Dr. Parul Tyagi(Asso. Prof.) & Dr. Neha Singh (Asst. Prof.

)
Electronics and Communication Engg.
JECRC. Jaipur (Raj), India
Email:- [email protected] , [email protected]

1
 Vision of JECRC
 To become a renowned center of outcome based learning, and work towards
academic, professional, cultural and social enrichment of the lives of individuals
and communities.

Mission of JECRC
 Focus on evaluation of learning outcomes and motivate students to inculcate
research aptitude by project based learning.
 Identify, based on informed perception of Indian, regional and global needs,
areas of focus and provide platform to gain knowledge and solutions.
 Offer opportunities for interaction between academia and industry.
 Develop human potential to its fullest extent so that intellectually capable and
imaginatively gifted leaders can emerge in a range of professions.
 Vision of the Department
 To contribute to the society through excellence in scientific and technical
education, teaching and research aptitude in Electronics and Communication
Engineering to meet the needs of Global Industry.

Mission of the Department

 M1: To equip the students with strong foundation of basic sciences and
domain knowledge of ECE, so that they are able to creatively their
knowledge to the solution of problems arising in their career path.
 M2: To induce the habit of lifelong learning to continuously enhance overall
performance.
 M3: Students are able to communicate their ideas clearly and concisely so
that they can work in team as well as an individual.
 M4: To make the students responsive towards the ethical, social,
environmental and in economic context for the society.
Course Outcomes
Upon successful completion of this course the students will have
developed following skills/abilities:

CO1: Represent signals mathematically in

continuous and discrete time and frequency
domain
CO 2 Get the response of an LSI system to
different signals
CO 3 Design of different types of digital filters
for various applications
CO 4 Estimation of spectral parameters
CO 5 Application of Digital Signal Processing
 Disclaimer
The contents used in this presentation are
taken from the text books mentioned in the
references. I do not hold any copyrights for
the contents. It has been prepared to use in
the class lectures, not for commercial
purpose.
UNIT-1
 OVERVIEW OF UNIT-1
 Course Outline
 Text & Reference Books
 Prerequisites of the subject
 Introduction of Digital Signal Processing
 Discrete time signals: Sequences; representation of
signals on orthogonal basis
 Sampling and reconstruction of signals
 Z-Transform
 Analysis of LSI systems
 frequency Analysis
 Inverse Systems
 Examples

8
 Course Outline
 Digital Signal Processing (DSP) is at the heart of
many applications in a wide array of fields:
speech and audio processing, system monitoring
and fault detection, biomedical signal analysis,
mobile and internet communications, radar and
sonar, vibration measurement and analysis,
seismograph analysis, image/video coding and
decoding etc.
 The objective of this course is to strengthen the
students’ knowledge of DSP fundamentals, and
to familiarize them with the practical aspects of
DSP algorithm development and implementation.
 A.V. Oppenheim and R.W. Schafer, Discrete-Time
Signal Processing, 3rd Edition, Pearson Higher
Education Inc., 2010.

 R. Chassaing and D. Reay, Digital Signal Processing

and Applications with the TMS320C6713 and
TMS320C6416 DSK, 2nd Edition, Wiley IEEE Press,
2008.

 S.K. Mitra, Digital Signal Processing: A Computer-

Based Approach, third edition, McGraw-Hill Inc.,
New York, 2005.

 J.G. Proakis and D.G. Manolakis, Digital Signal

Processing: Principles, Algorithms, and
Applications, fourth edition, Prentice Hall 10
 PREREQUISITES

 Fourier analysis
 Laplace Transform
 SAMPLING
 Fourier series representation of signals
 Continuous time and discrete time signals and
systems

11
Introduction

By Dr. Y. Narasimha Murthy, Ph.D

 Signal
It carries information
It can be a function of time, temp, pressure, distance etc.
It represents some independent variable which are
associated with system.
System can be electrical and mechanical.
Electrical e.g. are current , voltage, electric.
Mechanical e.g. are force, speed etc.
Signal Processing
Signal processing is the analysis interpretation, and
manipulation of signals like sound, images time-varying
measurement values and sensor data etc.

