermal Noise
Fig 1: Continuous Random Processes

2. Discrete Random Processes

If a random variable X is discrete and time t is continuous, then X(t) is called as a
discrete random process. The sample functions will have only two discrete values.
Fig 2: Discrete Random Processes
3. Continuous random Sequence

If a random variable X is continuous and time t is discrete, then X(t) is called as a
continuous random sequence. Since a continuous random sequence is defined at only discrete
times, it is also called as discrete time random process. It can be generated by periodically
sampling the ensemble members of continuous random processes. These types of processes
are important in the analysis of digital signal processing systems.
Fig 3 : Continuous Random Sequence

4. Discrete Random Sequence

If a random variable X and time t areboth discrete, then X(t) is called as a discrete
random sequence. It can be generated by sampling the sample functions of discrete random
process or rounding off the samples of continuous random sequence.
Fig 4: Discrete Random Sequence
4.3. Deterministic and Non-deterministic processes
In addition to the processes, discussed above a random process can be described by the form of
its sample functions.
A Process is said to be deterministic process, if future values of any sample function can be
predicted from past values. These are also called as regular signals,which have a particular
shape.
Example: X(t) = A Sin (ωt + ϴ), A, ω and ϴ may be random variables
Fig 5: Example of Deterministic Process

A Process is said to be non-deterministic process, if future values of any sample function cannot
be predicted from past values.
Fig 6: Example of Non-Deterministic Processes
4.4. Distribution Function and Density function
Distribution Function:
Probability distribution function (PDF) which is also be called as Cumulative Distribution
Function (CDF) of a real valued random variable ‘X ‘ is the probability that X will take value
less than or equal to X. It is given by
In case of random process X(t), for a particular time t, the distribution function associated with
the random variable X is denoted as
In case of two random variables, X1 = X(t1) and X2 = X (t2), the second order joint distribution
function is two dimensional and given by
and can be similarly
extended to N random variables, called as Nth order joint distribution function
Density Function:
The probability density function (pdf) in case of random variable is defined as the derivative of
the distribution function and is given by
In case of random process, density function is given by
In case of two random functions, two dimensional density function is given by

4.5 Independence and Stationary Random Process
In probability theory, two events are independent, statistically independent, or

stochastically independent if the occurrence of one does not affect the probability of occurrence
of other i.e., having the probability of their joint occurrence equal to the product of their individual
probabilities. A random process becomes a random variable when time is fixed at some particular
value. The random variable will possess statistical properties such as a mean value, moments,
variance etc. that are related to its density function. If two random variables are obtained from the
process for two time instants they will have statistical properties related to their joint density
function. More generally, N random variables will possess statistical properties related to their N
dimensional joint density function. A random process is said to be stationary if all its statistical
properties do not change with time other processes are called non-stationary. It can also be
defined as a stationary time series is one whose statistical properties such as mean, variance,
autocorrelation, etc. are all constant over time.
Fig 1: Stationary and Non – Stationary Time Series
As an example, white noise is stationary. The sound of a cymbal clashing, if hit only once, is not
stationary because the acoustic power of the clash (and hence its variance) diminishes with time.
Statistical Independence
In case of two random variables X and Y, defined by the events A = {X ≤ x}, B = {Y ≤ y}, to be
statistically independent, Consider two processes X(t) and Y(t). The two processes, X(t) and Y(t)
are said to be statistically independent, if the random variable group X(t1), X(t2), X(t3) ……
X(tN) is independent of the group Y(t1’), Y(t2’), Y(t3’)……….Y(tM’). Independence requires that
the joint density function be factorable by groups.

Independence requires that the joint density be factorable by groups
fx,Y(x1 ,.... xN, y 1 ,.... xN; t1,......tN ) fy(y1 ,.... yN;
Statistical independence means one event conveys no information about the other; statistical
dependence means there is some information.
Stationary
To define Stationary, Distribution and density functions of random process X(t) must be defined.
For a particular time t1, the distribution function associated with the random variable
X1=X(t1)will be denoted Fx(x1; t1 ). It is defined as
Fx(x1; t1)=P{X (t1) ≤x1}

For any real number x1.
For two random variables X1=X(t1) and X2=X(t2) , the second order joint distribution function
is the two dimensional extension of above equation
Fx(x1 , x2; t1,t2 )=P{X(t1)≤x1, X(t2)≤x2}
In a similar manner, for N random variables Xi=X(ti) , i=1,2,3,....N, the Nth order joint
distribution function is
Fx(x1 ,.... xN; t1,......tN )=P{X(t1)≤x1,........ X(tN)≤xN}
Joint density functions are found from derivatives of the above joint distribution functions
fx(x1; t1 )=d Fx(x1; t1 )/dx1
fx(x1 , x2; t1,t2 )=d2Fx(x1 , x2; t1,t2 )/(dx1dx2 )
fx(x1 ,.... xN; t1,......tN )=dNFx(x1 ,.... xN; t1,......tN )/(dx1.......dxN)
First order stationary processes

A Random process is called stationary to order one, if its first order density function does not
change with a shift in time origin. In other words fx(x1; t1 ) =fx(x1; t1+Δ) must be true for any t1
and any real number Δ if X(t) is to be a first order stationary process.
If fx(x1; t1) is independent of t1, the process mean value E[X(t)]= =constant.
Second Order and wide sense Stationary

A process is called stationary to order two if its second order density function satisfies
fx(x1 , x2; t1,t2 )= fx(x1 , x2; t1+Δ,t2+Δ ) for all t1 , t2 and Δ.
The correlation E[X(t1) X(t2)]of a random process will be a function of t1 and t2. Let
us denote the function by RXX(t1,t2) and call it the autocorrelation function of the random process
X(t): RXX(t1,t2)= E[X(t1) X(t2)]

The autocorrelation function of second order stationary process is a function only of time
differences and not absolute time, that is , if =t2-t1, then
RXX(t1,t1+ )= E*X(t1) X(t1+ )= RXX( )
A Second order stationary process is wide sense stationary for which the following two
conditions are true: E[X(t)]= =constant and E[X(t1) X(t1+ )= RXX( )
N – Order and Strict Sense Stationary
A Random process is stationary to order N if its Nth order density function is invariant to a time
origin shift ; that is, if
fx(x1 ,....... xN; t1,.......tN )= fx(x1 ,....... xN; t1+Δ,......tN+Δ )
for all t1,.......tN and Δ. Stationary of order N implies stationarity to all orders k≤N. A process
stationary to all orders N=1,2,3,.... is called Strict sense Stationary.
Example 1.

4.6 Time Averages and Ergodicity: In probability theory, an ergodic system is one that has the
same behaviour averaged over time as averaged over the samples. A random process is ergodic if
its time average is the same as its average over the probability space
The time average of a quantity is defined as

The time autocorrelation function is denoted Rxx=A[ ( ) ( +𝜏)]. These functions are defined by
Rxx
By taking the expected value on both sides of the above two equations and assuming the
expectation can be brought inside the integrals, then
E[ ]=
E[Rxx( )]= RXX( )
Ergodic Theorem: The Ergodic Theorem states that for any random process X(t), all the time
averages of the sample functions of the X(t) are equal to the corresponding statistical or ensemble
averages of X(t).
i.e., =
Rxx( ) = RXX( )
Ergodic Process: Random processes that satisfy the ergodic theorem are called ergodic
processes. The analysis of Ergodicity is extremely complex. In most physical applications, it is
assumed that all stationary processes are ergodic processes.
Mean Ergodic process: A process X(t) with a constant mean value is called mean ergodic or
ergodic in the mean, if its statistical average equals the time average of any sample function
( ) with probability 1 for all sample functions; that is, if
E[ ]= =A[ ( )] = with probability 1 for all ( ).
Correlation Ergodic Processes: Analogous to a mean ergodic process, a stationary continuous
process X(t) with autocorrelation function RXX( ) is called autocorrelation ergodic or ergodic in
the autocorrelation if, and only if, for all .
4.7. Autocorrelation Function and its Properties: Consider that a random process X(t) is at
least WSS and is a function of time difference = t2-t1. Then the following are the properties of
the autocorrelation function of X(t).
1. Mean square value of X(t) is E[X2(t)] = RXX(0). It is equal to the power (average) of the
process, X(t).
Proof: We know that for X(t), RXX( ) = E[X(t) X(t+ )] . If = 0, then RXX(0) = E[X(t) X(t)] =
E[X2(t)] hence proved.
2. Autocorrelation function is maximum at the origin i.e. RXX(0).

