MUTHAYAMMAL ENGINEERING COLLEGE

(An Autonomous Institution)

(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-1

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Introduction to DSP

Introduction :
 DSP manipulates different types of signals with the intention of filtering, measuring, or
compressing and producing analog signals.
 An analog-to-digital converter is needed in the real world to take analog signals (sound,
light, pressure, or temperature) and convert them into 0's and 1's for a digital format
Prerequisite knowledge for Complete understanding and learning of Topic:
( Max. Four important topics)
1.Signals and System
2.Fourier Transform
3.Z-Transform
Introduction

Signal:

A signal is defined as any physical quantity that varies with time, space, or any other in
dependent variable or variables. Mathematically, we describe a signal as a function of one
or mo re independent variables. For example, the functions s(t)= 5t
describe a signal, one that varies linearly with the in d e p e n d e n t variable t (time).

This function describes a signal of two in dependent variables x and y that could represent
the two spatial coordinates in a p lane.
System:

A system may also be defined as a physical device that performs an operation on a signal.
For ex ample, a filter used to reduce the noise and interference corrupting desired in
formation bearing signal is called a system .
signal processing:
When we pass a signal through a system , as in filtering, we say that we have processed the
signal. In this case the processing of the signal involves filtering the noise and interference
from the desired signal. If the operation on the signal is n o n linear, the system is said to be
nonlinear, and so forth . Such operations are usually referred to as signal processing.
Analog signal processing:

Digital signal processing:

Advantages of Digital over Analog Signal Processing :

 A digital programmable system allows s flexibility in re configuring the digital

signal processing operations simply by changing the program.
 A digital system provides much better control of accuracy.

 Digital signals are easily stored on magnetic media (tape or disk) without
deterioration or loss of signal fidelity beyond that introduced in the A /D
conversion.
 Digital implementation of the signal processing system is cheaper than analog
signal processing.

Limitations:

 One practical limitation is the speed of operation of A /D converters and digital

signal processors. We shall see that signals having extremely wide band widths
require fast-sampling
 -rate A /D converters and fast digital signal processors. Hence there are analog
signals with large bandwidths for which a digital processing approach is beyond the
state of the art of digital hardware.
Video Content / Details of website for further learning (if any):
https://www.allaboutcircuits.com/technical-articles/an-introduction-to-digital-signal-processing/
Important Books/Journals for further learning including the page nos.:
John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg:210-212

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-2

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Introduction to DFT

Introduction : ( Maximum 5 sentences)

 The discrete Fourier transform (DFT) is a fundamental transform in digital signal processing,
with applications in frequency analysis, fast convolution, image processing, etc.
 DFT is to compute the frequency responses of filters, to implement convolution, and spectral
estimation
Prerequisite knowledge for Complete understanding and learning of Topic:
( Max. Four important topics)
1.Signals and System
2.Fourier Transform
3.Z-Transform
Detailed content of the Lecture:

The discrete Fourier transform (DFT) derived from the Fourier series The exponential Fourier series
of a continuous time periodic signal x(t) with fundamental period T0 is given by the synthesis
equation

where the Fourier coefficients Xk are given by the analysis equation

with the fundamental frequency F0 and the period T0 related by F0 (Hz) = 1/T0 (sec).
To obtain finite-sum approximations for the above two equations, consider the analog periodic signal
x(t) shown in Figure and its sampled version . Using , we can approximate the integral

for Xk by the sum where we used the relation

F0T = 1/N, and approximated by T, and have used the shorthand notation x(n) = xs(nT).
(This procedure is similar to that used in a typical introduction to integral calculus). Discrete time
domain signal is converting into discrete frequency domain
The DFT of X(k)={1, 1, 1, 1}
The IDFT of X(n)={2, 0, 0, 1}

Video Content / Details of website for further learning (if any):

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Important Books/Journals for further learning including the page nos.:
John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg:399 -401

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-3

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Properties of Discrete Fourier Transform

Introduction :
 The properties of the DFT (for finite duration sequences) are essentially similar to those of
the DFS for periodic sequences and result from the implied periodicity in the DFT
representation.

Prerequisite knowledge for Complete understanding and learning of Topic:

( Max. Four important topics)
1.Signals and System
2.Fourier Transform
3.Z-Transform
Properties of Discrete Fourier Transform

 Periodicity:
Due to the N-sample periodicity of the complex exponential basis functions ei2πnkN in
the DFT and IDFT, the resulting transforms are also periodic with N samples.

X(k+N)=X(k)x(n)=x(n+N)

 Linearity : Ax(n)  Bx (n)  AX (k )  BX (k )

 Time Shift: x(n  m)  X (k )e

 j 2km / N
 X (k )WN k  m
 Frequency Shift:

x(n)e j 2km / N  X (k  m)
 Time Reversal : x ( n)  X (  k )
N 1

 x(m) y(n  m)  x(n)y(n)  X (k )Y (k )

 Circular convolution m 0

 Circular Convolution Property

Circular convolution is defined as x(n)*h(n)=N−1∑m=0x(m)x((n−m)modN)

 Circular convolution of two discrete-time signals corresponds to multiplication of their

DFTs: x(n)*h(n)=X(k)H(k)

 Multiplication Property

A similar property relates multiplication in time to circular convolution in

frequency.x(n)h(n)=1NX(k)*H(k)

 Multiplication: x(n) y (n)  X (k )Y (k )

 Parseval’s Theorem
N 1 N 1

 | x ( n) |
n 0
2
N 1
 | X (k ) |
k 0
2

Video Content / Details of website for further learning (if any):

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Important Books/Journals for further learning including the page nos.:
John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 409-414

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-4

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture

:

Topic of Lecture: Circular Convolution

Introduction :
 Find the response of filter with zero padding.The output samples y (n) =Max(L,M)
 Properties of Convolution: 1- Commutative law, 2- Associative law , 3-Distributive law

Prerequisite knowledge for Complete understanding and learning of Topic:

1.Signals and System
2.Fourier Transform
3.Z-Transform
The Convolution Sum :
An arbitrary input signal x( n) in to a weighted sum of impulses, We are now ready to
determine the response of any relaxed linear system to any Input signal. First, we denote the
response y(n,k) of the system to the input unit Sample sequence at n = k by the special
symbol h(n, k), -∞<k

< ∞. T h a t is,
if the input is the arbitrary signal x(n) that is expressed as a sum of weighted impulses, that is.

then the response of the system to x(n) is the corresponding sum of weighted outputs, that is,

Clearly, the above equation follows from the superposition property of linear systems, and
is know n as the superposition summation.th en by the time-invariance property , the
response of the system to the delayed unit sample sequence δ(n - k) is

Consequently , the superposition summation formula in reduces to

 The above formula gives the response y(n) of the LTI system as a function of the input
signal
x ( n ) and the unit sample (impulse) response h(n) is called a convolution sum.