For example biological data such electrocardiograms, control

system signals, telecommunication transmission signals such
as radio signals, and many others.
Need of Signal Processing
When a signal is transmitted from one point to
another there is every possibility of contamination
/deformation of the signal by external noise. So to
retrieve the original signal at the receiver suitable
filters are to be used. i.e. the signal is processed to
obtain the pure signal.
Analog Signal Processing
The analog signal processing is basically, filtering of
the signal . It can be denoted by the following diagram
Digital Signal Processing
The digital signal processor consists of anti-aliasing
filter, analog to digital converter (ADC), a digital
filter represented by the transfer function H(z), a
digital to analog converter and a reconstruction filter.
Advantages of Digital over Analog Signal
Processing
Accuracy: The analog circuits are prone to temperature
and external effects, but the digital filters have no such
problems.
Flexibility: Reconfiguration of analog filters is very
complex whereas the digital filters can be reconfigured
easily by changing the program coefficients.
Digital signals can be easily stored on any magnetic
media or optical media are using semiconductor chips.
Easy operation: Even complex mathematical operations
can be performed easily using computers, which is not the
case with analog processing.
System
Integrated unit composed of diverse interacting structure
to perform a specific task.
e.g. filtering of noise in a communication receiver,
detection of range of a target in RADAR, monitoring
steam pressure in a boiler.
Function of a system- process a given input sequence to
generate an output sequence. e.g. CRO, function
generator, open loop, closed loop.
Continuous time (CT) & discrete time
(DT) signals
CT signals take on real or complex values as a function of an
independent variable that ranges over the real numbers and are
denoted as x( t ).
DT signals take on real or complex values as a function of an
independent variable that ranges over the integers and are
denoted as x[n].
The subtle use of parentheses and square brackets to distinguish
between CT and DT signals.
Representation of Discrete Time Signal
Graphical Representation
E.g. Consider a signal x[n] with values
x[-1]=0.5,x[0]=1,x[1]=2,x[2]=0.5,x[3]=2
Representation of Discrete Time
Signal
Functional Representation
E.g. Consider a signal x[n] with values
x[-1]=0.5,x[0]=1,x[1]=2,x[2]=0.5,x[3]=2
Representation of Discrete Time
Signal
Tabular Representation
E.g. Consider a signal x[n] with values
x[-1]=0.5,x[0]=1,x[1]=2,x[2]=0.5,x[3]=2

Sequence Representation
Sequence in Discrete time (DT)
signals
Today smart phone has become one of the most important
gadget.
The influencing application of smart phone is speech
recognisation.
With sppech recognisation the smart phone makes a call
when given a specific name. E.g. Call ayushi
The sound vibration are created in air are in the form of
analog signals.
The analog signals are first converted into discrete
signal.so A to D converter is required.
Discrete time sequence are compared with the sequence that are
already stored in the system.

When both the sequence matches, the smart phone makes the call.

Discrete signals are discrete in time but continuous in amplitude.

Orthogonal Sequence

By Neso Academy, GATE Lecture

Representaion of Signal on Orthogonal
Basis
Orthogonality: Property that allow transmission of more than
one signal on a coomon channel with successful detection.
Orthogonal Signal: If the signals are mutually independent.
Consider a figure

Not Independent
Properties of Orthogonal Sequence
Properties of Orthogonal Sequence
3

4
Numerical on Orthogonal Sequence

Find average and total energy

Numerical on Orthogonal Sequence

Find average and total energy

Q. Calculate average power and rms value of signal x(t)
 Sampling
The signals we use in the real world, such as our voices,
are called "analog" signals.
 To process these signals in computers, we need to
convert the signals to "digital" form.
 Analog signal is continuous in both time and amplitude, a
digital signal is discrete in both time and amplitude
The process of converting an analog to digital signal is
‘Analog-to-Digital Conversion’.
The ADC involves three steps which are:
1) Sampling 2) Quantization 3) Coding

• To convert a signal from continuous time to discrete time, a

process called sampling is used. The value of the signal is
measured at certain intervals in time.

•Each measurement is referred to as a sample

Sampling can be done for functions varying in space, time, or
any other dimension, and similar results are obtained in two or
more dimensions.

The sampling frequency or sampling rate, fs , is the average

number of samples obtained in one second (samples per second),
thus

f s= 1/T

During sampling process, a continuous-time signal is converted

into discrete -time signals by taking samples of continuous-time
signal at discrete time intervals.
x(nTs) = x(t)
T=Sampling Interval
x (t)=Analog input signal
Sampling theorem
Sampling theorem gives the criteria for minimum number of
samples that should be taken.
Sampling criteria:-”Sampling frequency must be twice of the
highest frequency”
fs=2W fs=sampling frequency
w=higher frequency content
2w also known as Nyquist rate

• Nyquist rate is defined as the minimum sampling rate for the

perfect reconstruction of the continuous time signals from
samples.