Proof: Consider two random variables X(t1) and X(t2) of X(t) defined at time intervals t1 and t2
respectively. Consider a positive quantity [X(t1) X(t2)]2 0
Taking Expectation on both sides, we get E[X(t1) X(t2)]2 0
E[X2(t1)+ X2(t2) 2X(t1) X(t2)] 0

E[X2(t1)]+ E[X2(t2) 2E[X(t1) X(t2)] 0

RXX(0)+ RXX(0) 2 RXX(t1,t2) 0 [Since E[X2(t)] = RXX(0)]
Given X(t) is WSS and = t2-t1.
Therefore 2 RXX(0 2 RXX( ) 0
RXX(0 RXX( ) 0 or
RXX(0) hence proved.
3. RXX( ) is an even function of i.e. RXX(- ) = RXX( ).

Proof: We know that RXX( ) = E[X(t) X(t+ )]
Let = - then
RXX(- ) = E[X(t) X(t- )]
Let u=t- or t= u+
Therefore RXX(- ) = E[X(u+ ) X(u)] = E[X(u) X(u+ )]
RXX(- ) = RXX( ) hence proved.
4. If a random process X(t) has a non zero mean value, E[X(t)] 0 and Ergodic with no periodic
components, then = ̅.
Proof : Consider a random variable X(t)with random variables X(t1) and X(t2). Given the mean
value is E[X(t)] = ̅ 0 . We know that
RXX( ) = E[X(t)X(t+ )] = E[X(t1) X(t2)]. Since the process has no periodic components,
as , the random variable becomes independent, i.e. = E[X(t1)
X(t2)] = E[X(t1)] E[ X(t2)]
Since X(t) is Ergodic E[X(t1)] = E[ X(t2)] = ̅
Therefore = hence proved.
5. If X(t) is periodic then its autocorrelation function is also periodic.

Proof: Consider a Random process X(t) which is periodic with period T0
Then X(t) = X(t T0) or
X(t+ ) = X(t T0). Now we have RXX( ) = E[X(t)X(t+ )] then
RXX( T0) = E[X(t)X(t+ T0)]
Given X(t) is WSS, RXX( T0) = E[X(t)X(t+ )]
RXX( T0) = RXX( ) Therefore RXX( ) is periodic
hence proved.

6. If X(t) is Ergodic has zero mean, and no periodic components then

= .
Proof: It is already proved that = . Where ̅ is the mean value of

X(t) which is given as zero.
Therefore = hence proved.
4.8. Properties of Cross Correlation Function:

Consider two random processes X(t) and Y(t) are at least jointly WSS. And the cross
correlation function is a function of the time difference = t2-t1. Then the following are the
properties of cross correlation function.
1. RXY( ) = RYX(- ) is a Symmetrical property.
Proof: We know that RXY( ) = E[X(t) Y(t+ )] also
RYX( ) = E[Y(t) X(t+ )]
Let = - then
RYX(- ) = E[Y(t) X(t- )]
Let u=t- or t= u+ . then
RYX(- ) = E[Y(u+ ) X(u)] = E[X(u) Y(u+ )]
Therefore RYX(- ) = RXY( ) hence proved.
2. If RXX( ) and RYY( ) are the autocorrelation functions of X(t) and Y(t) respectively then the
cross correlation satisfies the inequality: .
Proof: Consider two random processes X(t) and Y(t) with auto correlation functions and
. Also consider the inequality
E[
E[
E[
We know that E[X2(t)] = RXX(0) and E[Y2(t)] = RYY(0) and E[X(t) X(t+ )] = RXY( )
Therefore

Or
hence proved.
3. If RXX( ) and RYY( ) are the autocorrelation functions of X(t) and Y(t) respectively then the
cross correlation satisfies the inequality: RYY(0)].
Proof: We know that the geometric mean of any two positive numbers cannot exceed their
arithmetic mean that is if RXX( ) and RYY( ) are two positive quantities then at =0,
RYY(0)].We know that
Therefore RYY(0)]. Hence proved.
4.9. Covariance functions for random processes:
Auto Covariance function: Consider two random processes X(t) and X(t+ ) at two time
intervals t and t+ . The auto covariance function can be expressed as CXX(t, t+ ) = E[(X(t)-
E[X(t)]) ((X(t+ ) – E[X(t+ )])] or
CXX(t, t+ ) = RXX(t, t+ ) - E[(X(t) E[X(t+ )]
If X(t) is WSS, then CXX( ) = RXX( ) - . At = 0, CXX(0) = RXX(0) - =E[X2]- =
Therefore at = 0, the auto covariance function becomes the Variance of the random process.
The autocorrelation coefficient of the random process, X(t) is defined as
(t, t+ ) = if =0,
=1. Also 1.
Cross Covariance Function: If two random processes X(t) and Y(t) have random variables
X(t) and Y(t+ ), then the cross covariance function can be defined as
CXY(t, t+ ) = E[(X(t)-E[X(t)]) ((Y(t+ ) – E[Y(t+ )])] or
CXY(t, t+ ) = RXY(t, t+ ) - E[(X(t) E[Y(t+ )]. If X(t) and Y(t) are jointly WSS, then CXY( ) =
RXY( ) - ̅ ̅. If X(t) and Y(t) are Uncorrelated then CXY(t, t+ ) =0.
The cross correlation coefficient of random processes X(t) and Y(t) is defined as
(t, t+ ) = if =0,

4.10. Gaussian Random Process: Consider a continuous random process X(t). Let N random
variables X1=X(t1),X2=X(t2), . . . ,XN =X(tN) be defined at time intervals t1, t2, . . . tN respectively.
If random variables are jointly Gaussian for any N=1,2,…. And at any time instants t 1,t2,. . . tN.
Then the random process X(t) is called Gaussian random process. The Gaussian density function
is given as
fX(x1, x2…… xN ; t1, t2,….. tN) =
where CXX is a covariance matrix.
Poisson’s random process: The Poisson process X(t) is a discrete random process which
represents the number of times that some event has occurred as a function of time. If the number
of occurrences of an event in any finite time interval is described by a Poisson distribution with
the average rate of occurrence is λ, then the probability of exactly occurrences over a time interval
(0,t)
P[X(t)=K] = , K=0,1,2, . . .
And the probability density function is -k).
4.11. System Response

Consider a continuous LTI system with impulse response h(t). Assume that the system is
always causal and stable. When a continuous time random process X(t) with ensemble members
is applied on this system, the output response is also a continuous time random process Y(t)
with ensemble members .
Let a random process X(t) be applied to a continuous linear time invariant system whose
impulse response h(t) . Then the output response is also a random process. It can be
expressed by the convolution integral, Y(t)=h(t)*X(t)
i.e. the output response is Y(t)=
Mean value of output Response:
Consider that the random process X(t) is wide sense stationary process.
Mean value of output response = E[Y(t)]
Then, E[Y(t)] =E [h(t)*X(t)]
But = = constant , since X(t) is wide sense stationary

Then E[Y(t)]=

Mean Square value of Output Response:
Where 𝜏1and 𝜏2 are the shift in time variables

If the input X(t) is a wide sense stationary random process, then
Therefore,
The above expression is independent of time t. It represents the output power.
Autocorrelation Function of Output Response:
The autocorrelation of Y(t) is
We know that
if the input X(t) is a wide sense stationary random process, let the time difference
𝜏 = t2 – t1 and t = t1, Then
Cross Correlation Function of Response:

If the input X(t) is a wide sense stationary random process, then the cross correlation
function of input X(t) and output Y(t) is
Therefore,
This expression show the relationship between the autocorrelation functions and cross
correlation functions.

UNIT – V
Random Process - Spectral Characteristics
5.1. Introduction:
To design any LTI filter which is intended to extract or suppress the signal, it is
necessary to understand how the strength of a signal is distributed in the frequency domain. There
are two immediate challenges in trying to find an appropriate frequency-domain description for a
WSS random process. The present module focuses on the expected power in the signal which is a
measure of signal strength and will be shown that it meshes nicely with the second moment
characterizations of a WSS process.
It is possible to define a meaningful PSD for a stationary (at least in the wide sense)
random process. For non-stationary processes, PSD does not exist. On these lines, the PSD of a
random process is defined as a weighted mean of the PSDs of all sample functions, as it is not
known exactly which of the sample functions may occur in a given trial.
5.2. Power Spectral Density of a random Process:

Consider
As
will represent the random process X (t).
Applying Fourier Transform, We get
Using Parseval’s theorem to find the energy of the signal
Therefore, the power associated with XT (t) is
The average power is given by
Where

is the contribution to the average power at frequency w and represents the power spectral density
the power spectral density for
As T tends to infinite, the left-hand side in the above expression represents the average power of
X (t). Therefore, the PSD (𝜔) of the process X(t) is defined by
Thus, The PSD (𝜔) of a random process X(t) is defined as the ensemble average of the PSDs of
all sample functions, Thus,
Relation Between Power-spectral Density and Autocorrelation function of the Random

Process (Wiener-Khinchin Relation)
By Taking Inverse FT of PSD, we have

Since, we know that
Similarly, we have
We know that
Let,
Therefore,
Hence,
The RHS of the above eq. is the time average of Auto correlation function.
Thus, Time average of Autocorrelation function and the PSD form a Fourier Transform Pair.
Thus, for a WSS process, Autocorrelation and Power Spectral Density form a Fourier
Transform Pair and this is referred to as Wiener-Khintchine relation.