To summarize, the process of computing the convolution between x ( k ) and h(k)

involves the following four steps.
1. Folding. Fold h(k) about k = 0 to obtain h ( - k ) .
2. Shifting, Shift h ( —k) by n0 to the right (left) if n0 is positive (negative), to obtain h(n0—
k).
3. Multiplication. Multiply x ( k ) by h(n0— k) to obtain the product sequencevn0(k) =
x ( k ) h(n0— k).
4. Summation. Sum all the values o f the product sequence vn0(k) to obtain the value
of the output at time n = n0.

This operation is analogous to linear convolution, but with a subtle difference

Consider two length-N sequences, g[n] and h[n], respectively
 Their linear convolution results in a length-(2N-1) sequence yL[n] given by In computing
yL[n] we have assumed that both length-N sequences have been zero-padded to extend
their lengths to 2N-1
 The longer form of yL[n] results from the time-reversal of the sequence h[n] and its linear
shift to the right
The first nonzero value of yL[n] is yL[n]=g[0]h[0], and the last nonzero value is
yL[2N-2]=g[N-1]h[N-1]
 To develop a convolution-like operation resulting in a length-N sequence yC[n], we need
to define a circular time-reversal, and then apply a circular time-shift, resulting operation,
called a circular convolution, is defined by

Since the operation defined involves two length-N sequences, it is often referred to as an N-point
circular convolution, denoted as
y[n] = g[n] h[n]
The circular convolution is commutative, i.e.
g[n] h[n] = h[n] g[n]

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Important Books/Journals for further learning including the page nos.:
John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 415-420

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-5

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Filtering methods based on DFT

Introduction :

In signal processing, the function of a filter is to remove unwanted parts of the signal, such as random
noise, or to extract useful parts of the signal, such as the components lying within a certain frequency
range .

Prerequisite knowledge for Complete understanding and learning of Topic:

Filters may be classified as either digital or analog.

 Digital filters are implemented using a digital computer.
 Analog filters may be classified as either passive or active and are usually implemented with R,
L, and C components and operational amplifiers.
Digital filter Specifications:

 These filters are unrealizable because (one of the following is sufficient)their impulse
responses infinitely long non-causal.
 Their amplitude responses cannot be equal to a constant over a band of frequencies.
 The realizable squared amplitude response transfer function (and its differential) is continuous
in Such functions
o If IIR can be infinite at point but around that point cannot be zero.
o If FIR cannot be infinite anywhere.

Detailed content of the Lecture:

DFT FOR LINEAR FILTERING:

Consider that input sequence x(n) of Length L & impulse response of same system is h(n)
having M samples. Thus y(n) output of the system contains N samples where N=L+M-1. If DFT of
y(n) also contains N samples then only it uniquely represents y(n) in time domain. Multiplication of
two DFT s is equivalent to circular convolution of corresponding time domain sequences. But the
length of x(n) & h(n) is less than N. Hence these sequences are appended with zeros to make their
length N called as “Zero padding”.

METHOD 1: OVERLAP SAVE METHOD OF LINEAR FILTERING:

Step 1:
In this method L samples of the current segment and M-1 samples of the previous segment forms the
input data block. Thus data block will be

X1(n) ={0,0,0,0,0,………………… ,x(0),x(1),…………….x(L-1)}

X2(n) ={x(L-M+1), …………….x(L-1),x(L),x(L+1),,,,,,,,,,,,,x(2L-1)} X3(n) ={x(2L-M+1),
…………….x(2L-1),x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1)}
Step2:

Unit sample response h(n) contains M samples hence its length is made N by padding zeros. Thus h(n)
also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,……………………(L-1 zeros)}

Step3:The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT of
output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4: The sequence ym(n) can be obtained by taking N point IDFT of Y`m(k). Initial (M-1) samples
in the corresponding data block must be discarded. The last L samples are the correct output samples.
Such blocks are fitted one after another to get the final output.
METHOD 2: OVERLAP ADD METHOD OF LINEAR FILTERING:
Step 1: In this method L samples of the current segment and M-1 samples of the previous segment
forms the input data block. Thus data block will be

X1(n) ={x(0),x(1),…………….x(L-1),0,0,0,……….}

X2(n) ={x(L),x(L+1),x(2L-1),0,0,0,0}

X3(n) ={x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1),0,0,0,0}

Step2: Unit sample response h(n) contains M samples hence its length is made N by padding zeros.
Thus h(n) also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,……………………(L-1 zeros)}

Step3: The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT
of output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4: The sequence ym(n) can be obtained by taking N point IDFT of Y`m(k). Initial (M-1) samples
are not discarded as there will be no aliasing. The last (M-1) samples of current output block must be
added to the first M-1 samples of next output block. Such blocks are fitted one after another to get the
final output.
The comparison between overlap save method and overlap add method are tabulated as
follows,

S.No. OVERLAP SAVE METHOD OVERLAP ADD METHOD

1. L samples of the current segment and (M- L samples from the input sequence and
1) samples of the previous segment forms padding (M-1) zeros forms data block of
the input data block. size N
2. Initial M-1 samples of output sequence There will be no aliasing in the output data
are discarded which occurs due to aliasing block.
effect.
3. To avoid loss of data due to aliasing last Last M-1 samples of current output block
M-1 samples of each data record are must be added to the first M-1 samples of
saved. next output block. Hence called as overlap
add method.