• Nyquist rate=2*highest frequency component =2*W

• So sampling rate must be greater than or equal to nyquist rate

Sampling theorem Proof
Sampling theorem Proof
Aliasing/Under Sampling
While providing sampling theorem we considered fs=2W

Consider the case that fs < 2W

When one under samples a band pass signal, the samples are
indistinguishable from the samples of a low-frequency samples of
the high-frequency signal.

In such a way that the lowest-frequency alias satisfies the Nyquist
criterion, because the band pass signal is still uniquely represented
and recoverable. Such under sampling is also known as band pass
sampling, harmonic sampling, IF sampling, and direct IF to digital
conversion.
Over Sampling
In Oversampling a signal is sampled faster than its
Nyquist rate.
Oversampling is used in most modern analog-to-
digital converters to reduce the distortion or noise
effects introduced by practical digital to-analog
converters
• Audio sampling

 Digital audio uses pulse-code modulation and digital signals for sound
reproduction.

Includes analog-to-digital conversion (ADC), digital-to-analog conversion

(DAC), storage, and transmission.

•PCM(Pulse Code Modulation)

 Pulse-code modulation (PCM) is a method used to digitally represent

sampled analog signals. It is the standard form of digital audio in computers,
Compact Discs, digital telephony and other digital audio applications
 • Video Sampling

 Standard-definition television (SDTV) uses either 720 by 480 pixels or 704

by 576 pixels for the visible picture area.

 High-definition-television (HDTV)uses 720p , and 1080p.

 In digital video, the sampling rate is defined as the frame rate.

• Speech sampling

 Speech signals, i.e., signals intended to carry only human speech, can usually
be sampled at a much lower rate.

Mostly almost all of the energy is contained in the 5Hz-4 kHz range, allowing
a sampling rate of 8 kHz.

This is the sampling rate used by nearly all telephony systems, which use the
G.711 sampling and quantization specifications.
Effects of Aliasing
1.Distortion.
2.The data is lost and it cannot be recovered.

To avoid Aliasing
1.sampling rate must be fs>=2W.
2. strictly band limit the signal to ’W’.
Numerical
Determine nyquist rate for continuous signal

s(t) = 5cos(50πt) + 20sin(300πt) -10cos(100πt)

In general form, any continuous signal can be written as
S(t)=A1 cos (jw1 t)+ A2 cos (jw2 t)+ A3 cos(jw3 t)
F1= w1/2π = 50 π /2 π = 25HZ
F2= w2/2 π = 300 π /2 π = 150HZ
F3= w3/2 π = 100 π /2 π = 50HZ
Here, highest frequency component=150HZ
Hence Nyquist rate=2*150HZ=300HZ
Given Continuous time signal
s(t) = 5cos(200πt)
What is the minimum sampling rate(nyquist rate)?
Highest frequency=100HZ So, Nyquist rate=2W=2*100=200HZ

•If sampling frequency is 400HZ then what is the discrete time signal
obtained?

f=freq of continuous signal/sampling freq =100/400=1/4 Discrete time

signal=5 cos(2πfn)=5 cos (2 π *1/4 n) =5 cos(π n/2)
 Question / Answers

What is an ideal Sampler?

 A theoretical ideal sampler produces samples equivalent to the

instantaneous value of the continuous signal at the desired points.

Write the relation between continuous-time signal f(t) and a discrete time
(sampled) signal f (k T)?

What is Distortion?

 In DAC , conversion from digital back to analog, the deviations from the
theoretically perfect reconstruction, collectively referred to as distortion.
 Question / Answers

What is PCM(Pulse Code Modulation)?

 A method used to digitally represent sampled analog signals. It’s the standard
form of digital audio in PC’s, CD’s & digital telephony etc. In a PCM stream,
the amplitude of the analog signal is sampled regularly at uniform intervals, and
each sample is quantized to the nearest value within a range of digital steps.

Difference Between Under sampling and Oversampling?

In Under sampling a band pass signal is sampled slower than its Nyquist
rate, while in Oversampling a signal is sampled faster than its Nyquist rate.
The Z-Transform
By
Shahbaz Goshtasebi
Introduction
 The Laplace Transform (s domain) is a valuable tool for representing,
analyzing & designing continuos-time signals & systems.

 The z-transform is convenient yet invaluable tool for representing,
analyzing & designing discrete-time signals & systems.

 The resulting transformation from s-domain to z-domain is called z-
transform.