5.3.Properties of power density spectrum:

The properties of the power density spectrum SXX(ω) for a WSS random process X(t) are given
as
(1) SXX(ω)
Proof: From the definition, the expected value of a non negative function E[ ] is always
non-negative.
Therefore SXX(ω) hence proved.
(2) The power spectral density at zero frequency is equal to the area under the curve of the
autocorrelation RXX ( ) i.e. SXX
Proof: From the definition we know that SXX at ω=0,
SXX hence proved
(3) The power density spectrum of a real process X(t) is an even function i.e.
SXX(-ω)= SXX(ω)
Proof: Consider a WSS real process X(t), then SXX also
SXX
Substitute = - then
SXX
Since X(t) is real, from the properties of autocorrelation we know that, RXX (- ) = RXX ( )
Therefore SXX
SXX(-ω)= SXX(ω) hence proved.
(4) SXX(ω) is always a real function
Proof: We know that SXX
Since the function is a real function, SXX(ω) is always a real function hence proved.

(5) If SXX(ω) is a psd of the WSS random process X(t), then

= A {E[X 2(t)]} = RXX (0) or The time average of the mean square value of a
WSS random process equals the area under the curve of the power spectral density.
Proof: We know that RXX ( ) = A {E[X (t + ) X(t)]}
at =0,
RXX (0) = A {E[X2(t)]} = = Area under the curve of the power spectral density.
Hence proved.
1. The autocorrelation function of a WSS process X(t) is given by
Find the power spectral density of the process.

Solution:
5.4 Cross Power density spectrum:

Consider two real random processes X(t) and Y(t). which are jointly WSS random
processes, then the cross power density spectrum is defined as the Fourier transform of the cross
correlation function of X(t) and Y(t).and is expressed as
SXY and SYX

By inverse Fourier transformation, we can obtain the cross correlation functions as
RXY and RYX
Therefore the cross psd and cross correlation functions are forms a Fourier transform pair.

If XT(ω) and YT(ω) are Fourier transforms of X(t) and Y(t) respectively in interval [-T,T], Then
the cross power density spectrum is defined as
SXY and SYX
Relation Between cross Power-spectral Density and Cross Correlation function
Consider the inverse Fourier Transform of cross PSD

i.e.
 Since,
Similarly,

Hence
Let 1 = 𝜏 → 1= 𝜏
Hence,
Thus, time average of cross-correlation function and the cross spectral density form a
Fourier Transform Pair. Thus, for a two jointly WSS processes, cross-correlation and cross
Spectral Density form a Fourier Transform Pair.
5.5. Properties of cross power density spectrum:

The properties of the cross power for real random processes X(t) and Y(t) are given by
(1) SXY(-ω)= SXY(ω) and SYX(-ω)= SYX(ω)
Proof: Consider the cross correlation function The cross power density spectrum is
SXY
Let = - Then
SXY Since RXY(- ) = RXY( )
SXY
Therefore SXY(-ω)= SXY(ω) Similarly SYX(-ω)= SYX(ω) hence proved.
(2) The real part of SXY(ω) and real part SYX(ω) are even functions of ω i.e.
Re [SXY(ω)] and Re [SYX(ω)] are even functions.
Proof: We know that SXY and also we know that
=cosωt-jsinωt, Re [SXY
Since cos ωt is an even function i.e. cos ωt = cos (-ωt)

Re [SXY
Therefore SXY(ω)= SXY(-ω) Similarly SYX(ω)= SYX(-ω) hence proved.

(3) The imaginary part of SXY(ω) and imaginary part SYX(ω) are odd functions of ω i.e.
Im [SXY(ω)] and Im [SYX(ω)] are odd functions.
Proof: We know that SXY and also we know that
=cosωt-jsinωt,
Im [SXY - Im [SXY(ω)]
Therefore Im [SXY(ω)] = - Im [SXY(ω)] Similarly Im [SYX(ω)] = - Im [SYX(ω)] hence proved.
(4) SXY(ω)=0 and SYX(ω)=0 if X(t) and Y(t) are Orthogonal.

Proof: From the properties of cross correlation function, We know that the random processes
X(t) and Y(t) are said to be orthogonal if their cross correlation function is zero. i.e. RXY( ) =
RYX( ) =0.
We know that SXY
Therefore SXY(ω)=0. Similarly SYX(ω)=0 Hence proved.
(5) If X(t) and Y(t) are uncorrelated and have mean values and , then
SXY(ω)=2 .
Proof: We know that SXY
Since X(t) and Y(t) are uncorrelated, we know that

=
Therefore SXY
SXY
SXY
Therefore SXY(ω)=2 . Hence proved.
5.6 Problems:
1.A random process is defined as , where is a WSS process, 𝜔0
is a real constant and is a uniform random variable over(0,2𝜋) an is independent of X(t). Find
the PSD of Y(t).
Soln.: . ]
Since, and X(t) are independent of each other,
. 𝐶(𝜔0( +𝜏)+𝜃)]

PSD of Y(t) is Fourier Transform of . i.e
2.A random process is given by , where A and B are real constants and
X(t) and
Y(t) are jointly WSS processes.
(i) Find the Power spectrum of Z(t) (ii) Find the cross power spectrum (𝜔)
Soln.:
]]
Power spectrum of Z(t) is
(ii)
= }]
= ]=
3.A stationary random process X(t) has a spectral density and an independent
stationary Y(t) has a spectral density . Assuming X(t) and Y(t) are of zero
mean, find the (i) PSD of U(t)=X(t)+Y(t) (ii)
Soln.:
(i) PSD of U(t)= PSD of X(t) +PSD of Y(t) =1
(ii)

5.7. Spectral characteristics of system response
Let the input for an LTI systemof impulse response h(t), be a WSS process x(t), resulting an
output y(t).
x(t) LTI System y (t )

Impulse Response
h(t)
The response of the system y(t)= x(t)*h(t)=
Mean value of the Output Process
Expected Value of the output is
Since x(t) is WSS, .

Then,
For an LTI system, Unit impulse response h(t) and its Transfer function H(𝜔) form a Fourier
Transform Pair, i.e.
Then,
Hence, .
Thus, the mean of the output process is independent of time and is a constant.
Autocorrelation Function of the Output Process
Where 𝜏= 2 − 1
Thus, the Autocorrelation function of the output process is independent of time and is
dependent on “𝜏". Hence, y(t) is a stationary process.

Mean Squared Value of the Output Process
Since, y(t) is stationary, the above equation can be written as
The MS value of y(t) is
Cross correlation Function between the input process and output process
In the above eq.1, replace 𝜏 −𝜏 i.e .

Since, Autocorrelation function is an even function of
.
In the above eq.2, replace second ‘x’ in the prefix by ‘y’ i.e.
In the above eq.3, replace 𝜏 −𝜏, i.e. .

It same as , since Autocorrelation is having even symmetry.
Since, ----4.
Relation between the PSDs of the input process and output process of an LTI Systems
Take Fourier Transform on both sides for eq.2
---5
Take Fourier Transform on both sides for eq.4
---6
Substituting from eq.5 in eq.6
=
The power at the output of the LTI system is the area enclosed by the output PSD
i.e.
Output Power

5.8 Problems:
1. The input to an LTI system with Impulse response h(t)=δ(t)+t.exp(-at).U(t) is a WSS process
with mean of 3.Find the mean of the output of the system.
Soln.:
Hence,
2. A WSS process X(t) is applied to the following System:
+
Add
X(t) Y(t)
Delay 2T
If the input PSD is K Watts/Hz, find the out put PSD

Soln.:
The system is represented as
The Transfer function of the above system is and the corresponding
(𝜔) 2 =2+2.𝐶 2𝜔
The output PSD is W/Hz.