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https://www.youtube.com/watch?v=aEvBea7Mxaw

Important Books/Journals for further learning including the page nos.:

John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 458-460

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-6

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Fast Fourier Transform(FFT)

Introduction :

 A fast Fourier transform (FFT) is an algorithm that computes the discrete Fourier
transform (DFT) of a sequence, or its inverse (IDFT).

Prerequisite knowledge for Complete understanding and learning of Topic:

 Discrete Fourier Transform (DFT) is a transform like Fourier transform used with
digitized signals.
 It is the discrete version of the FT that views both the time domain and frequency
domain as periodic.
 Fast Fourier Transform (FFT) is just an algorithm for fast and efficient computation of
the DFT.
 The Fast Fourier Transform (FFT) is an approach to reduce the computational
complexity that produces the same result as a DFT (same result, significantly fewer
multiplications).
 The Fast Fourier Transform (FFT) is an implementation of the DFT which produces
almost the same results as the DFT, but it is incredibly more efficient and much faster
Detailed content of the Lecture:

 FFT is an implementation of the DFT used for used for fast computation of the DFT.
 FFT is an efficient way of computing the DFT.
 The FFT is a fast algorithm for computing the DFT.
 To compute the DFT of an N-point sequence using equation (1) would take O(N2 )
multiplies and adds.
 The FFT algorithm computes the DFT using O(N log N) multiplies and adds.
 The inverse FFT (IFFT) is identical to the FFT, except one exchanges the roles of a and
A, the signs of all the exponents of W are negated, and there’s a division by N at the
end.

FFT vs DFT Comparison Table

S.No. FFT DFT
1 FFT stands for Fast Fourier DFT stands for Discrete Fourier
Transform Transform
2 It is much faster version of DFT It is the discrete version of Fourier
algorithm. transform.
3 Various fast DFT computation It is the algorithm that transforms the
techniques are collectively known as time domain signals to the frequency
FFT algorithm. domain components.
4 It is the implementation of DFT It establishes the relation between the
time domain and frequency domain
representation
5 Applications include integer and Applications of DFT includes solving
polynomial multiplication, filtering partial differential applications,
algorithm, calculating Fourier series computing polynomial multiplication,
coefficients etc. spectral analysis etc.

Algorithm of FFT and DFT:

 The most commonly used FFT algorithm is the Cooley-Tukey algorithm.

 It’s a divide and conquer algorithm for the machine calculation of complex Fourier
series.

 It breaks the DFT into smaller DFTs.

  Other FFT algorithms include the Rader’s algorithm, Wino grad Fourier transform
algorithm, Chirp Z-transform algorithm, etc.

 The DFT algorithms can be either programmed on general purpose digital computers
or implemented directly by special hardware.

 The FFT algorithm is used to compute the DFT of a sequence or its inverse.

 A DFT can be performed as O (N2) in time complexity, whereas FFT reduces the time
complexity in the order of O (NlogN).

Applications of FFT and DFT:

 FFT can be used in many digital processing systems across a variety of applications
such as calculating a signal’s frequency spectrum, solving partial differential
applications, spectral analysis etc.

 FFT has been widely used for acoustic measurements in churches and concert halls.

 FFT include spectral analysis in analog video measurements, large integer and
polynomial multiplication, filtering algorithms, computing isotopic distributions,
calculating Fourier series coefficients, calculating convolutions, generating low
frequency noise etc.

Video Content / Details of website for further learning (if any):

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Important Books/Journals for further learning including the page nos.:

John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 415-417

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-7

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: FFT Algorithms

Introduction :

 The Fast Fourier Transform (FFT) is a very efficient algorithm for performing a discrete Fourier
transform.

Prerequisite knowledge for Complete understanding and learning of Topic:

FAST FOURIER ALGORITHM (FFT):

 Large number of the applications such as filtering, correlation analysis, and spectrum analysis
require calculation of DFT.
 Direct computation of DFT requires large number of computations.
 Hence special algorithms are developed to compute DFT quickly called as Fast Fourier
algorithms (FFT).
 The radix-2 FFT algorithms are based on divide and conquer approach.
 In this method, the N-point DFT is successively decomposed into smaller DFT s. Because of this
decomposition, the numbers of computations are reduced.

Detailed content of the Lecture:

There are two types of FFT algorithms. They are,

 Decimation in Time
 Decimation in Frequency
RADIX-2 FFT ALGORITHMS:

DECIMATION IN TIME (DITFFT)

 There are three properties of twiddle factor WN

N point sequence x(n) be splitted into two N/2 point data sequences f1(n) and f2(n).

f1(n) contains even numbered samples of x(n) and f2(n) contains odd numbered samples of x(n).

This splitted operation is called decimation.

Since it is done on time domain sequence it is called “Decimation in Time”.

Thus
f1(m)=x(2m) where n=0,1,………….N/2-1
f2(m)=x(2m+1) where n=0,1,………….N/2-1
N point DFT is given as

Since the sequence x(n) is splitted into even numbered and odd numbered samples, thus

Fig 1 shows that 8-point DFT can be computed directly and hence no reduction in computation.
Video Content / Details of website for further learning (if any):

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Important Books/Journals for further learning including the page nos.:

John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 420-424

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-8

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Decimation in time Algorithms

Introduction :

 Decimation in time FFT algorithms are based upon decomposition of the input sequence into
smaller and smaller sub sequences.
 DIT FFT input sequence is in bit reversed order while the output sequence is in natural order.
Prerequisite knowledge for Complete understanding and learning of Topic:

 Decimation is the process of reducing the sampling rate.

 Decimation is a term that historically means the removal of every tenth one.
 But in signal processing, decimation by a factor of 10 actually means keeping only every
tenth sample.
 This factor multiplies the sampling interval or, equivalently, divides the sampling rate.

