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System and Signal (Notes)

The document outlines the course structure for a B.Tech Signals and Systems class, detailing course objectives, outcomes, and a comprehensive syllabus divided into five units covering signal analysis, Fourier series, transforms, and linear systems. It also includes information on sampling techniques such as impulse, natural, and flat top sampling, along with the Nyquist rate and considerations for band pass signals. Textbooks and reference materials are provided for further study.

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0% found this document useful (0 votes)
101 views184 pages

System and Signal (Notes)

The document outlines the course structure for a B.Tech Signals and Systems class, detailing course objectives, outcomes, and a comprehensive syllabus divided into five units covering signal analysis, Fourier series, transforms, and linear systems. It also includes information on sampling techniques such as impulse, natural, and flat top sampling, along with the Nyquist rate and considerations for band pass signals. Textbooks and reference materials are provided for further study.

Uploaded by

swapanrajwar75
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

INTERNAL EXTERNAL TOTAL TOTAL

L T P
II B. TECH MARKS MARKS MARKS CREDITS
I SEMESTER
3 0 0 30 70 100 3
SUB CODE:
SIGNALS AND SYSTEMS
R23EC2103

COURSE OBJECTIVES:

1. To explain about signals and perform various operations on it and to summarize the sampling
2. To build Trigonometric and Exponential Fourier series of various signals
3. To develop Fourier transforms for various signals.
4. To solve Laplace transforms and z-transforms for various signals.
5. To summarize the principle, filter characteristics, band width of a LTI Systems and
COURSE OUTCOMES:
After completion of this course, the students will be able to
CO1: Define basic signals and its operations.
CO2: Identify Trigonometric and Exponential Fourier Series of signals.
CO3: Develop Fourier Transforms for various signals.
CO4: Solve Laplace Transform and z-Transform for various signals.
CO5: Compare LTI system responses for different inputs and illustrate sampling concepts.

SYLLABUS:
UNIT- I: SIGNAL ANALYSIS & SAMPLING:
Classification of Signals: Analog, Discrete, Digital, Deterministic & Random, Periodic &
Aperiodic, Even & Odd, Energy & Power signals. Basic Operations on Signals: Time-Shifting,
Time-Scaling, Time-Reversal, Amplitude Scaling and Signal Addition. Elementary Signals:
Unit Step, Unit Ramp, Unit Parabolic, Impulse, Sinusoidal function, Exponential function,
Gate function, Triangular function, Sinc function and Signum function, Convolution,
Correlation, Relation between convolution & correlation.
Sampling: Sampling theorem, Types of Sampling: Impulse Sampling, Natural and flattop
Sampling, Reconstruction of signal from its samples, Effect of under sampling – Aliasing

UNIT- II: ORTHOGONAL FUNCTIONS & FOURIER SERIES:


Orthogonal Functions: Signal approximation using orthogonal functions, Mean square error,
Orthogonality in complex functions.
Fourier series: Representation of Fourier series, Continuous time periodic signals, Properties
of Fourier Series, Dirichlet conditions, Trigonometric Fourier Series and Exponential Fourier
Series, Conversion of Exponential Fourier Series from Trigonometric Fourier series.

UNIT- III: 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, Introduction
to Hilbert Transform.

UNIT- IV: LAPLACE TRANSFORMS AND Z- TRANSFORMS:


Laplace Transforms: Laplace transforms, Partial fraction expansion, Inverse Laplace
transform, Concept of Region of Convergence (ROC), Constraints on ROC for various classes
of signals, Properties of Laplace transforms, Relation between Laplace transform and Fourier
transform.
Z-Transforms: Concept of Z-transform of a discrete sequence, Region of convergence in Z-
Transform, constraints on ROC for various classes of signals, inverse Z-transform, properties
of Z-Transforms.

UNIT-V: SIGNAL TRANSMISSION THROUGH LINEAR SYSTEMS:

Signal transmission through linear systems: System and its types: Linear & Non-Linear,
Time Variant & Time Invariant, Causal & Non Causal, Static & Dynamic, Stable & Unstable.
Impulse response of a linear time invariant (LTI) system and 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
and its characteristics.

TEXT BOOKS:

1. B.P. Lathi, “Signals, Systems and Communications”, BS Publications, 2008.

2. Simon Haykin and Van Veen, Wiley, “Signals and Systems”, Second Edition, 2003.

3. A.V. Oppenheim, A.S. Will sky and S.H. Nawab, “Signals and Systems”, PHI, Second
Edition, 2013.

REFERENCE BOOKS:

1. Ramesh Babu, “Signals and Systems”, SciTech Publications, Third Edition, 2011.
2. A. Anand Kumar, “Signals and Systems”, PHI Publications, Third Edition, 2013.
3. Tarun Kumar and Rawat - SIGNALS AND SYSTEMS, Oxford Publications, 2010.

