Course Title: IE471 Digital Signal Processing

[ Cr. Hrs = 3 + 1 ] [ Marks: 100 + 50 ]

Teacher: Dr. Farah Haroon; far_iiee@yahoo.com; farah@iiee.edu.pk

COURSE OBJECTIVES:
To provide the students with knowledge of both the theory and practical applications of digital
signal processing.

COURSE OUTLINE:
Review of: Sampling & Reconstruction, Quantization, Relationship between sampling
frequency and Shannon’s theorem, Continuous time and discrete time signals.
Discrete-Time Systems: Input/output Rules, Impulse Response, FIR and IIR filters, Causality
Stability, Z-transform, Inverse Z-transform, Discrete Fourier transform, Fast Fourier transform.
Digital Filter Design, Implementation techniques for sample-by-sample for real time DSP
application.
FIR Filtering and Convolution: Block Process Methods, Convolution, Direct Form,
Convolution Table, LTI Form, Matrix Form, Transient & Steady State Behaviour, Convolution
of Infinite Sequence.Sample Processing Methods, Pure Delays, FIR Filtering in Direct Form.
Transfer Functions: Equivalent Description of Digital Filters, Transfer Functions, Sinusoidal
Response, Poles/Zeros Design, De-convolution, Inverse Filters and Stability.
DFT/FFT Algorithm: Frequency Resolution and Windowing, DTFT, Inverse DFT, FFT and
Fast Convolution.

LAB OUTLINE:
Labs of the course consist of experiments that confirm the basic concepts of DSP. Initial labs
will be carried out on MATLAB and then students will be assigned to develop DSP system
(software and hardware), by integrating subsystems or using trainer of DSP.

TEXT BOOK:
1. Introduction to Signal Processing, by Sophocles J. Orfanidis, Prentice Hall, Inc. Latest Ed.

RECOMMENDED BOOKS:
1. Discrete-Time Signal Processing by A. V. Oppenheim and R. W. Schafer.
2. Digital Signal Processing: Principles, Algorithms, and Applications by J. G. Proakis and D. G.
Manolakis.
Introduction

Today, Digital Signal Processor (DSP) chips are used in a vast number of
different commercial and consumer applications. Examples of some typical
signals that can be manipulated by Digital Signal Processing (DSP) tech-
niques are:

1. Electrocardiography (ECG) Signal

• represents the electrical activity of heart
• one period of ECG represents one cycle of the blood transfer process

2. Electroencephalogram (EEG) Signal

• represents the summation of the electrical activity of individual neurons
in the brain

3. Speech Signal
• formed by exciting the human vocal tract
• composed of two types of sounds: voiced and unvoiced

4. Musical Sounds
• generally produced by mechanical vibrations of the instrument

5. Time Series
• yearly average of number of sunspots, daily stock prices, the annual yield
per km2 of crops etc.

6. Seismic Signals
• reflections activated by seismic sources, such as high-energy explosives

7. Images
• two-dimensional signals, e.g. photographs, still video images and x-rays

8. Modulation Signals
• signals used in digital modulation methods, usually complex envelopes
of bandpass signals

1
Most naturally occurring signals are analog signals, and in many applications
of DSP the final result of processing is also an analog signal.

⇒ Why it is desirable to use DSP instead of analog electronics?

Notable advantages of Digital Signal Processing are:

Programmability
• A hardware implementation that is based on a Digital Signal Processor
can be modified or upgraded simply by replacing the software (a memory
chip), whereas modifications of analog implementations require changes
in the hardware, which in many cases leads to complete redesign of the
whole system.

Resistance to Changes in External Parameters

• Analog components (resistors, capacitors, operational amplifiers etc.)
change their characteristics with changes in temperature, aging and other
external parameters. Digital circuitry is relatively insensitive to all of
these parameters.

Repeatability and Reproducibility

• DSP implementations are exactly repeatable: for same digital input dif-
ferent units produce exactly same digital output. The outputs of different
units of analog circuitry always produce slightly different outputs due to
variations of the basic analog components. For this reason digital imple-
mentations can also be reproduced easily in large quantities without any
adjustments in production or in installation.

Easy Implementation of Adaptive Algorithms

• Digital implementation permits easy adjustment of parameters of the
signal processing algorithm. Since parameters, like filter coefficients,
are just stored numbers in memory, they can be easily recalculated, when
adaptive behavior is required. Changing of filter coefficients for an ana-
log filter is very complicated and in many cases practically impossible.

2
Error Correction and Detection Capabilities
• Digital signals can be stored or transmitted without any loss of informa-
tion. If the data storage or the transmission medium is unreliable (i.e. it
can create bit errors), error detection and error correction techniques can
be easily adopted for digital signals. Corresponding functions are again
practically impossible to implement for analog signals.