 The relation between s-plane and z-plane is described below :
z = esT
 The z-transform maps any point s = σ + jω in the s-plane to z-plane (r θ).
The Z-Transform
Z-Transform Definition
Geometrical interpretation of
z-transform
Pole-zero Plot
Example
Region Of Convergence (ROC)
Properties of ROC
 A ring or disk in the z-plane centered at the origin.

 The Fourier Transform of x(n) is converge absolutely if the ROC includes the unit

circle.
 The ROC cannot include any poles

 Finite Duration Sequences: The ROC is the entire z-plane except possibly z=0 or

z=∞.
 Right sided sequences (causal seq.): The ROC extends outward from the outermost

finite pole in X(z) to z=∞.

 Left sided sequences: The ROC extends inward from the innermost nonzero pole in

X(z) to z=0.
 Two-sided sequence: The ROC is a ring bounded by two circles passing through two

pole with no poles inside the ring

Properties of z-Transform
Rational z-Transform
Commonly used z-Transform pairs
Z-Transform & pole-zero
distribution &
Stability considerations
Z-Transform & pole-zero
distribution &
Stability considerations – cont.
Stable and Causal Systems
By Fariza Zahari
linear time-invariant / linear shift-
invariant
Linear time-invariant(LTI)/linear shift-
invariant(LSI) system. They are basically
equivalent: the linear time invariant systems
refers to an analog system and shift-invariant
system refers to a discrete-time system.
Linear Time Invariant (LTI)
Systems
Linearity – Linear system is a system that
possesses the property of superposition.

Time Invariance – A system is time

invariant if the behavior and characteristics
of the system are fixed over time.
Discrete – Time LTI systems
The Convolution Sum
Continuous– Time LTI systems :
The Convolution Integral
A similar approach can be drawn for
continuous time LTI systems and following
results can be derived.
Properties of LTI systems
Commutative
x[n] h[n] = h[n] x[n]
Distributive
x[n] (h1[n]+h2[n]) = x[n] h1[n] + x[n]
h2[n]
Associative
x[n] (h1[n] h2[n]) = (x[n] h1[n]) h2[n]
Stability for LTI Systems
Causality for LTI Systems
A system is casual if the output at any time
depends only on the values of the input at the
present time and in the past.

Therefore, for LTI systems, y[n] must not

depend upon x[k] for k > n.
Hence,
h[n] = 0 for n < 0
The Frequency Response Function
Recall for an LTI system: y(n) = h(n) ∗ x(n).
Suppose we inject a complex exponential into
the LTI system:

Note: we consider x(n) to be comprised of a

pure frequency of ω rad/s
The Frequency Response Function

Thus, y(n) = H(ω)x(n) when x(n) is a pure frequency.

The Frequency Response Function
LTI System Eigen function
Eigen function of a system:
1) an input signal that produces an output that
differs from the input by a constant multiplicative
factor
2) multiplicative factor is called the eigen value .

 Therefore, a signal of the form A ejωn is an eigen

function of an LTI system.
Example:
Determine the magnitude and phase of H(ω)
for the three-point moving average (MA)
system
Frequency Response of LTI Systems
Frequency Response of LTI Systems
D. Richard Brown III
Inverse Systems and Equalization
Inverse System Region of
Convergence
Another Example
Some Properties of Inverse
Systems
Inverse in Multirate Systems (part
1 of 2)
Inverse in Multirate Systems (part
2 of 2)
 References
 Digital Signal Processing, Principles, Algorithms, and Applications
by John G. Proakis, Dimitris G. Manolakis.
 A.V. Oppenheim and R.W. Schafer, Discrete-Time Signal
Processing, 3rd Edition, Pearson Higher Education Inc., 2010.

 R. Chassaing and D. Reay, Digital Signal Processing and

Applications with the TMS320C6713 and TMS320C6416 DSK, 2nd
Edition, Wiley IEEE Press, 2008.

 S.K. Mitra, Digital Signal Processing: A Computer-Based

Approach, third edition, McGraw-Hill Inc., New York, 2005.

 J.G. Proakis and D.G. Manolakis, Digital Signal Processing:

Principles, Algorithms, and Applications, fourth edition, Prentice
Hall

 https://www.slideshare.net/SWATIMISHRA24/z-transfrm-ppt
 https://nptel.ac.in/content/storage2/courses/112106175/Module
%201/Lecture%201.pdf
 https://www.slideshare.net/farizazahari50/lti-system
Thank You