Detailed content of the Lecture:

Decimation in Time:

The DFT is defined by,

where is the input signal amplitude at time , and

Note that

When is even, the DFT summation can be split into sums over the odd and even indexes of the
input signal:

where and denote the even- and odd-indexed samples from .

Thus, the length DFT is computable using two length DFTs. The complex

factors are called twiddle factors.

The splitting into sums over even and odd time indexes is called decimation in time.
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Important Books/Journals for further learning including the page nos.:

John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 429-431

Course Faculty

Verified by HOD
 MUTHAYAMMAL ENGINEERING COLLEGE
(An Autonomous Institution)
(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-9

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu.Prof./ECE

Unit : FOURIER ANALYSIS OF DISCRETE TIME SIGNALS Date of Lecture :

Topic of Lecture: Decimation in frequency algorithms

Introduction :

 The Decimation in Frequency FFT algorithms are based upon decomposition of the output
sequence into smaller and smaller sub sequences.
 In Decimation in Frequency FFT, input sequence is in natural order.
 DFT should be read in bit reversed order.

Prerequisite knowledge for Complete understanding and learning of Topic:

 The decimation-in-frequency FFT is a flow-graph reversal of the decimation-in-time FFT: it has

the same twiddle factors (in reverse pattern) and the same operation counts.

Detailed content of the Lecture:

 In a decimation-in-frequency radix-2 FFT as illustrated in Figure, the output is in bit-reversed

order.
In DIF N Point DFT is splitted into N/2 points DFT s. X(k) is splitted with k even and k odd this is
called Decimation in frequency(DIF FFT).

N point DFT is given as,

Since the sequence x(n) is splitted N/2 point samples, thus

Let us split X(k) into even and odd numbered samples

Fig 2 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=4
DIFFERENCE BETWEEN DECIMATION IN TIME FFT AND DECIMATION IN
FREQUENCY FFT:

S.No DIT FFT DIF FFT

.

1 DIT FFT algorithms are based upon DIF FFT algorithms are based upon
decomposition of the input sequence decomposition of the output sequence into
into smaller and smaller sub sequence smaller and smaller sub sequence.

2 In this input sequence x(n) is splitted In this input sequence X(k) is considered to be
into even and odd numbered samples splitted into even and odd numbered samples

3 Splitting operation is done on time Splitting operation is done on frequency domain

domain sequence. sequence.

4 In DIT FFT input sequence is in bit In DIF FFT, input sequence is in natural order
reversed order while the output and DFT should be read in bit reversed order.
sequence is in natural order

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Important Books/Journals for further learning including the page nos.:

John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 434-441

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Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-10

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU.Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Structure of IIR

Introduction :
 IIR system has infinite duration unit sample response h(n)=0 for n<0
 Thus the unit sample response exists for the duration from 0 to ∞
 IIR filter is usually more efficient design in terms of computation time and memory
requirement.
 IIR systems usually requires less processing time and storage as compared with FIR
Prerequisite knowledge for Complete understanding and learning of Topic:
( Max. Four important topics)
1.Signals and System
2.Fourier Transform
3.Z-Transform
Introduction

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 themselves to a parallel form realization. We begin by describing two direct-form realizations.
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Important Books/Journals for further learning including the page nos.:
John G. Proakis & Dimitris G.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg:298-305

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LECTURE HANDOUTS
L-11

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU. Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Analog Filter Design

Introduction : ( Maximum 5 sentences)

 Analog filters are designed with various components like resister, inductor and
capacitance
 Analog filters less accurate & because of component tolerance of active components &
more sensitive to environmental chandes
 An analog filter can only be changed by redesigning the filter circuit
Prerequisite knowledge for Complete understanding and learning of Topic:
( Max. Four important topics)
1.Signals and System
2.Filter
3.Analog Signal Processing
Detailed content of the Lecture:

Digital filters can be designed using analog design methods by following these steps:
1. Filter specifications are specified in the digital domain. The filter type (highpass,
lowpass,bandpass etc.) is specified.
2. An equivalent lowpass filter is designed that meets these specifications.
3. The analog lowpass filter is transformed using spectral transformations into the correct type of
filter.
4. The analog filter is transformed into a digital filter using a particular mapping.
5. There are many different types of spectral transformations and there are many mappings from
analog to digital filters. the most famous mapping is known as the bilinear transform, and we
will discuss that in a different chapter.
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg:262 -268

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LECTURE HANDOUTS
L-12

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU. Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Discrete time IIR filter from analog filter

Introduction :
 To design a discrete-time lowpass filter using both the impulse invariance method and the
bilinear transform method.

Prerequisite knowledge for Complete understanding and learning of Topic:

( Max. Four important topics)
1.Infinite Impulse Response

Properties of Discrete Fourier Transform

Video Content / Details of website for further learning (if any):
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 298-305

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Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-13

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU.Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: IIR filter design by Impulse Invariance

Introduction :
 Impulse invariance is a technique for designing discrete-time infinite-impulse-
response (IIR) to produce the impulse response of the discrete-time system.
 The frequency response of the discrete-time system will be a sum of shifted copies of the
frequency response of the continuous-time system

Prerequisite knowledge for Complete understanding and learning of Topic:

1. Infinite Impulse Response
2.Filter
Detailed content of the Lecture:
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 415-420

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(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-14

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU.Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: IIR filter design by Bilinear Transformation

Introduction :
 Impulse invariance is a technique for designing discrete-time infinite-impulse-
response (IIR) to produce the impulse response of the discrete-time system.
 The frequency response of the discrete-time system will be a sum of shifted copies of the
frequency response of the continuous-time system

Prerequisite knowledge for Complete understanding and learning of Topic:

1. Infinite Impulse Response
2.Filter
Detailed content of the Lecture:
The bilinear transform (also known as Tustin's method) is used in digital signal processing and
discrete-time control theory to transform continuous-time system representations to discrete-time
and vice versa.The bilinear transform is a special case of a conformal mapping (namely, a Möbius
transformation), often used to convert a transfer function of a linear, time-invariant (LTI) filter in
the continuous-time domain (often called an analog filter) to a transfer function of a linear, shift-
invariant filter in the discrete-time domain (often called a digital filter although there are analog
filters constructed with switched capacitors that are discrete-time filters). It maps positions on
the in the s-plane to the unit circle in the z-plane. Other bilinear transforms can be used to warp
the frequency response of any discrete-time linear system (for example to approximate the non-
linear frequency resolution of the human auditory system) and are implementable in the discrete
domain by replacing a system's unit delays {\displaystyle \left(z^{-1}\right)} with first order all-
pass filters.