QUESTION BANK
SIGNALS AND STOCHASTIC PROCESS DEPARTMENT OF ECE
INDEX
S.NO OBJECTIVE PAGE NO
1. TUTORIAL SHEETS 2
2. ASSIGNMENT QUESTIONS 5
3. UNITWISE IMPORTANT QUESTIONS 7
4. OBJECTIVE TYPE QUESTIONS 13
26
5. INTERNAL QUESTION PAPERS
6. EXTERNAL QUESTION PAPERS 28
1
TUTORIAL SHEETS
UNIT – I
1. Compute the energy E∞ and the power P∞ of the following continuous-time signal
x(t)=e−2πjt.
2. Make a sketch of each of the following signals (a) f(n) = X∞ k=0 (−0.9)k δ(n − 3 k) (b)
g(n) = X∞ k=−∞ (−0.9)|k| δ(n − 3 k) (c) x(n) = cos(0.25 π n) u(n) (d) x(n) = cos(0.5 π n)
u(n)
3. Classify each of the following discrete-times systems. (a) y(n) = cos(x(n)). (b) y(n) = 2 n
2 x(n) + n x(n + 1). (c) y(n) = max {x(n), x(n + 1)} Note: the notation max{a, b} means
for example; max{4, 6} = 6. (d) y(n) = ฀ x(n) when n is even x(n − 1) when n is odd (e)
y(n) = x(n) + 2 x(n − 1) − 3 x(n − 2). (f) y(n) = X∞ k=0 (1/2)k x(n − k). That is, y(n) =
x(n) + (1/2) x(n − 1) + (1/4) x(n − 2) + · · · y(n) = x(2 n)
4. A discrete-time system is described by the following rule y(n) = ( x(n), when n is an even
integer −x(n), when n is an odd integer where x is the input signal, and y the output
signal. (a) Sketch the output signal, y(n), produced by the 5-point input signal, x(n)
illustrated below. 1 2 3 2 1 -2 -1 0 1 2 3 4 5 6 n x(n) (b) Classify the system as: i.
linear/nonlinear ii. time-invariant/time-varying iii. stable/unstable.
UNIT – II
1. State whether y =tan x can be expressed as a Fourier series. If so how?. If not why?
2. State the convergence condition on Fourier series.
3. To what value does the sum of Fourier series of f (x) converge at the point of
continuity x =a ?
4. To what value does the sum of Fourier series of f ( x) converge at the point of
discontinuity x =a ?
5. Write the formulae for Fourier constants for f ( x) in the interval (-p, p).
2
UNIT – III
1. The Z-transform of the discrete-time signal x(n) is X(z) = −3 z 2 + 2 z −3

Accurately sketch the signal x(n).
2. Define the discrete-time signal x(n) as x(n) = −0.3 δ(n + 2) + 2.0 δ(n) + 1.5 δ(n
− 3) − δ(n − 5) (a) Sketch x(n). (b) Write the Z-transform X(z). (c) Define G(z) = z −2
X(z). Sketch g(n).
Let x(n) be the length-5 signal x(n) = {1, 2, 3, 2, 1} where x(0) is underlined. Sketch
the signal corresponding to each of the following Z-transforms. (a) X(2z) (b) X(z 2 )
(c) X(z) + X(−z) (d) X(1/z)
3. Sketch the discrete-time signal x(n) with the Z-transform X(z) = (1 + 2 z) (1 + 3 z −1 )
(1 − z −1 ).
4. Define three discrete-time signals: a(n) = u(n) − u(n − 4) b(n) = δ(n) + 2 δ(n − 3) c(n)
= δ(n) − δ(n − 1)
Define three new Z-transforms: D(z) = A(−z), E(z) = A(1/z), F(z) = A(−1/z) (a)
Sketch a(n), b(n), c(n) (b) Write the Z-transforms A(z), B(z), C(z) (c) Write the Z-
transforms D(z), E(z), F(z) (d) Sketch d(n), e(n), f(n)
5. Find the Z-transform X(z) of the signal x(n) = 4 ฀ 1 3 ฀n u(n) − ฀ 2 3 ฀n u(n).
UNIT-IV
1. Differentiate between Random Processes and Random variables with example

2. Prove that the Auto correlation function has maximum value at the origin i.e
│RXX( )│= RXX(0)
3. A stationary ergodic random processes has the Auto correlation function with the
periodic components as RXX( ) = 25 + 4 1+6𝜏 2
4. Define mean ergodic random processes and correlation ergodic Random processes. 5.
Find the mean value of Response of a linear system.
5. a) Define Wide Sense Stationary Process and write it‟s conditions. b) A random
process is given as X(t) = At, where A is a uniformly distributed random variable on
(0,2). Find whether X(t) is wide sense stationary or not.
6. X(t) is a stationary random process with a mean of 3 and an auto correlation function
of 6+5 exp (-0.2 │ │). Find the second central Moment of the random variable Y=Z-
W, where „Z‟ and „W‟ are the samples of the random process at t=4 sec and t=8 sec
respectively.
7. Explain the following i) Stationarity ii) Ergodicity iii) Statistical independence with
respect to random processes
3
UNIT – V
1. Define wiener khinchine relations
2. State any two properties of cross-power density spectrum.
3. Define cross power spectral density and its examples.
4. Derive the relationship between cross-power spectral density and cross correlation
function.
a) Check the following power spectral density functions are valid or not )
𝑜 8(𝜔) 2+𝜔4 ) −(𝜔−1) 2 b) Derive the relation between input PSD and
output PSD of an LTI system
5. X(t) is a stationary random process with a mean of 3 and an auto correlation function
of 6+5 exp (-0.2 │ │). Find the second central Moment of the random variable Y=Z-
W, where ‘Z’ and ‘W’ are the samples of the random process at t=4 sec and t=8 sec
respectively.
6. a) Define Wide Sense Stationary Process and write it’s conditions.
b) A random process is given as X(t) = At, where A is a uniformly distributed random

variable on (0,2). Find whether X(t) is wide sense stationary or not.
1. A stationary process has an autocorrelation function
Find Mean, Mean square and Variance of the process X(t).
4
ASSIGNMENT QUESTIONS
UNIT-I
1. Define Root Mean Square value of a function.

2. Find the R.M.S value of y =x 2 in (-p, p).
3. (a) For the systems represented by the following functions. Determine whether
every system is (1) stable (2) Causal (3) linear (4) Shift invariant
(i) T[x(n)]= ex(n) (ii) T[x(n)]=ax(n)+6
4. 2. Determine whether the following systems are static or Dynamic, Linear or
Nonlinear,Shift variant or Invarient, Causal or Non-causal, Stable or unstable.
(i) y(t) = x(t+10) + 2 x (t) (ii) dy(t)/dt + 10 y(t) = x(t)
5. Explain about the properties of continuous time fourier series.
6. Find the fourier coefficients of the given signal. (4)
x(t) = 1+ sin 2wt + 2 cos 2wt + cos (3wt)
7. Determine the Fourier series coefficient of exponential representation of x(t)
x(t) = 1, |t| (8)
0, T1< |t|< T/ 2
UNIT-II
1. Find the constant term a0 in the Fourier series corresponding to f (x )= x -x3 in (-

π, π).
2. If f (x)=x 2 -x4 is expanded as a Fourier series in (-l,l ), find the value of bn .
3. Write the Fourier sine series of k in (0,p).
4. Find the R.M.S value if f ( x ) = x2 in -π £x £.π
5. State the Parseval’s Identity (or) theorem
UNIT-III
1. The signal x is defined as x(n) = a |n| Find X(z) and the ROC. Consider separately the
cases: |a| < 1 and |a| ≥ 1
2. Find the Unilateral Z-transform and R.O.C of x(n) = sin ω0 n u(n)
3. A finite sequence x[n]=[5,3,-2,0,4,-3]. Find X[z]and its ROC
4. Find Z- transform of x (n) =cos (now) u (n)
5. Find the z-transform of the following sequences
i) x[n] = a-n u[-n-1] ii) x[n] = u[-n] iii)x[-n] = -an u[-n-1]
UNIT-IV
1. Explain about Poisson Random process and also find its mean and variance.
2. The function of time Z(t) = X1cosω0t- X2sinω0t is a random process. If X1 and
X2are independent Gaussian random variables, each with zero mean and variance 2 ,
find E[Z]. E[Z2 ] and var(z).
5
3. Briefly explain the distribution and density functions in the context of stationary
and independent random processes.
4. Explain about the following random process (i) Mean ergodic process (ii)
Correlation ergodic process (iii) Gaussian random process
5. State and prove the auto correlation and cross correlation function properties.
6. a) Define Wide Sense Stationary Process and write it’s conditions. b) A random
process is given as X(t) = At, where A is a uniformly distributed random variable on
(0,2). Find whether X(t) is wide sense stationary or not.
UNIT-V
1. a) Check the following power spectral density functions are valid or not )
𝑜 8(𝜔) 2+𝜔4 ) −(𝜔−1) 2 b) Derive the relation between input PSD and
output PSD of an LTI system
function.
3. Give the statement of ergodic theorem.
4. Differentiate between Random Processes and Random variables with example (3
marks) i. Define Power Spectrum Density.
5. Show that SXX(-ω) = SXX(ω). i.e., Power spectrum density is even function of ω
6. A stationery random process X(t) has spectral density SXX(ω)=25/ (𝜔 2+25) and
an independent stationary process Y(t) has the spectral density SYY(ω)= 𝜔 2 / (𝜔
2+25). If X(t) and Y(t) are of zero mean, find the: a) PSD of Z(t)=X(t) + Y(t) b)
Cross spectral density of X(t) and Z(t)
7. a) The input to an LTI system with impulse response h(t)= 𝛿 + 2 −𝑎 . U(t) is
a WSS process with mean of 3. Find the mean of the output of the system.
8. State and prove Power Spectral density with three properties.
6
UNIT WISE IMPORTANT QUESTION QUESTIONS
UNIT-I
PART-A-UNIT-I (2 or 3 marks)
1. State the properties of LTI system.
2. Draw the function π(2t+3) when π(t) = 1 ; for t ≤ ½ 0 ; otherwise
3. Determine whether the following signal is energy or power signal. And calculate its
Energy or Power. x(t)=e-2tu(t).
4. Check whether the following system is static or dynamic & also causal or noncausal.
y(n)=x(2n).
5. Define Fourier transform pair.

