Chapter 1

Digital Signal
Processing (ECE 306)

Prerequisite - Signals and Systems
(ECE 202)
Mekelle University – Mekelle Institute of Technology

Course Objective
 Introduce students to methods of discrete-time signals and

systems representation and analysis.
 Introduce design methods and realization structures of
discrete-time systems.
 Introduce signal processing applications using DFTs.
 Introduce spectral analysis using DFTs.
 Introduce FFT algorithms.
 Introduce FIR and IIR filter design techniques.
 Introduce 2D transformations.
 DSP applications
 Introduce simulation using MATLAB.
11/10/22 MIT - Digital Signal Processing 2

Orientation
Instructors
Textbooks
Content to be covered
Grading

Instructor
1. Yaecob Girmay
(M.Tech in Signal Processing and Communication)
 MIT- Academic building
Room number: 110
Telephone :
Mobile : +251910461215
Email: yaecob.girmay@mu.edu.et
2. Goitom Yisfa (M.Tech in Micro-Electronics)

Books
Reference textbooks
1. Digital Signal Processing- Oppenheim and Schaffer, PHI
2. Digital Signal Processing- W.D.Stanley
3. Digital filters analysis and design-A.Antoniou, TMH
4. Digital Signal Processing: A Computer-Based Approach
By Prof. Sanjit K. Mitra (3rdedition or 4thedition)
5. Digital Signal Processing Using Matlab
By Vijay K. Ingle, and John G. Proakis
6. The Scientist and Engineer's Guide to Digital Signal
Processing
▪ By Steven W. Smith (You can download this from www.dspguide.com/pdfbook.htm )

Contents to be covered
Unit One Introduction to digital signal processing, Discrete time
signals and sequences, linear shift invariant systems,
Stability and causality, Frequency domain representation
of discrete time signals and systems.
Unit Two Discrete Fourier series: Properties, DFS representation of
periodic sequences, Discrete Fourier transforms: Properties
of DFT, Linear convolution of sequences using DFT,
Computation of DFT, Fast Fourier transforms (FFT)-
Radix 2 decimation in time and decimation in frequency
FFT algorithms, For a composite number N , Goertzel
algorithm and Chirp-Z algorithm, Inverse FFT
Unit Applications of Z-transforms, Solution of difference
Three equations of digital filters, system function, state space
analysis of digital system, Stability criterion, Frequency
response of stable systems, Realization of digital filters-
direct, Canonic, linear phase structure, Cascade, lattice and
parallel forms.
Unit IIR digital filters: Analog filters approximations,
Four Butterworth and Chebyshev.
Design of IIR digital filters from analog filters,
Step and impulse invariance techniques,
Bilinear transformation methods,
Spectral transformations.
Unit FIR digital filters: Characteristics of FIR digital

Five filters, Frequency response,
Design of FIR digital filters,
Comparison of IIR and FIR filters.
Applications of FFT in spectral analysis and linear
filtering, Introduction to DSP hardware,
Applications of DSP.

Grading
• Mid Semester Exam – 30%
• Final exam – 40%
• Assessments – 30%
– Assignments
– Mini Project
– Quizzes

Introduction
 Signals and signal processing

 Operations of Signal Processing
 How Are Signals Processed?
 What is DSP?
 DSP Building Blocks
 Advantages of DSP over ASP
 History of DSP
 DSP At a Glance

Signals
 The world of science and engineering is filled with signals such as images
from remote space probes, voltages generated by the heart and brain,
and countless other applications.
 In a communication system, the word 'signal' is very commonly used.
Therefore we must know its exact meaning.
Mathematically, signal is described as a function of one or more
independent variables.
 Basically it is a physical quantity. It varies with some dependent or
independent variables.
 So the term signal is defined as "A physical quantity which contains some
information and which is function of one or more independent variables.”

Signals and Signal Processing
 Signal is flow of information.
 Information carried is mathematically represented

as function of independent variables such as time ,
Distance, position, temperature, Pressure.
 The objective of signal processing is to extract the

useful information carried by the signal.
 Method of information extraction: Depends on the
type of signal and the nature of the information
being carried by the signal.

