UNIT -1

Introduction to Digital Signal Processing

DSP is a technique of performing the mathematical operations on the signals in digital domain.
As real time signals are analog in nature we need first convert the analog signal to digital, then we
have to process the signal in digital domain and again converting back to analog domain. Thus ADC is
required at the input side whereas a DAC is required at the output end. A typical DSP system is as
shown in figure 1.1.
• The main function of low pass ant aliasing filter is to band limit the input signal to the
folding frequency without distortion.
• It should be noted that even if the signal is band limited, there is always wide-band
additive noise which will be folded back to create aliasing.
• When an analog voltage is connected directly to an ADC, the conversion
process can be adversely affected if the voltage is changing during the
conversion time.
• The quality of conversion process can be improved by using sample and hold circuit

Advantages of DSP
 Programmability: software digital signal processes can be quickly modified, in contrast to analog
circuits, which must be physically rearranged.
 Versatility: Flexible and easy to upgrade.
 Stability: Less sensitive environmental changes such as electromagnetic interference.

3
 Need for DSP

Analog signal Processing has the following drawbacks:

 They are sensitive to environmental changes
 Aging
 Uncertain performance in production units
 Variation in performance of units
 Cost of the system will be high
 Scalability
If Digital Signal Processing would have been used we can overcome the above shortcomings of
ASP.
REVIEW OF DISCRETE TIME SIGNALS AND SYSTEMS
Anything that carries some information can be called as signals. Some examples are
ECG, EEG, ac power, seismic, speech, interest rates of a bank, unemployment rate of
a country, temperature, pressure etc.
A signal is also defined as any physical quantity that varies with one or more independent
variables.
A discrete time signal is the one which is not defined at intervals between two successive
samples of a signal. It is represented as graphical, functional, tabular representation and
sequence.
Some of the elementary discrete time signals are unit step, unit impulse, unit ramp,
exponential and sinusoidal signals (as you read in signals and systems).

Classification of discrete time signals

Energy and Power signals

If the value of E is finite, then the signal x(n) is called energy signal.

If the value of the P is finite, then the signal x(n) is called Power signal.

Periodic and Non periodic signals

A discrete time signal is said to be periodic if and only if it satisfies the condition X
(N+n) =x (n), otherwise non periodic

Symmetric (even) and Anti-symmetric (odd) signals

The signal is said to be even if x(-n)=x(n) The
signal is said to be odd if x(-n)= - x(n)

Causal and non causal signal

The signal is said to be causal if its value is zero for negative values of ‘n’.

Some of the operations on discrete time signals are shifting, time reversal, time scaling,
signal multiplier, scalar multiplication and signal addition or multiplication.

Discrete time systems

A discrete time signal is a device or algorithm that operates on discrete time signals
and produces another discrete time output.

Classification of discrete time systems

Static and dynamic systems
A system is said to be static if its output at present time depend on the input at present
time only.
 Causal and non causal systems
A system is said to be causal if the response of the system depends on present and past
values of the input but not on the future inputs.

Linear and non linear systems

A system is said to be linear if the response of the system to the weighted sum of
inputs should be equal to the corresponding weighted sum of outputs of the systems. This
principle is called superposition principle.

Time invariant and time variant systems

A system is said to be time invariant if the characteristics of the systems do not
change with time.

Stable and unstable systems

A system is said to be stable if bounded input produces bounded output only.

TIME DOMAIN ANALYSIS OF DISCRETE TIME SIGNALS AND SYSTEMS

Representation of an arbitrary sequence
Any signal x(n) can be represented as weighted sum of impulses as given below

The response of the system for unit sample input is called impulse response of the system
h(n)

By time invariant property, we have

The above equation is called convolution sum.

Some of the properties of convolution are commutative law, associative law and
distributive law.

Correlation of two sequences

It is basically used to compare two signals. It is the measure of similarity between two
signals. Some of the applications are communication systems, radar, sonar etc.
The cross correlation of two sequences x(n) and y(n) is given by
One of the important properties of cross correlation is given by

The auto correlation of the signal x(n) is given by

Linear time invariant systems characterized by constant coefficient difference

equation
The response of the first order difference equation is given by

The first part contain initial condition y(-1) of the system, the second part contains input
x(n) of the system.
The response of the system when it is in relaxed state at n=0 or
y(-1)=0 is called zero state response of the system or forced response.

The output of the system at zero input condition x(n)=0 is called zero input response of
the system or natural response.

The impulse response of the system is given by zero state response of the system

The total response of the system is equal to sum of natural response and forced responses.
 UNIT-II
Discrete Fourier Transform

Discrete Fourier Series

The Fourier series representation o f a continuous-time periodic signal can consist of an

infinite number of frequency components, where the frequency spacing between two successive
harmonically related frequencies is 1 / T p, and where Tp is the fundamental period.
Since the frequency range for continuous-time signals extends infinity on both sides it is
possible to have signals that contain an infinite number of frequency components.
In contrast, the frequency range for discrete-time signals is unique over the interval. A
discrete-time signal of fundamental period N can consist of frequency components separated by 2n /
N radians.
Consequently, the Fourier series representation o f the discrete-time periodic signal will
contain at most N frequency components. This is the basic difference between the Fourier series
representations for continuous-time and discrete-time periodic signals.
PROPERTIES OF DFT:

LINEAR FILTERING METHODS BASED ON THE DFT

Since the D F T provides a discrete frequency representation o f a finite-duration Sequence in

the frequency domain, it is interesting to exp lore its use as a computational tool for linear system
analysis and, especially, for linear filtering. We have already established that a system with
frequency response H { w ) y w hen excited with an input signal that has a spectrum possesses an
output spectrum.
The output sequence y(n) is determined from its spectrum via the inverse Fourier transform.
Computationally, the problem with this frequency domain approach is that are functions o f the
continuous variable. As a consequence, the computations cannot be done on a digital computer, since
the computer can only store and perform computations on quantities at discrete frequencies.
On the other hand, the DFT does lend itself to computation on a digital computer. In the discussion
that follows, we describe how the DFT can be used to perform linear filtering in the frequency
domain. In particular, we present a computational procedure that serves as an alternative to time-
domain convolution.
In fact, the frequency-domain approach based on the DFT, is computationally m ore efficient
than time-domain convolution due to the existence of efficient algorithms for computing the DFT .
These algorithms, which are described in Chapter 6, are collectively called fast Fourier transform
(FFT) algorithms.
 FAST FOURIER TRANSFORM
In this section we represent several methods for computing dft efficiently. In view of the
importance of the DFT in various digital signal processing applications such as linear filtering,
correlation analysis and spectrum analysis, its efficient computation is a topic that has received
considerably attention by many mathematicians, engineers and scientists. Basically the computation
is done using the formula method.

Divide-and-Conquer Approach to Computation of the DFT

The development of computationally efficient algorithms for the DFT is made possible if we
adopt a divide-and-conquer approach. This approach is based on the decomposition of an N-point
DFT into successively smaller DFT. This basic approach leads to a family o f computationally
efficient algorithm s know n collectively as FFT algorithms.
T o illustrate the basic notions, let us consider the computation of an N point DFT , where N can be
factored as a product of two integers, that is, N = L M
 An additional factor of 2 savings in storage of twiddle factors can be obtained by introducing
a 90° phase offset at the mid point of each twiddle array , which can be removed if necessary at the
ouput of the SRFFT computation. The incorporation of this improvement into the SRFFT results in
an other algorithm also due to price called the PFFT algorithm.

Implementation of FFT Algorithms

Now that w e has described the basic radix-2 and radix -4 F FT algorithm s, let us consider
some of the implementation issues. Our remarks apply directly to
 UNIT -III
IIR Digital Filters
IIR FILTER DESIGN
STRUCTURES FOR IIR SYSTEMS
In this section we consider different IIR system s structures described by the difference equation
given by the system function. Just as in the case o f FIR system s, there are several types o f
structures or realizations, including direct-form structures, cascade-form structures, lattice structures,
and lattice-ladder structures. In addition, IIR systems lend them selves to a parallel form realization.
We begin by describing two direct-form realizations.

DIRECT FORM STRUCTURES:

DIRECT FORM II

Signal Flow Graphs and Transposed Structures

A signal flow graph provides an alternative, N but equivalent, graphical representation to a

block diagram structure that we have been using to illustrate various system realizations. T he basic
elements o f a flow graph are branches and nodes. A signal flow graph is basically a set o f directed
branches that connect at nodes. By definition, the signal out of a branch is equal to the branch gain
(system function) times the signal into the branch. Furthermore, the signal at anode o f a flow graph
is equal to the sum o f the signals from all branches connecting to the node.
Cascade-Form Structures
Let us consider a high-order IIR system with system function given by equation. Without loss o
f generality we assume that N > M . T h e system can be factored into a cascade o f second-
order subsystem s, such that H (z) can b e expressed as
Parallel-Form Structures
A parallel-form realization o f an IIR system can be obtained by performing a partial-fraction
expansion o f H( z) . Without loss o f generality, w e again assume that N > M and that the poles are
distinct. Then, by performing a partial-fraction expansion o f H( z ), we obtain the result
The realization of second order form is given by

The general form of parallel form of structure is f\given by

Lattice and Lattice-Ladder Structures for IIR Systems

 UNIT – IV
FIR DIGITAL FILTERS
The transfer function is obtained by taking Z transform of finite sample impulse response. The filters
designed by using finite samples of impulse response are called FIR filters.
Some of the advantages of FIR filter are linear phase, both recursive and non recursive, stable and
round off noise can be made smaller.
Some of the disadvantages of FIR filters are large amount of processing is required and non integral
delay may lead to problems.
DESIGN OF FIR FILTERS
STRUCTURES FOR FIR SYSTEMS

Direct-Form Structure
The direct form realization follows immediately from the non recursive difference equation given
below

Cascade-Form Structures
The cascaded realization follows naturally system function given by equation. It is simple matter to
factor H(z) into second order FIR system so that
Frequency-Sampling Structures
The frequency-sampling realization is an alternative structure for an FIR filter in which the
parameters that characterize the filter are the values o f the desired frequency response instead of the
impulse response h(n). To derive the frequency sampling structure, we specify the desired frequency
response at a set o f equally spaced frequencies, namely

The frequency response of the system is given by

Lattice Structure
In this section w e introduce another F IR filter structure, called the lattice filter or
Lattice realization. Lattice filters are used extensively in digital speech processing
And in the implementation of adaptive filters. Let us begin the development by considering a
sequence of FIR filters with system functions
The general form of lattice structure for m stage is given by’

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UNIT – V
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