6. Explain how aperiodic signals can be represented by fourier transform.
7. Give the mathematical & graphical representation of CT & DT unit impulse function
8. What are the conditions for a system to be LTI system?
9. State any 2 properties of unit impulse function.
10. What is Aliasing?
11. What are the Conditions for a System to be LTI System?
12. Define time invariant and time varying systems.
13. Is the system describe by the equation y(t) = x(2t) Time invariant or not? Why?
PART-B-UNIT-I (5-10 marks)
1. Derive the expression for component vector of approximating the function f1(t) over
f2(t) and also prove that the component vector becomes zero if the f1(t) and f2(t) are
orthogonal.
2. Define Linearity and Time-Invariant properties of a system and write with an

example.
7
3. Find the convolution of two signals x(n) = { 1, 1,0, 1, 1} and h(n) = { 1, -2, -3, 4}
and represent them graphically
4. Find the convolution of x(t) = u(t+1) and h(t) = u(t-2)
UNIT-II
PART-A-UNIT-II (2 or 3 marks)
1. Find the convolution of the two signals x(t)= e -2t u(t) h(t)= u(t+2)
2. Find the Fourier transform of x(t) = t cos ωt
3. Discuss the block diagram representation of an LTI-DT system
4. Consider a causal LTI system as in the fig
a.
Determine the differential equation relating x(n) and y(n)
5. State Fourier integral theorem.
6. Show that f(x) = 1, 0 < x < ¥ cannot be represented by a Fourier integral.
PART-B-UNIT-II (5-10 marks)
1. Obtain the cosine fourier series representation of x(t)
2. Find the trigonometric fourier series of the figure shown below
8
3. State and prove Parseval’s Theorem

4. Find the fourier transform of a rectangular pulse of duration T and amplitude A
5. Show that e x2/2 is reciprocal with respect to Fourier transforms
6. State sampling theorem for band limited signals. Prove theorem graphically.
What is aliasing effect?
UNIT-III
PART-A-UNIT-III (2 or 3 marks)
1. Prove the following property of the z-transform: zn0x[n]→X(zz0)

2. Compute the inverse z-transform of X(z)=log(1+z),|z|<1.
3. Find the z transform of 3n + 2 × 3 n
4. Find the Unilateral Z-transform and R.O.C of x(n) = sin ω0 n u(n)
5. Find the z-transform of each of the following sequences: (a) x(n)= 2 n u(n)+3(½)n u(n)
0)u(n).(b)x(n)=cos(n)
6. Find the inverse of each of the following z-transforms:
7. Find the inverse z-transform of X(z) = sin z
PART-B-UNIT-III (5-10 marks)
1. Determine the Z=Transform of x1(n)=an and x2(n) =nu(n)

2. Find the inverse laplace transform of X(S) = S / S2+5S+6
3. State and prove initial value and final value theorems.
4.
Using convolution theorem find inverse Laplace transform of s/(s2+a2)2
5. State and prove differentiation and integration property of z-transform.
UNIT-IV
PART-A-UNIT-IV (2 or 3 marks)
1. Define random process?

2. Define ergodicity?
3. Define mean ergodic process?
4. Define wide sense stationary random processes.
9
5. Give the statement of ergodic theorem.

6. Define the auto covariance & cross covariance functions of Random processes X(t).
7. When two random processes X(t)& Y(t) are said to be independent.
8. Define the cross correlation function between two random processes X(t) & Y(t)..
PART-B-UNIT-IV (5-10 marks)
1. Define Wide Sense Stationary Process and write it‟s conditions.

2. A random process is given as X(t) = At, where A is a uniformly distributed
random variable on (0,2). Find whether X(t) is wide sense stationary or not. 2.
X(t) is a stationary random process with a mean of 3 and an auto correlation
function of 6+5 exp (-0.2 │ │). Find the second central Moment of the random
variable Y=Z-W, where „Z‟ and „W‟ are the samples of the random process at
t=4 sec and t=8 sec respectively.
3. Explain the following i) Stationarity ii) Ergodicity iii) Statistical independence
with respect to random processes
4. Given the RP X(t) = A cos(w0t) + B sin (w0t) where ω0 is a constant, and A and
B are uncorrelated Zero mean random variables having different density functions
but the same variance
5. Show that X(t) is wide sense stationary. b) Define Covariance of the Random
processes with any two properties.
6. A Gaussian RP has an auto correlation function RXX( )=6 sin (𝜋𝜏 ) 𝜋𝜏 .
Determine a covariance matrix for the Random variable X(t) b) Derive the
expression for cross correlation function between the input and output of a LTI
system.
UNIT-V
PART-A-UNIT-V (2 or 3 marks)
1. Define Power Spectrum Density.

2. Give the statement of Wiener-Khinchin relation.
3. Define spectrum Band width and RMS bandwidth.
4. Write any two properties of Power Spectrum Density. 5. Define linear system.
10
5. Show that SXX(-ω) = SXX(ω). i.e., Power spectrum density is even function of
ω.
6. If the Power spectrum density of x(t) is SXX(ω), find the PSD of 𝑥( ).
7. If the Auto correlation function of wide sense stationary X(t) is RXX( )=4+2e -2
𝜏 . Find the area enclosed by the power spectrum density curve of X(t).
8. Define linear system and derive the expression for output response. 5. If X(t) &
Y(t)are uncorrelated and have constant mean values & then show that
SXX(ω)= 2Π 𝛿(𝜔)
PART-B-UNIT-V (5-10 marks)
1. Check the following power spectral density functions are valid or not ) 𝑜 8(𝜔) 2 +
𝜔4 ) −(𝜔−1)
2. Derive the relation between input PSD and output PSD of an LTI system
function. 3. A stationery random process X(t) has spectral density SXX(ω)=25/ (𝜔
2+25) and an independent stationary process Y(t) has the spectral density SYY(ω)= 𝜔
2 / (𝜔 2+25). If X(t) and Y(t) are of zero mean, find the: a) PSD of Z(t)=X(t) + Y(t) b)
Cross spectral density of X(t) and Z(t)
4. The input to an LTI system with impulse response h(t)= 𝛿 + 2 −𝑎 . U(t) is a
WSS process with mean of 3. Find the mean of the output of the system.
5. Define Power Spectral density with three properties.
6. A random process Y(t) has the power spectral density SYY(ω)= 9 𝜔2+64 Find i) The
average power of the process ii) The Auto correlation function
7. State the properties of power spectral density
8. A random process has the power density spectrum SYY(ω)= 6𝜔 2 1+𝜔4 . Find the
average power in the process.
9. Find the auto correlation function of the random process whose psd is 16 𝜔2+4 7. a)
Find the cross correlation function corresponding to the cross power spectrum
SXY(ω)= 6 (9+𝜔2)(3+ 𝜔) 2 b) Write short notes on cross power density spectrum.
10. Consider a random process X(t)=cos(𝜔 + 𝜃)where 𝜔 is a real constant and 𝜃is a
uniform random variable in (0, π/2). Find the average power in the process.
11. Define and derive the expression for average power of Random process.
11
12. The power spectrum density function of a stationary random process is given by
SXX(𝜔)= A, -K< 𝜔< K 0, other wise Find the auto correlation function.
13. Derive the expression for power spectrum density. 10. a) Define and derive the
expression for average cross power between two random process X(t) and Y(t).
14. Find the cross power spectral density for RXX( )=𝐴 2 2 sin⁡(𝜔0𝜏)
12
OBJECTIVE QUESTIONS
UNIT-I:
1.Which mathematical notation specifies the condition of periodicity for a continuous