Classification of Signals
 Continuos signal Vs. Discret signal

 Depends on the nature of the independent variables and Value of the
function defining the signal. (Continuous independent variables means
Continuous signal & vice a versa)
 Real signals Vs. complex signals
 Signal can be either real valued function or complex-valued function
 Scalar signals vs. vector signals
 Signal generated by a single source (called as scalar signal) or by
multiple sources (called a vector or multichannel signal)
 One dimensional signals vs. multi-dimensional signals
 Signal can be function of a single independent variable (called as 1-D) or
function of more than one independent variables (called as M-D)
 Deterministic signal vs. random signal
 Uniquely determined by a well-defined process or generated in a
random fashion.

Classification of Signals
 Continuous valued and discrete valued signals

 Periodic and non-periodic signals
 Even and odd signals
 Energy and power signals
 Every signal is having its own characteristics.

 The processing of signal mainly depends on the characteristics
of that particular signal. So, classification of signal is necessary.

Operation of Signal Processing
 Typical Signal Processing Operations

 Basic TD Operations :- Scaling
 Basic TD Operations :- Time Shifting
 Elementary TD Operations :- Addition
 Elementary TD Operations :- Product
 Elementary TD Operations :- Integration & Differentiation
 Complex Operation :- Filtering
 Generation of Complex Signals

Operations of Signal Processing
 An operation designed for
 Extracting
 Analyzing
 Enhancing / Manipulating
 Storing
 Reproducing
 Representing
 Transmitting
 Distinction between useful and unwanted information is
often subjective as well as objective
 Signal processing tends to be application dependent

How Are Signals Processed?
 Signals can be represented in domain of original
independent variables or in a transformed domain.
 Likewise, the information extraction process may be carried
out in the original domain of the signal or in transformed
domain.
 Analog signal processing  Digital signal processing

 Continuous in time & amplitude  Analog signal can be processed using
 Processed using electrical networks digital hardware like adders, multipliers,
containing active and passive circuit logic elements & special-purpose
elements processors
 Example: radio & television receiver  Needs to convert analog signals into a
form suitable for digital hardware :
Analog Signal digital signal
xa(t) Processing ya(t)

What is DSP?
 It is hard to get exact definition of DSP.
 Dictionary meanings of these words:
 Digital: Operating by the use of discrete signals to represent data
in the form of numbers.
 Signal: A variable parameter by which information is conveyed
through an electronic circuit.
 Processing: To perform operations on data according to
programmed instructions. This leads to a simple definition of DSP
 DSP is defined as changing or analyzing information which is
measured as discrete sequences of numbers.

What is DSP?
 DSP is concerned with the mathematical representation of the signal and
the algorithmic operation carried out on it.
 Converting a continuously changing waveform (analog) into a series of
discrete levels (digital) for extracting / handling / processing the
information contained in it .
 DSP is a discipline concerned with the acquisition, representation,
manipulation, and transformation of signals required in a wide range of
practical applications.
This course is concerned with the discrete time representation of signals and
their discrete-time processing.

DSP Building Blocks
xa(t) PrF ADC DSP DAC PoF ya(t)
 PrF (Prefilter or antialiasing filter)

▪ Conditions analog signal to prevent aliasing
 ADC (analog-to-digital converter)
▪ Produces a stream of binary number from analog signal
 DSP (digital signal processor)
▪ Heart of DSP
▪ General-purpose computer, special-purpose processor digital hardware
 DAC (digital-to-analog converter)
▪ Produces a staircase waveform from a sequence of binary numbers
 PoF (post filter)
▪ Smooth our staircase waveform into the desired analog signal

DSP Building Blocks
 Input signal : It is the signal generated from some transducer or from some
communication system. It may be biomedical signal like ECG or EEG.
Generally, input signal is analog in nature. It is denoted by x(t).
 Anti-aliasing filter : it is basically a low pass filter. It is used for the following
purposes.
 It removes the high frequency noise contain in input signal.
 As the name indicates; it avoids aliasing effect. That means it is used to band
limit the signal.
 Sample and hold circuit : this block takes the samples of input signal. It
keeps the voltage level of input signal relatively constant which is the
requirement of ADC.
Some times amplifiers are used to bring the voltage level of input signal up to
the required voltage level of ADC.
 Analog to digital converter (ADC) : this block is used to convert analog

signal into digital form. This is required because digital signal processor
accepts the signal which is digital in nature.
DSP Building Blocks
 Digital signal processor : It processes input signal digitally. In a simple
languages processing of input signal making modifying the signal as per
requirement.
For this purpose DSP processors like ADSP 2100 or TMS 320 can be used.
 Digital to analog converter (DAC) : The output of digital signal processor is

digital in nature. But, the required final output is analog in nature. So, to
convert digital signal into analog signal DAC is used.
 Reconstruction filter : Output signal of DAC is analog, that means it is a

continuous signal. But, it may contain high frequency components. Such high
frequency components are unwanted. To remove these components;
reconstruction filter is used.