The transform preserves stability and maps every point of the frequency response of the
continuous-time filter, to a corresponding point in the frequency response of the discrete-time
filter, although to a somewhat different frequency, as shown in the Frequency warping section
below. This means that for every feature that one sees in the frequency response of the analog
filter, there is a corresponding feature, with identical gain and phase shift, in the frequency
response of the digital filter but, perhaps, at a somewhat different frequency. This is barely
noticeable at low frequencies but is quite evident at frequencies close to the Nyquist frequency.
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 415-420

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(Approved by AICTE, New Delhi, Accredited by NAAC & Affiliated to Anna University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-15

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU.Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Bilinear transformation

Introduction :
The Bilinear transform is a mathematical relationship which can be used to convert the transfer
function of a particular filter in the complex Laplace domain into the z-domain, and vice-versa. The
resulting filter will have the same characteristics of the original filter, but can be implemented using
different techniques. The Laplace Domain is better suited for designing analog filter components,
while the Z-Transform is better suited for designing digital filter components.
Prerequisite knowledge for Complete understanding and learning of Topic:

1.Infinite Impulse Response

Detailed content of the Lecture:

Bilinear transformation technique:

This is the most common method for transforming the system function H a (s) of an analogue filter to
the system function H(z) of an IIR discrete time filter. It is not the only possible transformation, but a
very useful and reliable one.

Introduction

Definition: Given analogue transfer function H a (s), replace s by :

2  z  1
T  z  1

to obtain H(z). For convenience we can take T=1.

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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 285-290

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Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

LECTURE HANDOUTS
L-16

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANES BABU.Prof./ECE

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Problems in IIR filter

Introduction :
The impulse invariance method of IIR filter design is based upon the notion that we can design a
discrete filter whose time-domain impulse response is a sampled version of the impulse response of a
continuous analog filter
Prerequisite knowledge for Complete understanding and learning of Topic:

1.Infinite Impulse Response

Detailed content of the Lecture:

Video Content / Details of website for further learning (if any):
https://www.alljntuworld.in/download/digital-signal-processing-dsp-materials-notes/#
Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 291-294

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LECTURE HANDOUTS
L-17

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Approximation of Derivatives

Introduction :
IIR Filter Design by Approximation of Derivatives Analogue filters having rational transfer function
H(s) can be described by the linear constant coefficient differential equation. One of the simplest
methods for converting an analog filter into a digital filter is to approximate the differential equation
by an equivalent difference equation. This approach is often used to solve a linear constant coefficient
differential equation numerically on a digital computer
Prerequisite knowledge for Complete understanding and learning of Topic:

1.Infinite Impulse Response

Detailed content of the Lecture:

Video Content / Details of website for further learning (if any):
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 301-309

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LECTURE HANDOUTS
L-18

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.GANESH BABU

Unit : DESIGN OF IIR FILTER Date of Lecture :

Topic of Lecture: Filter design using frequency translation.

Introduction :
Frequency translation is the process of moving a signal from one part of the frequency axis, to another
part of the axis. Frequency translation is often done in wireless communications systems to move a
pass band signal to base band before demodulation
Prerequisite knowledge for Complete understanding and learning of Topic:

1.Infinite Impulse Response

Detailed content of the Lecture:

Filter design using frequency translation:

Video Content / Details of website for further learning (if any):
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Important Books/Journals for further learning including the page nos.:
John G. Proakis&DimitrisG.Manolakis” Digital Signal Processing Principles Algorithms &
Applications” Pearson Education / Prentice Hall Fourth Edition, 2007 Pg: 310-312

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University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

L-19
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Structure of Finite Impulse Response filter

Introduction :

 The term digital filter arises because these filters operate on discrete-time signals
 The term finite impulse response arises because the filter output is computed as a weighted, finite
term sum, of past, present, and perhaps future values of the filter input, i.e.,

where both M1 and M2 are finite

 An FIR filter is based on a feed-forward difference equation
 Feed-forward means that there is no feedback of past or future outputs to form the present output, just
input related terms
Prerequisite knowledge for Complete understanding and learning of Topic:

 Filters
 Types of Filters
Detailed content of the Lecture:

 For a FIR design, we have a couple of options such as the windowing method, the frequency sampling
method, and more.
 Then, we need to choose a realization structure for the obtained system function. In other words, there
are several structures which exhibit the same system function H(z)
  One consideration for choosing the appropriate structure is the sensitivity to coefficient quantization.
 Since a digital filter uses a finite number of bits to represent signals and coefficients, we need
structures which can somehow retain the target filter specifications even after quantizing the
coefficients.
 In addition, sometimes we observe that a particular structure can dramatically reduce the
computational complexity of the system.

 The basic block diagram for an FIR filter of length N is shown below,

 The delays result in operating on prior input samples.

 The hk values are the coefficients used for multiplication, so that the output at time n is the summation of
all the delayed samples multiplied by the appropriate coefficients.
 The process of selecting the filter's length and coefficients is called filter design.
 The goal is to set those parameters such that certain desired stopband and passband parameters will result
from running the filter.

 The results of the design effort should be the same:

1.A frequency response plot verifies that the filter meets the desired specifications, including ripple
and transition bandwidth.

2.The filter's length and coefficients.

Advantages of FIR filter:

 They can easily be designed to be “linear phase”

 They are simple to implement
 They are suited to multi-rate applications.
 They have desirable numeric properties.

Disadvantages of FIR filter:

Compared to IIR filters, FIR filters sometimes have the disadvantage that they require more memory and/or
calculation to achieve a given filter response characteristic. Also, certain responses are not practical to
implement with FIR filters.