time signal?
a. x(t) = x(t +T0)
b. x(n) = x(n+ N)
c. x(t) = e-αt
d. None of the above
ANSWER: (a) x(t) = x(t +T0)
2. Which among the below specified conditions/cases of discrete time in terms of real
constant ‘a’, represents the double-sided decaying exponential signal?
a. a > 1
b. 0 < a < 1
c. a < -1
d. -1 < a < 0
ANSWER: (d) -1 < a < 0
3) Damped sinusoids are _____

a. sinusoid signals multiplied by growing exponentials
b. sinusoid signals divided by growing exponentials
c. sinusoid signals multiplied by decaying exponentials
d. sinusoid signals divided by decaying exponentials
ANSWER: (c) sinusoid signals
4. An amplitude of sinc function that passes through zero at multiple values of an

independent variable ‘x’ ______
a. Decreases with an increase in the magnitude of an independent variable (x)
b. Increases with an increase in the magnitude of an independent variable (x)
c. Always remains constant irrespective of variation in magnitude of ‘x’
d. Cannot be defined
ANSWER: (a) Decreases with an increase in the
5) A system is said to be shift invariant only if______

a. a shift in the input signal also results in the corresponding shift in the output
b. a shift in the input signal does not exhibit the corresponding shift in the output
c. a shifting level does not vary in an input as well as output
d. a shifting at input does not affect the output
ANSWER: (a) a shift in the input signal also results in the corresponding shift in the
output
6) Which condition determines the causality of the LTI system in terms of its impulse
response?
a. Only if the value of an impulse response is zero for all negative values of time
b. Only if the value of an impulse response is unity for all negative values of time
c. Only if the value of an impulse response is infinity for all negative values of time
d. Only if the value of an impulse response is negative for all negative values of time
13
ANSWER: (a) Only if the value of an impulse response is zero for all negative values of
time
7) Under which conditions does an initially relaxed system become unstable?

a. only if bounded input generates unbounded output
b. only if bounded input generates bounded output
c. only if unbounded input generates unbounded output
d. only if unbounded input generates bounded output
ANSWER: (a) only if bounded input generates unbounded output
8) Which among the following are the stable discrete time systems?
1. y(n) = x(4n)
2. y(n) = x(-n)
3. y(n) = ax(n) + 8
4. y(n) = cos x(n)
9) An equalizer used to compensate the distortion in the communication system by
faithful recovery of an original signal is nothing but an illustration of _________
a. Static system
b. Dynamic system
c. Invertible system
ANSWER: (c) Invertible system
10) Which block of the discrete time systems requires memory in order to store the
previous input?
a. Adder
b. Signal Multiplier
c. Unit Delay
d. Unit Advance
ANSWER: (c)Unit Delay
11) Which type/s of discrete-time system do/does not exhibit the necessity of any
feedback?
a. Recursive Systems
b. Non-recursive Systems
c. Both a & b
ANSWER: (b) Non-recursive Systems
12) Recursive Systems are basically characterized by the dependency of its output on
_______
a. Present input
b. Past input
c. Previous outputs
d. All of the above
ANSWER: (d) All of the above
13) What does the term y(-1) indicate especially in an equation that represents the
behaviour of the causal system?
14
a. initial condition of the system

b. negative initial condition of the system
c. negative feedback condition of the system
d. response of the system to its initial input
ANSWER: (a) initial condition of the system
14) Which type of system response to its input represents the zero value of its initial
condition?
a. Zero state response
b. Zero input response
c. Total response
d. Natural response
ANSWER: (a) Zero state response
15) Which is/are the essential condition/s to get satisfied for a recursive system to be
linear?
a. Zero state response should be linear
b. Principle of Superposition should be applicable to zero input response
c. Total Response of the system should be addition of zero state & zero input responses
d. All of the above
16) Which among the following operations is/are not involved /associated with the
computation process of linear convolution?
a. Folding Operation
b. Shifting Operation
c. Multiplication Operation
d. Integration Operation
ANSWER: (d) Integration Operation
17) A LTI system is said to be initially relaxed system only if ____

a. Zero input produces zero output
b. Zero input produces non-zero output
c. Zero input produces an output equal to unity
ANSWER:(a) Zero input produces zero output
18) What are the number of samples present in an impulse response called as?
a. string
b. array
c. length
d. element
ANSWER: (c) length
19) Double-sided phase & amplitude spectra _____

a. Possess an odd & even symmetry respectively
b. Possess an even & odd symmetry respectively
c. Both possess an odd symmetry
d. Both possess an even symmetry
ANSWER: (a) Possess an odd & even symmetry respectively
15
20) Which is/are the mandatory condition/s to get satisfied by the transfer function for
the purpose of distortionless transmission?
a. Amplitude Response should be constant for all frequencies
b. Phase should be linear with frequency passing through zero
c. Both a & b
ANSWER: (c)Both a & b
UNIT-II:
1) What does the first term ‘a0‘ in the below stated expression of a line spectrum
indicate?
x(t) = a0 + a1 cos w0 t + a2 cos2 w0t +……+ b1 sin w0 t + b2 sin w0 t + …..
a. DC component
b. Fundamental component
c. Second harmonic component
d. All of the above
ANSWER: (a) DC component
2) Which kind of frequency spectrum/spectra is/are obtained from the line spectrum of
a continuous signal on the basis of Polar Fourier Series Method?
a. Continuous in nature
b. Discrete in nature
c. Sampled in nature
d. All of the above
ANSWER: (b) Discrete in nature
3) Which type/s of Fourier Series allow/s to represent the negative frequencies by
plotting the double-sided spectrum for the analysis of periodic signals?
a. Trigonometric Fourier Series
b. Polar Fourier Series
c. Exponential Fourier Series
d. All of the above
ANSWER: (c) Exponential Fourier Series
4) What does the signalling rate in the digital communication system imply?
a. Number of digital pulses transmitted per second
b. Number of digital pulses transmitted per minute
c. Number of digital pulses received per second
d. Number of digital pulses received per minute
ANSWER: (a)Number of digital pulses transmitted per second
5) Duality Theorem / Property of Fourier Transform states that _________

a. Shape of signal in time domain & shape of spectrum can be interchangeable
b. Shape of signal in frequency domain & shape of spectrum can be interchangeable
c. Shape of signal in time domain & shape of spectrum can never be interchangeable
d. Shape of signal in time domain & shape of spectrum can never be interchangeable
16
ANSWER: (a) Shape of signal in time domain & shape of spectrum can be
interchangeable
6) Which property of fourier transform gives rise to an additional phase shift of -2π
ftd for the generated time delay in the communication system without affecting an
amplitude spectrum?
a. Time Scaling
b. Linearity
c. Time Shifting
d. Duality
ANSWER: (c) Time Shifting
7) Which among the below assertions is precise in accordance to the effect of time
scaling?
A : Inverse relationship exists between the time and frequency domain representation of
signal
B : A signal must be necessarily limited in time as well as frequency domainsa. A is true & B
is false
b. A is false & B is true
c. Both A & B are true
d. Both A & B are false
ANSWER: (a)A is true & B is false
8) What is/are the crucial purposes of using the Fourier Transform while analyzing
any elementary signals at different frequencies?
a. Transformation from time domain to frequency domain
b. Plotting of amplitude & phase spectrum
c. Both a & b
9) What is the possible range of frequency spectrum for discrete time fourier series
(DTFS)?
a. 0 to 2π
b. -π to +π
c. Both a & b
10) Which among the following assertions represents a necessary condition for the
existence of Fourier Transform of discrete time signal (DTFT)?
a. Discrete Time Signal should be absolutely summable
b. Discrete Time Signal should be absolutely multipliable
c. Discrete Time Signal should be absolutely integrable
d. Discrete Time Signal should be absolutely differentiable
ANSWER: (a)Discrete Time Signal should be absolutely summable
11)What is the nature of Fourier representation of a discrete & aperiodic signal?
a. Continuous & periodic
b. Discrete & aperiodic
c. Continuous & aperiodic
d. Discrete & periodic
ANSWER: (a) Continuous & periodic
17
12) Which property of periodic signal in DTFS gets completely clarified / identified by
the equation x (n – n0)?
a. Conjugation
b. Time Shifting
c. Frequency Shifting
d. Time Reversal
ANSWER: (b) Time Shifting
13) Which theorem states that the total average power of a periodic signal is equal to
the sum of average powers of the individual fourier coefficients?
a. Parseval’s Theorem
b. Rayleigh’s Theorem
c. Both a & b
ANSWER: (a) Parseval’s Theorem
14) Which among the below mentioned transform pairs is/are formed between the
auto-correlation function and the energy spectral density, in accordance to the property
of Energy Spectral Density (ESD)?
a. Laplace Transform
b. Z-Transform
c. Fourier Transform
d. All of the above
ANSWER: (c) Fourier Transform
UNIT-III:
1) A Laplace Transform exists when ______