History of DSP
 Prior to 1950’s:
 analog signal processing using electronic circuits or
mechanical devices
 1950’s:
 computer simulation before analog implementation, thus
cheap to try out
 1965:
 Fast Fourier Transforms (FFTs) by Cooley and Tukey –
make real time DSP possible
 1980’s:
 IC technology boosting DSP

Advantages of DSP Over ASP
 Versatility: Digital systems can be reprogrammed for other applications (where
programmable DSP chips are used) . Moreover, digital systems can be ported
to different hardware.
 Repeatability : Digital systems can be easily duplicated. These systems do not

depend upon component tolerances and temperature.
 Simplicity : It is easy to build any digital system as compared to an analog one.
 Accuracy : To design analog system; analog components like , resistors,

capacitors and inductors are used. The tolerance of these components reduce
accuracy of analog system. In case of DSP, much better accuracy is obtained.
 Remote processing : Analog signals are difficult to store because of problems

like noise and distortion. While digital signal can be easily stored on storage
media like magnetic tapes, disks etc. Thus, compared to analog signals; digital
signals can be easily transported. So, remote processing of digital signal can be
done easily.

Advantages of DSP Over ASP
 Implementation of algorithms: The mathematical processing algorithms can

be easily implemented in case of digital signal processing. But, such algorithms
are difficult to implement in case of analog signals.
 Can be easley upgraded: Because of the use of software; digital signal

processing systems can be easily upgraded compared to analog system.
 Compatibility : In case of digital systems; generally all applications need

standard hardware. Thus, operation of DSP system is mainly dependent on
software. Hence, universal compatibility is possible compared to analog
systems.
 Cheaper: In many applications; the digital systems are comparatively cheaper

than analog systems.

Limitations of DSP
 System complexity : The digital signal processing system, makes use of
converters. like ADC- and DAC. This increases the system complexity
compared to analog systems. Similarly in many applications; the time required
for this conversion is more.
 Bandwidth limitation : if input signal is having wide bandwidth then it demands
for high speed ADC. This is because, to avoid aliasing effect, the sampling rate
should be at least twice the band width. Thus, such signals require fast digital
signal processors. But, always there is a practical limitation in the speed of
processors and ADC.
 Power consumption : A typical digital signal processing chip contains more
than 4 x 105 transistors. Thus, power dissipation is more in DSP systems
compared to analog systems.
 For small applications digital signal processing systems are expensive

compared to analog systems.

Applications of DSP
 Image Processing: Pattern Recognition, Animation, Robotic Vision, Image
Enhancement, Image Reconstruction, Image Restoration, Segmentation, etc.
 Instrumentation and control: Spectral Analysis, Noise Reduction and Noise Reduction.
 Speech/Audio: Speech Recognition & Analysis, Equalization
 Biomedical: ECG Analysis, Scanners, Patient Monitoring
 Telecommunications: Echo Cancellation, Spread Spectrum, Data Communication.
 Military: Sonar Processing, Radar Processing, Secure Communication
 Consumer Applications: Digital Video and Audio, Television and Music Systems
 Industrial Applications: Robotics and Power line Monitoring
 Automotive Applications: Vibration Analysis, Voice Commands, Cellular Telephones

DSP at a Glance

Examples of Typical Signals
 Speech and music signals - Represent air pressure as
function of time at a point in space
 The speech signal is an example of a 1-D signal where
the independent variable is time

Analog Recording process and playback Process
 Sound waves are picked up by a microphone and converted to a small
analog voltage called the audio signal.
 The audio signal, which varies continuously to “mimic” the volume and
frequency of the sound waves, is amplified and then converted to a
magnetic field by the recording head.
 As the magnetic tape moves under the head, the intensity of the
magnetic field is recorded (“stored”) on the tape.
 As the magnetic tape moves under the read head, the magnetic field on
the tape is converted to an electrical signal, which is applied to a linear
amplifier.
 The output of the amplifier goes to the speaker, which changes the
amplified audio signal back to sound waves. The volume of the
reproduced sound waves is controlled by the amplifier.