WEB RESOURCES: [Link]


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SIGNALS SAMPLING TECHNIQUES
[Link] Copyright © [Link]

There are three types of sampling techniques:

Impulse sampling.

Natural sampling.

Flat Top sampling.

Impulse Sampling
Impulse sampling can be performed by multiplying input signal xt with impulse train
Σ∞n=−∞ δ(t − nT ) of period 'T'. Here, the amplitude of impulse changes with respect to amplitude
of input signal xt . The output of sampler is given by

y(t) = x(t) × impulse train

= x(t) × Σ∞
n=−∞ δ(t − nT )

y(t) = yδ (t) = Σ∞
n=−∞ x(nt)δ(t − nT ) . . . . . . 1

To get the spectrum of sampled signal, consider Fourier transform of equation 1 on both sides
1
Y (ω) = T
Σ∞
n=−∞ X(ω − nωs )

This is called ideal sampling or impulse sampling. You cannot use this practically because pulse
width cannot be zero and the generation of impulse train is not possible practically.

Natural Sampling
Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse train
of period T. i.e. you multiply input signal xt to pulse train Σ∞
n=−∞ P (t − nT ) as shown below

The output of sampler is

y(t) = x(t) × pulse train


= x(t) × p(t)
= x(t) × Σ∞
n=−∞ P (t − nT ) . . . . . . (1)
The exponential Fourier series representation of pt can be given as

p(t) = Σ∞
n=−∞Fn e
jnωs t . . . . . . (2)

= Σ∞
n=−∞Fn e
j2πnfs t

T
1
Where Fn = T
∫ 2
−T p(t)e−jnωs tdt
2

1
= TP
(nωs )
Substitute Fn value in equation 2

∴ p(t) = Σ∞ 1
n=−∞ T P(n ωs )e
jnωs t

= T1 Σ∞
n=−∞P(n ωs )e
jnωs t

Substitute pt in equation 1

y(t) = x(t) × p(t)


= x(t) × T1 Σ∞
n=−∞P(n ωs ) e
jnωs t

y(t) = T1 Σ∞
n=−∞P(n ωs ) x(t) e
jnωs t

To get the spectrum of sampled signal, consider the Fourier transform on both sides.

F . T [y(t)] = F . T[ T1 Σ∞
n=−∞P(n ωs ) x(t) e
jnωs t ]

= T1 Σ∞
n=−∞P(n ωs ) F . T [x(t) e
jnωs t ]

According to frequency shifting property

F . T [x(t) ejnωs t ] = X[ω − nωs ]


∴ Y [ω] = T1 Σ∞
n=−∞P(n ωs ) X[ω − n ωs ]

Flat Top Sampling


During transmission, noise is introduced at top of the transmission pulse which can be easily
removed if the pulse is in the form of flat top. Here, the top of the samples are flat i.e. they have
constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat top sampling
makes use of sample and hold circuit.

Theoretically, the sampled signal can be obtained by convolution of rectangular pulse pt with
ideally sampled signal say yδt as shown in the diagram:

y(t) = p(t) × (t) . . . . . . (1)


i.e. y(t) = p(t) × yδ (t) . . . . . . (1)

To get the sampled spectrum, consider Fourier transform on both sides for equation 1

Y [ω] = F. T [P (t) × yδ (t)]


By the knowledge of convolution property,

Y [ω] = P (ω) Yδ (ω)


Here P (ω) = T Sa( ωT
2
) = 2 sin ωT /ω

Nyquist Rate
It is the minimum sampling rate at which signal can be converted into samples and can be
recovered back without distortion.

Nyquist rate fN = 2fm hz

Nyquist interval = 1 = 1
seconds.
fN 2fm

Samplings of Band Pass Signals


In case of band pass signals, the spectrum of band pass signal X[ω] = 0 for the frequencies outside
the range f1 ≤ f ≤ f2 . The frequency f1 is always greater than zero. Plus, there is no aliasing effect
when fs > 2f2 . But it has two disadvantages:

The sampling rate is large in proportion with f2 . This has practical limitations.

The sampled signal spectrum has spectral gaps.

To overcome this, the band pass theorem states that the input signal xt can be converted into its
samples and can be recovered back without distortion when sampling frequency fs < 2f2 .

Also,

1 2f
fs = = 2
T m
f2
Where m is the largest integer <
B

and B is the bandwidth of the signal. If f2 =KB, then

1 2KB
fs = =
T m
For band pass signals of bandwidth 2fm and the minimum sampling rate fs= 2 B = 4fm,

1
the spectrum of sampled signal is given by Y [ω] = T
Σ∞
n=−∞ X[ω − 2nB]
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