A. The function is piece-wise continuous
B. The function is of exponential order
C. The function is piecewise discrete
D. The function is of differential order
a. A & B
b. C & D
c. A & D
d. B & C
ANSWER: (a) A & B
2) Where is the ROC defined or specified for the signals containing causal as well as
anti-causal terms?
a. Greater than the largest pole
b. Less than the smallest pole
c. Between two poles
d. Cannot be defined
ANSWER: (c) Between two poles
3) What should be the value of laplace transform for the time-domain signal equation
e-at cos ωt.u(t)?
a. 1 / s + a with ROC > – a
b. ω / (s + a) 2 + ω2 with ROC > – a
c. s + a / (s + a)2 + ω2 with ROC > – a
d. Aω / s2 + ω2 with ROC > 0
18
ANSWER: (c) s + a / (s + a)2 + ω2 with ROC σ > – a

4) According to the time-shifting property of Laplace Transform, shifting the signal in
time domain corresponds to the ______
a. Multiplication by e-st0 in the time domain
b. Multiplication by e-st0 in the frequency domain
c. Multiplication by est0 in the time domain
d. Multiplication by est0 in the frequency domain
ANSWER: (b) Multiplication by e-st0 in the frequency domain
5) Which result is generated/ obtained by the addition of a step to a ramp function?
a. Step Function shifted by an amount equal to ramp
b. Ramp Function shifted by an amount equal to step
c. Ramp function of zero slope
d. Step function of zero slope
ANSWER: (b) Ramp Function shifted by an amount equal to step
6) Unilateral Laplace Transform is applicable for the determination of linear constant
coefficient differential equations with ________
a. Zero initial condition
b. Non-zero initial condition
c. Zero final condition
d. Non-zero final condition
ANSWER: (b) Non-zero initial condition
7) What should be location of poles corresponding to ROC for bilateral Inverse
Laplace Transform especially for determining the nature of time domain signal?
a. On L.H.S of ROC
b. On R.H.S of ROC
c. On both sides of ROC
ANSWER: (c) On both sides of ROC
8) Generally, the convolution process associated with the Laplace Transform in time
domain results into________
a. Simple multiplication in complex frequency domain
b. Simple division in complex frequency domain
c. Simple multiplication in complex time domain
d. Simple division in complex time domain
ANSWER: (a) Simple multiplication in complex frequency domain
9) An impulse response of the system at initially rest condition is basically a response
to its input & hence also regarded as,
a. Black’s function
b. Red’s function
c. Green’s function
ANSWER: (c) Green’s function
10) When is the system said to be causal as well as stable in accordance to pole/zero of
ROC specified by system transfer function?
a. Only if all the poles of system transfer function lie in left-half of S-plane
b. Only if all the poles of system transfer function lie in right-half of S-plane
c. Only if all the poles of system transfer function lie at the centre of S-plane
ANSWER: (a) Only if all the poles of system transfer function lie in left-half of S-plane
11) Correlogram is a graph of ______
19
a. Amplitude of one signal plotted against the amplitude of another signal

b. Frequency of one signal plotted against the frequency of another signal
c. Amplitude of one signal plotted against the frequency of another signal
d. Frequency of one signal plotted against the time period of another signal
ANSWER: (a) Amplitude of one signal plotted against the amplitude of another signal
12) According to Rayleigh’s theorem, it becomes possible to determine the energy of a
signal by________
a. Estimating the area under the square root of its amplitude spectrum
b. Estimating the area under the square of its amplitude spectrum
c. Estimating the area under the one-fourth power of its amplitude spectrum
d. Estimating the area exactly half as that of its amplitude spectrum
ANSWER: (b) Estimating the area under the square of its amplitude spectrum
13) The ESD of a real valued energy signal is always _________
a. An even (symmetric) function of frequency
b. An odd (non-symmetric) function of frequency
c. A function that is odd and half-wave symmetric
ANSWER: (a) An even (symmetric) function of frequency
14) Which among the below mentioned assertions is /are correct?
a. Greater the value of correlation function, higher is the similarity level between two signals
b. Greater the value of correlation function, lower is the similarity level between two signals
c. Lesser the value of correlation function, higher is the similarity level between two signals
d. Lesser the value of correlation function, lower is the similarity level between two signals
a. Only C
b. Only B
c. A & D
d. B & C
ANSWER: (c) A & D
15) Which property is exhibited by the auto-correlation function of a complex valued
signal?
a. Commutative property
b. Distributive property
c. Conjugate property
d. Associative property
ANSWER: (c) Conjugate property
16) Where does the maximum value of auto-correlation function of a power signal
occur?
a. At origin
b. At extremities
c. At unity
d. At infinity
ANSWER: (a) At origin
UNIT-IV:
1. What does the set comprising all possible outcomes of an experiment known as ?
a. Null event
b. Sure event
20
c. Elementary event
ANSWER: b. Sure event
2. What does an each outcome in the sample space regarded as ?
a. Sample point
b. Element
c. Both a & b
ANSWER: c. Both a & b
3. Mutually Exclusive events ________
a. Contain all sample points
b. Contain all common sample points
c. Does not contain any common sample point
d. Does not contain any sample point
ANSWER: c. Does not contain any common sample point
4. What would be the probability of an event 'G' if G denotes its complement, according
to the axioms of probability?
a. P (G) = 1 / P (G)
b. P (G) = 1 - P (G)
c. P (G) = 1 + P (G)
d. P (G) = 1 * P (G)
ANSWER: b. P (G) = 1- P (G)
5. What would happen if the two events are statistically independent ?
a. Conditional probability becomes less than the elementary probability

b. Conditional probability becomes more than the elementary probability
c. Conditional probability becomes equal to the elementary probability
d. Conditional as well as elementary probabilities will exhibit no change
ANSWER: c. Conditional probability becomes equal to the elementary probability
6. What would be the joint probability of statistically independent events that occur
simultaneously ?
a. Zero
b. Not equal to zero
c. Infinite
ANSWER: b. Not equal to zero
7. Consider the assertions given below :

A : CDF is a monotonously increasing function
B : PDF is a derivative of CDF & is always positive
21
Which among them is correct according to the properties of PDF?

a. A is true & B is false
c. Both A & B are true but B is a reason for A
d. Both A & B are false since B is not a reason for A
ANSWER: c. Both A & B are true but B is a reason for A
8. The Joint Cumulative Density Function (CDF) _____
a. Is a non-negative function
b. Is a non-decreasing function of x & y planes
c. Is always a continuous function in xy plane
d. All of the above
ANSWER: d. All of the above
9. What is the value of an area under the conditional PDF ?
a. Greater than '0' but less than '1'

b. Greater than '1'
c. Equal to '1'
d. Infinite
ANSWER: c. Equal to '1'
10. When do the conditional density functions get converted into the marginally density
functions ?
a. Only if random variables exhibit statistical dependency

b. Only if random variables exhibit statistical independency
c. Only if random variables exhibit deviation from its mean value
ANSWER: b. Only if random variables exhibit statistical independency
11. Which among the below mentioned standard PDFs is/are applicable to discrete
random variables ?
a. Gaussian distribution
b. Rayleigh distribution
c. Poisson distribution
d. All of the above
ANSWER: c. Poisson distribution
12. A random variable belongs to the category of a uniform PDF only when __________
a. It occurs in a finite range

b. It is likely to possess zero value outside the finite range
c. Both a & b
ANSWER: c. Both a & b
13. What would happen if the value of term [(m-x) / (σ √2)] increases in the expression
of Guassian CDF?
a. Complementary error function also goes on increasing
22
b. Complementary error function goes on decreasing
c. Complementary error function remains constant or unchanged
d. Cannot predict
ANSWER: b. Complementary error function goes on decreasing
14. Which type of standard PDFs has/ have an ability to describe an integer valued
random variable concerning to the repeated trials carried /conducted in an experiment?
a. Binomial
b. Uniform
c. Both a & b
ANSWER: a. Binomial
UNIT-V:
1) What does the set comprising all possible outcomes of an experiment known as?
a. Null event
b. Sure event
c. Elementary event
ANSWER: (b) Sure event
2) What does an each outcome in the sample space regarded as?
a. Sample point
b. Element
c. Both a & b
ANSWER: (c) Both a & b
3) Mutually Exclusive events ________
a. Contain all sample points
b. Contain all common sample points
c. Does not contain any common sample point
d. Does not contain any sample point
ANSWER: (c) Does not contain any common sample point
4) What would be the probability of an event ‘G’ if G denotes its complement,
according to the axioms of probability?
a. P (G) = 1 / P (G)
b. P (G) = 1 – P (G)
c. P (G) = 1 + P (G)
d. P (G) = 1 * P (G)
ANSWER:(b) P (G) = 1 – P (G)
5) What would happen if the two events are statistically independent?
a. Conditional probability becomes less than the elementary probability
b. Conditional probability becomes more than the elementary probability
c. Conditional probability becomes equal to the elementary probability
d. Conditional as well as elementary probabilities will exhibit no change
23
ANSWER: (c) Conditional probability becomes equal to the elementary probability