Digital Recording process and playback Process
 The sound waves are converted to an electrical audio signal by the
microphone. The audio signal is amplified to a usable level and is applied to an
analog-to-digital converter.
• The amplified audio signal is converted into a series of numbers by the analog-
to-digital converter.
• The numbers representing the audio signal can be stored or manipulated by
software to enhance quality, reduce storage space, or add special effects.
• The digital data are converted into an analog electrical signal; this signal is then
amplified and sent to the speaker to produce sound waves.

 Electrocardiography (ECG) Signal - Represents the
electrical activity of the heart
 A typical ECG signal is shown below

 Black-and-white picture  Video signals
 Image signal, is example of 2-D  Consists of a sequence of
signal images, called frames,
 Independent variables are 2  It is function of 3 variables: 2
spatial variable & light spatial coordinates and time
intensity is represented as
function of them

 A color image signal is composed of three 2-D
signals representing three primary colors: red, green
and blue (RGB)
• The full color image obtained by displaying the previous 3

color components is shown below

Monochromatic picture
 Example of a monochrome picture
(a) The brightness at each point in space is a scalar function of
f (x, y) of the rectangular coordinates x and y.
(b) The brightness at a horizontal line at y = y0 is a function

s(x) = f (x, y = y0) of the horizontal space variable x, only.

Continuous Vs Discrete-Time Signal
 For a 1-D signal, independent variable is usually labeled as time
 Continuous time Signal  Discrete Time Signal

 Independent variable is continuous  Independent variable is discrete
 Defined at every instant of time  Defined at discrete instants of
 A continuous-time signal with a time & hence it is a sequence of
continuous amplitude is usually numbers
called an analog signal  Discrete-time signal with
 A continuous-time signal with discrete-valued amplitudes
discrete value amplitudes is represented by a finite number
usually called a quantized signal of digits is referred as digital
signal
 A discrete-time signal with
continuous valued amplitudes is
called a sampled-data signal
Graphical Representation
 Plots illustrating the graphical representation of
(a) continuous-time signals
(b) and (c)discrete-time signals
(d) digital signals

Digitization
 Digital signals and timing diagrams.
(a) Typical voltage assignments in digital system;
(b) (b) typical digital signal timing diagram.

Continuous Vs Discrete-Time Signal
Time t Time
t
Continuous Time Signal Sampled Data Signal

Continuous Time Discrete
Time t Time t
quantized boxcar signal Digital Signal

Representation of Signals
 Functional dependence of signal in its mathematical
representation is often explicitly shown
 For a continuous-time 1-D signal, the continuous
independent variable is usually denoted by t
▪ For example, u(t) represents a continuous time 1-D signal
 For a discrete-time 1-D signal, the discrete independent
variable is usually denoted by n
▪ For example, {v[n]} represents a discrete time 1-D signal
▪ Each member, v[n], of a discrete-time signal is called a sample
▪ Mostly, a discrete-time signal is generated by sampling a parent
continuous-time signal at uniform intervals of time
▪ If the discrete instants of time at which a discrete-time signal is
defined are uniformly spaced, the independent discrete variable n
can be normalized to assume integer values.
 In the case of a continuous-time 2-D signal, the 2

independent variables are the spatial coordinates,
usually denoted by x and y
 For example, the intensity of a black-and white image at
location (x,y) can be expressed as u(x,y).
 On the other hand, a digitized image is a 2-D

discrete-time signal, and its 2 independent variables
are discretized spatial variables, often denoted by m
and n
 Thus, a digitized image can be represented as v[m,n]
 A black-and-white video signal is a 3-D signal and

can be represented as u(x,y,t)
 A color video signal is a vector signal composed of 3