Special Functions
• Digital implementation enables realization of certain signal processing
functions, which are impossible in analog implementation. Examples of
such functions are linear phase filters, quadrature mirror filter banks etc.

Digital Signal Processing naturally has also some disadvantages, the most
important disadvantages of Digital Signal Processing are:

System Complexity
• For systems, where the input and output signals are analog signals, a
DSP implementation inevitably requires A/D and D/A converters with
their associated filters. Thus, simple signal processing functions can be
implemented with reduced system complexity by using analog process-
ing. The reduced system complexity also leads to reduced cost, reduced
power consumption, and reduced size, which are critical parameters in
many applications.

Bandwidth Limitation
• There is an inherent bandwidth limitation for Digital Signal Processing,
since the sampling rates of A/D and D/A converters as well as cycle
times of Digital Signal Processors, themselves, are limited by the state-
of-the-art of the technology. Currently, sampling frequency around
100-500 MHz possible.

3
I. Discrete-Time Signals and Systems

I.1. Discrete-Time Signals

I.1.1. Basic Definition

Signals are represented mathematically as functions of one or more inde-

pendent variables. The independent variable of a signal can be either con-
tinuous or discrete. In the first case we say the signal is continuous-time;
in the second discrete-time. A continuous-time signal is defined at every
instant of time, whereas a discrete-time signal is defined only at the set of
discrete time instants. A discrete-time signal is not defined at all at instants
between two successive samples.

Besides the independent variable (i.e. time) being either continuous or dis-
crete, the signal amplitude may be either continuous or discrete. When both
amplitude and time are continuous we call the signal analog. These are the
kind of signals occurring in the physical world, as discussed in introductory
physics courses. Such signals cannot be directly accessed by a digital com-
puter.

When the values of the signal are defined only at the discrete points in time,
and the values themselves are coded with a finite number of bits, the signal
is called a digital signal. These kinds of signals can be stored in the digital
storage media and used in digital computers.

Amplitude
Time Continuous Discrete
Continuous Analog "Quantized boxcar"
Discrete Sampled-data Digital

We will refer to x[n] as the "nth sample" of the discrete-time signal even if
the signal is not obtained by sampling an analog signal. If, indeed, the signal
is obtained from sampling of analog signal xa(t), then x[n]=xa(nT), where T
is the sampling period (i.e. the time interval between successive samples).
Analog signal

Continuous tim e, t

Sampled-data signal

Discrete tim e, n

"Quantized boxcar" signal

Continuous tim e, t

Digital signal

Discrete tim e, n
I.1.2. Some Elementary Discrete-Time Signals

1. The unit sample is denoted by δ[n] and is defined as

0, n ≠ 0
δ[n ]= 
1, n = 0

δ [n ]
1

0
-2 0 -1 5 -1 0 -5 0 5 10 15 20
n

This signal is often referred to as a discrete-time impulse, or simply as an

impulse. The important property of impulse is that an arbitrary discrete-time
signal can be represented as a sum of scaled, delayed impulses as

∞
x[n]= ∑ x[k ] δ[n − k ]
k =−∞

2. The unit step is denoted as u[n] and is defined as

0, n < 0
u[n ]= 
1, n ≥ 0

u [n ]
1

0
-2 0 -1 5 -1 0 -5 0 5 10 15 20
n
The unit sample and unit step are related to each other by

n
u[n ]= ∑ δ[k ]
k = −∞

δ[n ]=u[n ] − u[n − 1]

i.e., the value of the unit step at index n is equal to the accumulated sum of
the value at index n and all previous values of the unit sample, and the value
of unit sample is equal to the first backward difference of the unit step.
These relationships correspond respectively to integration and derivation
in the continuous-time case.

It should be noted, that while the continuous-time impulse function is only

a mathematical abstraction, and as such physically unrealizable, the discrete-
time impulse function is a perfectly admissible discrete-time signal, which
can be easily generated, for example within a digital signal processor.

The discrete-time unit step function can be perceived just as a sampled

version of the continuous-time unit step function. Although the discrete-time
impulse function plays the same role to that played by the continuous-time
impulse function, the discrete-time impulse function can not be interpreted
as samples of the continuous-time impulse function.

3. The complex exponential sequence is defined as

x[n ]=Ae jω n = A[cos(ω0 n ) + jsin( ω0 n )]

0

Since this sequence is complex-valued, separate plots of the real and imagi-
nary parts are required to illustrate it.
The following is a plot of a complex exponential sequence for ω0 = π/10.

Re{Aejω0n}
A

-A
-20 -15 -10 -5 0 5 10 15 20
n

Im{Aejω0n}
A

-A
-20 -15 -10 -5 0 5 10 15 20
n

A sinusoidal sequence is just the real part of the complex exponential

x[n] = A cos(ω0n)

A discrete-time signal x[n] is periodic with period N if

x[n] = x[n+N], for all n.