6) What would be the joint probability of statistically independent events that occur
simultaneously?
a. Zero
b. Not equal to zero
c. Infinite
ANSWER: (b) Not equal to zero
7) Consider the assertions given below
A : CDF is a monotonously increasing function
B : PDF is a derivative of CDF & is always positive
Which among them is correct according to the properties of PDF?
a. A is true & B is false
c. Both A & B are true but B is a reason for A
d. Both A & B are false since B is not a reason for A
ANSWER: (c) Both A & B are true but B is a reason for A
8) The Joint Cumulative Density Function (CDF) _____
a. Is a non-negative function
b. Is a non-decreasing function of x & y planes
c. Is always a continuous function in xy plane
d. All of the above
9) What is the value of an area under the conditional PDF?
a. Greater than ‘0’ but less than ‘1’
b. Greater than ‘1’
c. Equal to ‘1’
d. Infinite
ANSWER: (c) Equal to ‘1’
10) When do the conditional density functions get converted into the marginally
density functions?
a. Only if random variables exhibit statistical dependency
b. Only if random variables exhibit statistical independency
c. Only if random variables exhibit deviation from its mean value
ANSWER: (b) Only if random variables exhibit statistical independency
11) Which among the below mentioned standard PDFs is/are applicable to discrete
random variables?
a. Gaussian distribution
b. Rayleigh distribution
c. Poisson distribution
d. All of the above
ANSWER: (c) Poisson distribution
12) A random variable belongs to the category of a uniform PDF only when ______
a. It occurs in a finite range
b. It is likely to possess zero value outside the finite range
c. Both a & b
ANSWER: (c) Both a & b
24
13) What would happen if the value of term [(m-x) / (σ √2)] increases in the expression
of Guassian CDF?
a. Complementary error function also goes on increasing
b. Complementary error function goes on decreasing
c. Complementary error function remains constant or unchanged
d. Cannot predict
ANSWER: (b) Complementary error function goes on decreasing
14) Which type of standard PDFs has/ have an ability to describe an integer valued
random variable concerning to the repeated trials carried /conducted in an experiment?
a. Binomial
b. Uniform
c. Both a & b
ANSWER: (a) Binomial
25
INTERNAL QUESTION PAPER

PART-A
Answer All the following Questions (5*2=10 M)
1. a) A rectangular function f(t) is defined by f(t) =1for 0<t< π and f(t) =–1 for π<t<2π.
Approximate this function by a waveform `Sint' over the interval (0,1) such that the
mean square error is zero.
b) Explain how signum function is expressed in terms of unit step function.
2. Define orthogonal functions. Give some examples of orthogonal functions. Verify
that sinusoidal functions are orthogonal or not.
–|t|
3. Determine the Fourier transform of a two sided exponential pulse x (t) = e .
4. State and prove time convolution and time differentiation properties of

Fourier Transform.
5.Find the Fourier transform of a gate pulse of unit height, unit width and
centered at t=0.
PART-B
Answer the following Questions (3*10=30 M)
1. (a) Define autocorrelation function and explain with properties.
(b) Statistically Independent zero mean random processes X(t) and Y(t) have auto
correlation functions respectively. Find
the auto correlation of W1(t)=X(t)+Y(t).

(OR)
2. (a) Define Random process and explain classification of random processes with neat
sketch.
(b) A random process is described by X(t)=A2.cos2 ωt+θ here A a d ω are o sta ts a d θ
is a random variable uniformly distributed between (-π, π .
Is X(t) wide sense stationary or not.
3. (a) Check the given function is valid or not. [2M]
(b) Derive the equation for power density spectrum and state any four properties of it.
[8M]
(OR)
4. State and prove wiener-khintchine relation.
5. (a) Find the inverse z-Transform of
26
Using Residue Method.

(b) Sate and prove initial and final value theorems of z-Transform
(OR)
6. (a) Show that
(b) Auto correlation of a WSS random process X(t) is where are
constants. Find PSD.
27
EXTERNAL QUESTION PAPER
28
29
SIGNALS & STOCHASTIC PROCESS – PREFINAL PAPER
PART-A
1. (a) Define mean square error and give equation for it

[3M]
(b) Define system bandwidth and signal bandwidth [3M]
(c) State Dirichlet Conditions[2M]
(d) Find the Fourier Transform of a Symmetrical Gate Pulse [3M]
(e) Define ROC of Laplace Transform and state any 4 Properties [2M]
(f) Find the z-Transform of [3M]
(g) Assume that ergodic process has an autocorrelation function

Find its mean, mean square and variance. [3M]
(h) Define Nyquist rate and Nyquist Interval [3M]

(i) What is WSS and State its conditions [2M]
PART-B
1. Obtain the condition under which two signals are said to
orthogonal to each other. Hence prove that are
orthogonal to each other for all integer values of (m,n)

[10M]
(OR)
2. (a) Derive the relation between rise time and Bandwidth

[5M]
(b) Explain the method of Graphical approach for Convolution with an example [5M]
4. (a) Define even symmetry and derive the trigonometric fourier series coefficients
for odd function
[5M]
(b) For the Periodic function given below, derive the exponential form of fourier series
[5M]
(OR)
5. (a) State and prove the given properties of fourier transform

30
(i) Convolution (ii) Time Shifting (iii) Frequency Shifting [6M]

(b) What is Aliasing Effect and explain methods to avoid it [4M]
6. (a) State Sampling theorem and explain the construction of sampled Signal
[5M]
(b) Find the Inverse z-Transform of using power
series expansion [5M]
(OR)
7. (a) Find the inverse laplace transform of
(b) Prove that the sequences have same X(z)
and differ only in ROC. Plot their ROC

8. (a) What is Cross Correlaton and explain with Properties
[5M]
Pro e that the ra do pro ess X t =A. os ωt+θ is WSS of it is assu ed that ω a d A are
o sta ts a d θ is a U ifor ly Distri uted Ra do Varia le o er 0,2π
[5M]
(OR)
9. Prove the relation between Cross PSD and Cross Correlation

[10M]
10. State and prove wiener-khintchine relation
[10M]
(OR)
11. (a) Show that
[5M]
(b) Define Cross PSD and explain with Properties [5M]
********
31

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SIGNALS & STOCHASTIC PROCESS DEPT.

SIGNALS AND STOCHASTIC PROCESS

B. Vamsi Krishna Mrs. V. Poornima M. Sruthi

Department of Electronics & Communication Engineering

MALLAREDDY ENGINEERING COLLEGE FOR WOMEN

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 1

EC304ES: SIGNALS AND STOCHASTIC PROCESS

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 2

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 3

Unit-I Part I: Signal Analysis Page No

Part II: Signal Transmission through Linear Systems

Unit- II Fourier series, Fourier Transforms, and Sampling

Unit- III Part I: Laplace Transforms

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 4

3.6 Inverse Laplace Transform 75

UNIT – IV Random Processes – Temporal Characteristics

UNIT – V Random Process - Spectral Characteristics

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 5

1.2. Classification of the Signals:

1.2.2 Periodic Signals

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 6

Figure 1.1: Description of Continuous, Discrete and Digital Signals

Figure 1.2: A typical periodic signal

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 7

1.2.4 Energy and Power signals

1.2.5 Deterministic and Random signal

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 8

1.3. Analogy between Vectors and Signals

The vector V1 can be expressed in terms of vector V2

Where Ve is the error vector.

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 9

1.5 Signal approximation using Orthogonal functions

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 10

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 11

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 12

1.9. Standard Signals

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 13

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 14

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 15

PART II - Signal Transmission through Linear Systems

 Linear and Non-linear Systems

Linear and Non-linear Systems

A system is said to be linear when it satisfies superposition and homogeneity principles.

T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

Time Variant and Time Invariant Systems

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 16

Example: y(n) = x(-n)

Causal and Non-Causal Systems

Stable and Unstable Systems

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 17

1.12. Transfer Function of an LTI System:

A continuous time system is shown in fig:

i/p signal The o/p signal

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 18

1.13. Filter characteristics of Linear Systems:

1.14. Distortion less transmission through a system

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 19

Therefore inverse Fourier transform, the corresponding impulse response must be

So for distortion less transmission of a signal through a system, the magnitude

1.15. Signal bandwidth & System bandwidth:

Fig. Frequency response

MALLA REDDY ENGINEERING COLLEGE FOR WOMEN Page 20

1.16. Ideal LPF, HPF and BPF characteristics