signals representing the 3 primary colors: red,
green, and blue.
 r ( x, y , t ) 
u ( x, y, t )   y ( x, y, t ) 
b( x, y, t ) 
Typical Signal Processing Operations
 Most signal processing operations
 In the case of analog signals are carried out in the time-domain
 In the case of discrete-time signals, both time-domain or frequency-
domain operations are usually employed
 Bascic Time-Domain Operations :- Two basic operations are
 Scaling:- Multiplication of a signal either by a positive or negative
constant (i.e Amplification or Attenuation)
 Time Shifting:-Generates a signal that is a delayed / advanced replica
of the original signal ( i.e Delay / advance)
 Elementary Time-Domain Operations :- Three elementary operations
(involving two or more signals to generate new signal) are
 Addition:-
 Product
 Integration and Differentiation:-
Basic Operations :- Scaling
 Scaling is simply the multiplication of a signal either
by a positive or negative constant.
 In the case of analog signals, the operation is usually
called amplification
 If magnitude of the multiplying constant is
▪ greater than 1, operation is called gain / Amplification
▪ less than 1, the operation is called attenuation
 If x(t) is an analog signal that is scaled by a constant α,

▪ then the scaling operation generates a signal
y(t) = α x(t)
Basic Operations :- Time Shifting
 The time shifting operation generates a signal that
is a replica of the original signal either delayed or
advanced in time
 For an analog signal x(t),
y (t )  x (t  t 0 )
is the signal obtained by delaying x(t) by the amount of time
t0 which is assumed to be a positive number
 If t0 is negative, then it is an advance operation

Elementary Operations :- Addition
 Many applications require operations involving two

or more signals to generate a new signal
 For example,
y(t)  x1(t)  x2(t)  x3(t)
 is the signal generated by the addition of three analog

signals x1(t), x2(t) and x3(t)
Elementary Operations :- Product
 The product of two signals x1(t) and x2(t) generates

a signal
y(t)  x1(t)  x2(t)
Elementary Operations :- Integ & Diff
 Integration of an analog signal x(t) generates a
signal y(t) t
y (t )   x( )d

Differentiation of an analog signal x(t) generates a
signal w(t) dx(t )
w(t ) 
dt
 The elementary operations discussed so far are also carried out on
discrete-time signals
 More complex operations are implemented by combining two or more

elementary or basic operations
Complex Operation :- Filtering
 Filtering is one of the most widely used complex signal
processing operations
 The system implementing this operation is called a filter
 A filter passes certain frequency components without any

distortion and blocks other frequency components
 The range of frequencies that is allowed to pass through the

filter is called the passband,
 The range of frequencies that is blocked by the filter is called

the stopband
Anolog Signal = Linear Filtering
 In most cases, the filtering operation for analog

signals is linear
 The filtering operation of a linear analog filter is
described by the convolution integral

y (t )   h(t   ) x( )d

 where
▪ x(t) is the input signal,
▪ y(t) is the output of the filter, and
▪ h(t) is the impulse response of the filter
Basic Filters
 Lowpass filter passes all low-frequency components below
certain specified frequency , called the cutoff frequency fc ,
and blocks all high-frequency components.
 Highpass filter passes all high-frequency components above
certain cutoff frequency fc and blocks all low-frequency
components .
 Bandpass filter passes all frequency components between 2
cutoff frequencies, fc1 and fc2 , where fc1 < fc2, and blocks all
frequency components below the frequency fc1and above the
frequency fc2.
 Bandstop filter blocks all frequency components between 2

cutoff frequencies, , fc1 and fc2 , where fc1 < fc2, and passes all
frequency components below the frequency fc1 and above
the frequency f
Other Types of Filters
 A filter blocking a single frequency component is

called a notch filter
 A multiband filter has more than one passband

and more than one stopband.
 A comb filter blocks frequencies that are integral

multiples of a low frequency.
Systems
In Merriam-Webster’s dictionary, a system is broadly defined as a “regularly

interacting or interdependent group of items forming a unified whole.”
In the context of signal processing, a system is defined as a process where a signal

called input is transformed into another signal called output.
Systems are classified based on the category of input and output signals.
a) Continuous-time systems
A continuous-time system is a system which transforms a continuous-time input

signal x(t) into a continuous-time output signal y(t).
b) Discrete-time systems
A system that transforms a discrete-time input signal x[n] into a discrete-time output
signal y[n], is called a discrete-time system.

Systems Block diagram
 Pictorial or block-diagram representation of a

(a) continuous-time system
(b) discrete-time system

Sampling and Quantization
 Conceptually, the conversion of an analog (continuous time,

continuous-amplitude) signal into a digital (discrete-time,
discrete-amplitude) signal,
It is a simple process; it consists of two parts: sampling and
quantization.
Sampling converts a continuous-time signal to a discrete-time signal
by measuring the signal value at regular intervals of time.
Quantization converts a continuous-amplitude x into a discrete-
amplitude xd.
The result is a digital signal that is different from the discrete-time
signal by the quantization error or noise.