For sinusoidal sequence this means that

A cos(ω0n) = A cos(ω0n + ω0N),

which requires that

2π
ω0N = 2πk, or N = k .
ω0

Same result holds for the complex exponential. Thus, complex exponential
and sinusoidal sequences are not necessarily periodic sequences. Depending
on the value of ω0, there may or may not be integer values k and N to fulfill
the periodicity condition. For example, for ω0 = 1 no such integer values can
be found, and the sequence x[n] = A cos(n) is not periodic.

I.1.3. Representation of Signals

For a complete representation of a discrete-time signal, it must defined

at every discrete time instant from minus infinity to infinity. Discrete-time
signals can represented in multiple ways: by using graphical, functional
or sequence representation or as a weighted sum of (delayed) elementary
signals.

The previously presented definitions of

unit sample and sinusoidal sequence

0, n ≠ 0
δ[n ]=  x[n] = A cos(ω0n)
1, n = 0

are examples of functional representations.

The previously presented figure of the unit sample

δ [n ]
1

0
-2 0 -1 5 -1 0 -5 0 5 10 15 20
n

is an example of a graphical representation of a discrete-time signal. Natu-

rally, only finite-duration signals can be completely represented by using
graphical representation.

The unit sample can be depicted in sequence representation as

x[n] = {…, 0, 0, 1, 0, 0, …}
↑

where the symbol ↑ indicates the time origin (n = 0). As graphical represen-
tation, also the sequence representation can completely represent only finite-
duration signals.

Arbitrary sequences can be represented as a weighted sum of delayed ele-

mentary signals. The unit sample and unit step are commonly used elemen-
tary signals. The following finite-duration signal

x[n] = {…, 0, 0, 0.3, 0, 0, 1.7, -0.8, 0, 0.65, 0, 0, …}

↑

can be represented as a weighted sum of delayed unit samples as

x[n] = 0.3 δ[n+2] + 1.7 δ[n-1] – 0.8 δ[n-2] + 0.65 δ[n-4].

I.2. Discrete-Time Systems

A discrete-time system is defined mathematically as a transformation or op-

erator that maps an input sequence with values x[n] into an output sequence
with values y[n]. This is denoted as

y[n] = T{x[n]}.

I.2.1. Basic Operations on Sequences

The mapping from an input sequence into an output sequence can in most
cases be represented as a combination of basic operations of addition, multi-
plication, multiplication by a constant, delay and time reversal.

In addition, a new sequence z[n] is obtained by adding the sample values of

two sequences x[n] and y[n] at each discrete time instant n. For example

x[n] = {0.9, 2.0, 4.7, 4.6, 2.1, 4.5, 0.3}

↑

+ y[n] = {1.8, 4.1, 0, 0.7, 1.0}

↑

= z[n] = {2.7, 6.1, 4.7, 5.3, 3.1, 4.5, 0.3}

↑

In multiplication, a new sequence z[n] is obtained by forming the product

the sample values of two sequences x[n] and y[n] at each discrete time in-
stant n. For example

x[n] = {0.9, 2.0, 4.7, 4.6, 2.1, 4.5, 0.3}

↑

× y[n] = {1.8, 4.1, 0, 0.7, 1.0}

↑

= z[n] = {1.62, 8.2, 0, 3.22, 2.1}

↑
In multiplication by a constant, a new sequence z[n] is obtained by multi-
plying each sample of a sequence by a scalar value. For example

z[n] = 1.5 y[n] = {2.7, 6.15, 0, 1.05, 1.5}

↑

In delay, a new sequence z[n] is obtained by shifting each sample of a se-

quence to the left (right for negative delays) by a integer number of samples.
For example

z[n] = x[n-2] = {0.9, 2.0, 4.7, 4.6, 2.1, 4.5, 0.3}

↑

In time reversal, a new sequence z[n] is obtained by moving each sample of

a sequence to the corresponding negative discrete time instant. For example

z[n] = x[-n] = {0.3, 4.5, 2.1, 4.6, 4.7, 2.0, 0.9}

↑

All these basic operations require that the operand(s) and the result must
have the same sampling period. Later, in the Section III.4 we shall also
examine essential sampling period conversion operations.
SAMPLING AND RECONSTRUCTION

SAMPLING THEOREM
The analog signal x(t) is periodically measured every T seconds. Thus, time is discretized in
units of the sampling interval
 Fig: Ideal Sampler

Every frequency component of the original signal is periodically replicated over the entire
frequency axis, with period given by the sampling rate:

Fig Spectrum realization of sinusoid with single frequency f, sampled at fs

Fig: Signal 2 is oversampled

Fig: Typical band limited signal
ANTIALIASING PREFILTERS