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https://www.sciencedirect.com/topics/engineering/finite-impulse-response
https://www.allaboutcircuits.com/technical-articles/structures-for-implementing-finite-impulse-response-
filters/
https://www.vyssotski.ch/BasicsOfInstrumentation/SpikeSorting/Design_of_FIR_Filters.pdf

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University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

L-20
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof/ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Linear phase Finite Impulse Response filter

Introduction :

 Linear phase is a property of a filter, where the phase response of the filter is a linear function of
frequency.
 The result is that all frequency components of the input signal are shifted in time (usually delayed) by
the same constant amount (the slope of the linear function), which is referred to as the group delay.
 There is no phase distortion due to the time delay of frequencies relative to one another.
 For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter by having coefficients which are symmetric or anti-symmetric
Prerequisite knowledge for Complete understanding and learning of Topic:

 A filter is called a linear phase filter if the phase component of the frequency response is a linear
function of frequency.
 For a continuous-time application, the frequency response of the filter is the Fourier transform of the
filter's impulse response, and a linear phase version has the form:
Detailed content of the Lecture:

Linear-phase FIR filter can be divided into four basic types.

Type Impulse response

I symmetric length is odd
II symmetric length is even
III anti-symmetric length is odd
IV anti-symmetric length is even

When h(n) is nonzero for 0 ≤ n ≤ N −1 (the length of the impulse response h(n) is N), then the symmetry of
the impulse response can be written as h(n) = h(N − 1 − n) and the anti-symmetry can be written as h(n) =
−h(N − 1 − n).

TYPE I: ODD-LENGTH SYMMETRIC

The frequency response of a length N = 5 FIR Type I filter can be written as follows
TYPE II: EVEN-LENGTH SYMMETRIC

The frequency response of a length N = 4 FIR Type II filter can be written as follows.
TYPE III: ODD-LENGTH ANTI-SYMMETRIC
TYPE IV: EVEN-LENGTH ANTI-SYMMETRIC

The frequency response of a length N = 4 FIR Type IV filter can be written as follows.

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University)
Rasipuram - 637 408, Namakkal Dist., Tamil Nadu

L-21
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Finite Impulse Response filter

Introduction :

 Linear phase is a property of a filter, where the phase response of the filter is a linear function of
frequency.
 The result is that all frequency components of the input signal are shifted in time (usually delayed) by
the same constant amount (the slope of the linear function), which is referred to as the group delay.
 There is no phase distortion due to the time delay of frequencies relative to one another.
 For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter by having coefficients which are symmetric or anti-symmetric
Prerequisite knowledge for Complete understanding and learning of Topic:

 A filter is called a linear phase filter if the phase component of the frequency response is a linear
function of frequency.
 For a continuous-time application, the frequency response of the filter is the Fourier transform of the
filter's impulse response, and a linear phase version has the form:
Detailed content of the Lecture:
TYPE I: ODD-LENGTH SYMMETRIC

The frequency response of a length N = 5 FIR Type I filter can be written as follows

TYPE II: EVEN-LENGTH SYMMETRIC

The frequency response of a length N = 4 FIR Type II filter can be written as follows.

TYPE III: ODD-LENGTH ANTI-SYMMETRIC

TYPE IV: EVEN-LENGTH ANTI-SYMMETRIC

The frequency response of a length N = 4 FIR Type IV filter can be written as follows.
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L-22
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Fourier Series

Introduction :

 Fourier series is, in some way a combination of the Fourier sine and Fourier cosine series.
 The Fourier sine/cosine series are not the series will actually converge to f(x) or not at this point.
 The process of deriving the weights that describe a given function is a form of Fourier analysis. For
functions on unbounded intervals, the analysis and synthesis analogies are Fourier transform and
inverse transform.
Prerequisite knowledge for Complete understanding and learning of Topic:

A Fourier series is an expansion of a periodic function f(x) in terms of an infinite sum of sines and cosines.
Fourier series make use of the orthogonality relationships of the sine and cosine functions. The computation
and study of Fourier series is known as harmonic analysis and is extremely useful as a way to break up an
arbitrary periodic function into a set of simple terms that can be plugged in, solved individually, and then
recombined to obtain the solution to the original problem or an approximation to it to whatever accuracy is
desired or practical.The mathematical expression

is called fourier series.

Detailed content of the Lecture:

 One of the successful methods, for the design of FIR filters is based on the application of the Fourier
series.
 In this method, it is observed that the frequency of an FIR filter is a periodic function of frequency
with a period equal to the sampling frequency and consequently, it can be expressed in terms of the
Fourier series.
 The Fourier series by itself does not lead to satisfactory results but by using the Fourier series in
conjunction with a special class of functions known as window functions, good results can be
obtained.
 Approximation obtained by this method is suboptimal but the amounts of design effort and
computation required are relatively insignificant.
The frequency response of an FIR digital filter can be represented by the Fourier series as,

where the Fourier coefficients h(n) are the desired impulse response sequence of the filter, which can be
determined from

If we substitute e)“ = z, we obtain the transfer function ofthe digital filter, that is,

Therefore, if an expression is available for the frequency response, a transfer function can be obtained, which
happens to be a non causal and of infinite order. A finite order transfer 35 function can be obtained by
truncating the Fourier series. This can be accomplished by letting,
  This modification does not change the amplitude response of the filter, however the abrupt truncation
of the Fourier series results in oscillations in the passband and stopband.
 These oscillations are due to slow convergence of the Fourier series, particularly near points of
discontinuity. This effect is known as the Gibbs phenomenon

The amplitude of Gibbs oscillations can be reduced using discrete window functions.

A window function, represented by w(n) has the following time-domain properties:

1. w(n) = 0 for | n j> (N-1)/2

2. It is symmetrical, i.e., w(-n) = w(n).

The application of a window function consists of multiplying the impulse response obtained by applying the
Fourier series by the window function to obtain a modified impulse response as

hw(n) = w(n) h(n)

Since the window function is of finite duration, a finite-order transfer function given by
Through the application of the complex convolution, the frequency response of the modified filter can be
expressed as,

The application of the window function has two effects on the amplitude response of the filter. First, the
amplitudes of Gibbs’ oscillations in the pass bands and stop bands are directly related to the ripple ratio of the
window. Second, transition bands are introduced between pass bands and stop bands whose width is directly
related to the main-lobe width of the window.