ADC
(a) Block diagram representation of the analog-to-digital conversion process.

(b) Examples of the signals x(t), x[n], and xd[n] involved in the process.
The amplitude of x[n] is known with infinite precision, whereas the amplitude of
xd[n] is known with finite precision delta(quantization step or resolution).

Block diagram showing ADC
 Block diagram representation of an ideal (a) and a practical (b) analog-to-
digital converter, and the corresponding input and output signals.
 The input to the ideal ADC is a function and the output is a sequence of
numbers; the input to the practical ADC is an analog signal and the output is a
sequence of binary code words.
 The number of bits B, in each word, determines the accuracy of the converter.
 As the number of bits B increases, the accuracy of delta, the quantizer
increases, and the difference between discrete-time and digital signals
diminishes. For this reason, we usually refer to the sampler as an ideal analog-
to-digital (A/D) converter.

Digital-to-analog conversion
The conversion of a discrete-time signal into continuous time form is done with an
interface system called digital-to-analog (D/A) converter (DAC).
The ideal D/A converter or interpolator is essentially filling the gaps between the
samples of a sequence of numbers to create a continuous-time function
A practical DAC takes a value represented in digital code and converts it to a
voltage or current that is proportional to the digital value.
 Summary Based on the type of input and output signal, there are three classes
of practical system: analog systems, digital systems, and analog-digital
interface systems.
ADC-DAC
 The three classes of system: analog systems, digital
systems, and interface systems from analog-to-digital
and digital-to-analog.

ASP
 Processing (ASP) is concerned with the conversion of analog
signals into electrical signals by special transducers or sensors
and their processing by analog electrical and electronic
circuits.
 The output of the sensor requires some form of conditioning,
usually amplification, before it can be processed by the
analog signal processor.
 Fig. Simplified block diagram of an analog signal processing

system.

Examples of digital signal processing applications and algorithms.

Simplified block diagram of a digital cellular phone

Discrete-time signal representations

Representation of a sampled signal.

Energy and power
 The energy of a sequence x[n] is defined by the formula:
 Similarly, the power of a sequence x[n] is defined as the average energy per
sample
 When x[n] represents a physical signal, both quantities are directly related
to the energy and power of the signal.
 Finite duration sequences have finite energy but zero power. However,
when the duration of a sequence increases, the energy or power may or
may not remain finite.

Sequences
 Unit sample sequence The simplest discrete-time signal is the unit sample
or unit impulse sequence, defined by
 Unit step sequence The unit step sequence is given by
and can be thought of as an infinite succession of unit samples starting at n = 0.

Sequences
 Sinusoidal sequence: the real sinusoidal sequence has the general form
 where A (amplitude) and φ (phase) are real constants. The quantity ω0 is the
frequency of the sinusoid and has units of radians per sampling interval. The
values of this sequence keep on oscillating between ±|A|.
 Exponential sequence : the exponential sequence has the general form

defined by
 where A and a can take real or complex values.

 If both A and a are real numbers in then x[n] is termed as a real exponential
 sequence. For −1 < a < 1 (a > 1 or a < −1) the absolute value |x[n]| of the
sequence decreases (increases) in magnitude with increasing n for
 0 < a < 1 and −1 < a < 0). The values of x[n] alternate in sign when a is negative.

Discrete-time signals.
 Examples of a discrete-time sinusoidal signal (a), and two real exponential
sequences (b).

Sequences
 Complex sinusoidal sequence One special case of the exponential sequence is
when A is real-valued but
 Periodic sequence: a sequence x[n] is called periodic if
 The smallest value of N is known as the fundamental period or simply period of

x[n].
 The sinusoidal sequence is periodic, if cos(ω0n+φ) = cos(ω0n+ω0N +φ). This is
possible if ω0N = 2πk, where k is an integer. When k and N are prime numbers, N
is equal to the number of samples in one fundamental period of the sequence.