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L-23
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 The window method for digital filter design is fast, convenient, and robust, but generally suboptimal.
 It is easily understood in terms of the convolution theorem for Fourier transforms, making it
instructive to study after the Fourier theorems and windows for spectrum analysis.
 It can be effectively combined with the frequency sampling method.
 The window method consists of simply ``windowing'' a theoretically ideal filter impulse
 response by some suitably chosen window function ,

 Window functions are always time limited. This means there is always a finite integer such

that for all .

The final windowed impulse response is thus always time-limited

Detailed content of the Lecture:

The unit sample response hd(n) obtained from the above relation is infinite in duration, so it must be
truncated at some point say n= M-1 to yield an FIR filter of length M (i.e. 0 to M-1). This truncation of hd(n)
to length M-1 is same as multiplying hd(n) by the rectangular window defined as

Thus the unit sample response of the FIR filter becomes

The multiplication of the window function w(n) with hd(n) is equivalent to convolution of Hd(w) with W(w),
where W(w) is the frequency domain representation of the window function

Thus the convolution of Hd(w) with W(w) yields the frequency response of the truncated FIR filter
But direct truncation of hd(n) to M terms to obtain h(n) leads to the Gibbs phenomenon effect which
manifests itself as a fixed percentage overshoot and ripple before and after an approximated discontinuity in
the frequency response due to the non-uniform convergence of the fourier series at a discontinuity. Thus the
frequency response obtained by using (8) contains 3 ripples in the frequency domain. In order to reduce the
ripples, instead of multiplying hd(n) with a rectangular window w(n), hd(n) is multiplied with a window
function.

The several effects of windowing the Fourier coefficients of the filter on the result of the frequency response
of the filter are as follows:

(i) A major effect is that discontinuities in H(w) become transition bands between values on either side of the
discontinuity.

(ii) The width of the transition bands depends on the width of the main lobe of the frequency response of the
window function, w(n) i.e. W(w).

(iii) Since the filter frequency response is obtained via a convolution relation, it is clear that the resulting
filters are never optimal in any sense.

(iv) As M (the length of the window function) increases, the main lobe width of W(w) is reduced which
reduces the width of the transition band, but this also introduces more ripple in the frequency response.

(v) The window function eliminates the ringing effects at the band edge and does result in lower side lobes at
the expense of an increase in the width of the transition band of the filter.
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L-24
LECTURE HANDOUTS

ECE III/V
Course Name with Code : 16ECD09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : FIR FILTER DESIGN Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Rectangular window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample response
hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function to
be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
 These two requirements are interrelated .
  Window Technique implicates a function called window Function.
 It is also known as Tapering Function.

Detailed content of the Lecture:

Rectangular Window

The rectangular window (sometimes known as the Boxer or Dirichlet window) is the simplest window,
equivalent to replacing all but N values of a data sequence by zeros, making it appear as though the waveform
suddenly turns on and off.

where is the window length in samples (assumed odd for now). A plot of the rectangular window appears
in Fig.3.1 for length . It is sometimes convenient to define windows so that their dc gain is 1, in

which case we would multiply the definition above by

also called the Dirichlet function or periodic sinc function. This (real) result is for the zero-centered rectangular
window. For the causal case, a linear phase term appears:

The term ``aliased sinc function'' refers to the fact that it may be simply obtained by sampling the length-

continuous-time rectangular window, which has Fourier transform sinc (given

amplitude in the time domain). Sampling at intervals of seconds in the time domain corresponds to

aliasing in the frequency domain over the interval Hz, and by direct derivation. The window duration
increases continuously in the time domain: the magnitude spectrum can only change in discrete jumps as new
samples are included, even though it is continuously parametrized in .As the sampling rate goes to infinity,
the aliased sinc function therefore approaches the sinc function.

Figure: Fourier transform of the rectangular window.

The phase of rectangular-window transform is zero for , which is the width of
the main lobe. This is why zero-centered windows are often called zero-phase windows; while the phase
actually alternates between 0 and radians, the values occur only within side-lobes which are routinely
neglected.

Figure: Magnitude (dB) of the rectangular-window transform

Figure: Magnitude of the rectangular-window Fourier transform.

 Figure: Phase of the rectangular-window Fourier transform.

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L-25
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Hamming window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function
to be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
 These two requirements are interrelated .
 Window Technique implicates a function called window Function.
  It is also known as Tapering Function.

Detailed content of the Lecture:

Hamming Window

The Hamming window is determined by choosing (with ) to cancel the largest side
lobe. Doing this results in the values

The peak side-lobe level is approximately dB for the Hamming window . It happens that this choice

is very close to that which minimizes peak side-lobe level (down to dB--the lowest possible within
the generalized Hamming family).

Since rounding the optimal to two significant digits gives , the Hamming window can be
considered the ``Chebyshev Generalized Hamming Window'' (approximately). Chebyshev-type designs
normally exhibit equiripple error behavior, because the worst-case error (side-lobe level in this case) is
minimized. Generalized Hamming windows can have a step discontinuity at their endpoints, but no impulsive
points.

 The Hamming window and its DTFT magnitude are shown in Figure. The Hamming window is also
one period of a raised cosine.
 However, the cosine is raised so high that its negative peaks are above zero, and the window has
a discontinuity in amplitude at its endpoints (stepping discontinuously from 0.08 to 0).
 This is 10 dB better than the Hann case of Figure and 28 dB better than the rectangular window.
 The main lobe is approximately wide, as is the case for all members of the generalized Hamming
family
For the Hamming window, the side-lobes nearest the main lobe have been strongly shaped by the

optimization. As a result, the nearly dB per octave roll-off occurs only over an interior interval of
the spectrum, well between the main lobe and half the sampling rate. The optimized side-lobes nearest the
main lobe occupy a smaller frequency interval about the main lobe.