Operations on sequences
 Addition, subtraction, multiplication, division, and scaling of sequences
can be performed on a sample-by-sample basis:
 The operations of shifting and folding are not commutative. Indeed, we have

Cusuality
 A system is called causal if the present value of the output does not depend
on future values of the input, that is, y[n0] is determined by the values of x[n] for
n ≤ n0, only.
 If the output of a system depends on future values of its input, the system is
noncausal.
 Causality implies that if x[n] = 0 for n < n0, then y[n] = 0 for n < n0; that is, a
causal system cannot produce an output before the input is applied.
 describes a three-point moving average filter, which is often used to smooth a
signal corrupted by additive noise, for all values of n.
 But, the five-point median filter, used to remove spikes from experimental
data, which is defined by
 This system is non causal because the input samples x[n+1] and x[n + 2] are not
available when the output sample y[n] needs to be computed. This non causal
system can be implemented in real-time if we delay the generation of its
output by two sampling intervals, that is, compute y[n] at time t = (n+2)T.

Stability
 A system is said to be stable, in the Bounded-Input Bounded-Output (BIBO)
sense, if every bounded input signal results in a bounded output signal, that is
 A signal x[n] is bounded if there exists a positive finite constant Mx such that |
x[n]| ≤ Mx for all n.
 Since unstable systems generate unbounded output signals, that is overflows,

they cannot be used in practical applications.

Linearity and time invariance
 Stability is a property that should be satisfied by every practical system,
whereas causality is required for systems that should operate in real-time.
However, the properties that make the analysis of discrete-time systems
mathematically tractable are linearity and time-invariance.
 A system is called linear if and only if for every real or complex constant a1, a2
and every input signal x1[n] and x2[n]
for all values of n.

 If the output y[n] for every n depends only on the input x[n] at the same time,
the system is said to be memory less; otherwise it is said to be dynamic.
However, we emphasize that the practical implementation of a memory less
system, like y[n] = 2x2[n], requires memory to store the multiplying factor 2
and the value of x[n].

Time-invariant
 If the characteristics of a system do not change with time, the system is called
time invariant; otherwise it is called time-varying.
 This means that the shape of the output of a time-invariant system depends
only on the shape of the input signal and not on the time instant the input was
applied into the system.
 i.e. a system is called time-invariant or fixed if and only if
 for every input x[n] and every time shift n0. That is, a time shift in the input
results in a corresponding time shift in the output.
 In summary, linearity means that the output due to a sum of input signals
equals the sum of outputs due to each signal alone. Time-invariance means
that the system does not change over time.

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Digital Signal

Processing (ECE 306)

Mekelle University – Mekelle Institute of Technology

 Introduce students to methods of discrete-time signals and

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2. Goitom Yisfa (M.Tech in Micro-Electronics)

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Unit FIR digital filters: Characteristics of FIR digital

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• Mid Semester Exam – 30%

• Final exam – 40%

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 Signals and signal processing

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 Information carried is mathematically represented

 The objective of signal processing is to extract the

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 Continuos signal Vs. Discret signal

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 Continuous valued and discrete valued signals

 Every signal is having its own characteristics.

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 Typical Signal Processing Operations

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 Analog signal processing  Digital signal processing

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 PrF (Prefilter or antialiasing filter)

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 Analog to digital converter (ADC) : this block is used to convert analog

 Digital to analog converter (DAC) : The output of digital signal processor is

 Reconstruction filter : Output signal of DAC is analog, that means it is a

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 Repeatability : Digital systems can be easily duplicated. These systems do not

 Simplicity : It is easy to build any digital system as compared to an analog one.

 Accuracy : To design analog system; analog components like , resistors,

 Remote processing : Analog signals are difficult to store because of problems

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 Implementation of algorithms: The mathematical processing algorithms can

 Can be easley upgraded: Because of the use of software; digital signal

 Compatibility : In case of digital systems; generally all applications need

 Cheaper: In many applications; the digital systems are comparatively cheaper

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 For small applications digital signal processing systems are expensive

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 Speech/Audio: Speech Recognition & Analysis, Equalization

 Biomedical: ECG Analysis, Scanners, Patient Monitoring

 Telecommunications: Echo Cancellation, Spread Spectrum, Data Communication.

 Military: Sonar Processing, Radar Processing, Secure Communication

 Industrial Applications: Robotics and Power line Monitoring

 Automotive Applications: Vibration Analysis, Voice Commands, Cellular Telephones

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 A typical ECG signal is shown below