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L-26
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Hanning window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function
to be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
  These two requirements are interrelated .
 Window Technique implicates a function called window Function.
 It is also known as Tapering Function.
Detailed content of the Lecture:

Hanning Window

 Hanning window is the shape of one cycle of a cosine wave with 1 added to it so it is always positive.
 The sampled signal values are multiplied by the Hanning function.
 The ends of the time record are forced to zero regardless of what the input signal is doing.
 While the Hanning window does a good job of forcing the ends to zero, it also adds distortion to the
wave form being analyzed in the form of amplitude modulation; i.e., the variation in amplitude of the
signal over the time record.
 Amplitude Modulation in a wave form results in sidebands in its spectrum, and in the case of the
Hanning window, these sidebands, or side lobes as they are called, effectively reduce the frequency
resolution of the analyzer by 50%.
 It is as if the analyzer frequency "lines" are made wider.
 The highest-level side lobes are about 32 dB down from the main lobe.

 The Hanning window should always be used with continuous signals, but must never be used with
transients. The reason is that the window shape will distort the shape of the transient, and
the frequency and phase content of a transient is intimately connected with its shape.
 The measured level will also be greatly distorted. Even if the transient were in the center of
the Hanning window, the measured level would be twice as great as the actual level because of
the amplitude correction the analyzer applies when using the Hanning weighting.
 A Hanning weighted signal actually is only half there, the other half of it having been removed by the
windowing. This is not a problem with a perfectly smooth and continuous signal like a sinusoid, but
most signals we want to analyze, such as machine vibration signatures are not perfectly smooth.
 If a small change occurs in the signal near the beginning or end of the time record, it will either be
analyzed at a much lower level than its true level, or it may be missed altogether.
  For this reason, it is a good idea to employ overlap processing. To do this, two time buffers are
required in the analyzer.
 For 50% overlap, the sequence of events is as follows:
 When the first buffer is half full, i.e., it contains half the samples of a time record, the second buffer is
connected to the data stream and also begins to collect samples. As soon as the first buffer is full,
the FFT is calculated, and the buffer begins to take data again.
 When the second buffer is filled, the FFT is again calculated on its contents, and the result sent to
the spectrum-averaging buffer. This process continues on until the desired number of averages is
collected.

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L-27
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Frequency sampling method

Introduction :

 The frequency-sampling method for FIR filter design is perhaps the simplest and most direct
technique imaginable when a desired frequency response has been specified.
 It consists simply of uniformly sampling the desired frequency response, and performing an
inverse DFT to obtain the corresponding (finite) impulse response.
 The results are not optimal, however, because the response generally deviates from what is
desired between the samples.
 When the desired frequency-response is undersampled, which is typical, the
resulting impulse response will be time aliased to some extent.
Prerequisite knowledge for Complete understanding and learning of Topic:

 The main idea of the frequency sampling design method is that a desired frequency response can be
approximated by sampling it of N evenly spaced points and then obtaining an interpolated frequency
response that passes through the frequency samples.
 For filters with reasonably smooth frequency responses,the interpolation error is generally small.
 In the case of band select filters, where the desired frequency response changes radically across
bands, the frequency samples which occur in transition bands are made to be unspecified variables
whose values are chosen by an optimization algorithm which minimizes some function of the
 approximation error of the filter finally, it was shown that there were two distinct types of frequency
sampling filters, depending on where the initial frequency sample.

Detailed content of the Lecture:

The frequency sampling method allows us to design recursive and nonrecursive FIR filters for both standard
frequency selective and filters with arbitrary frequency response.

No recursive frequency sampling filters :

The problem of FIR filter design is to find a finite–length impulse response h (n) that corresponds to desired
frequency response. In this method h (n) can be determined by uniformly sampling, the desired frequency
response HD (ω) at the N points and finding its inverse DFT of the frequency samples

where H (k), k = 0, 1, 2,……., N-1, are samples of the HD (ω) . For linear phase filters, with positive
symmetrical impulse response, we can write

where  = (N-1)/2. For N odd, the upper limit in the summation is (N - 1)/2, to obtain a good approximation
to the desired frequency
Recursive frequency sampling filter :
In recursive frequency sampling method the DFT samples H (k) for an FIR sequence can be regarded as
samples of the filters z– transform, evaluated at N points equally spaced around the unit circle.

thus the z–transform of an FIR filter can easily be expressed in terms of its DFT coefficients,
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L-19
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Structure of Finite Impulse Response filter

Introduction :

 The term digital filter arises because these filters operate on discrete-time signals
 The term finite impulse response arises because the filter output is computed as a weighted, finite
term sum, of past, present, and perhaps future values of the filter input, i.e.,

where both M1 and M2 are finite

 An FIR filter is based on a feed-forward difference equation
 Feed-forward means that there is no feedback of past or future outputs to form the present output, just
input related terms
Prerequisite knowledge for Complete understanding and learning of Topic:

 Filters
 Types of Filters
Detailed content of the Lecture:

 For a FIR design, we have a couple of options such as the windowing method, the frequency sampling
method, and more.
 Then, we need to choose a realization structure for the obtained system function. In other words, there
are several structures which exhibit the same system function H(z)
  One consideration for choosing the appropriate structure is the sensitivity to coefficient quantization.
 Since a digital filter uses a finite number of bits to represent signals and coefficients, we need
structures which can somehow retain the target filter specifications even after quantizing the
coefficients.
 In addition, sometimes we observe that a particular structure can dramatically reduce the
computational complexity of the system.

 The basic block diagram for an FIR filter of length N is shown below,

 The delays result in operating on prior input samples.

 The hk values are the coefficients used for multiplication, so that the output at time n is the summation of
all the delayed samples multiplied by the appropriate coefficients.
 The process of selecting the filter's length and coefficients is called filter design.
 The goal is to set those parameters such that certain desired stopband and passband parameters will result
from running the filter.

 The results of the design effort should be the same:

1.A frequency response plot verifies that the filter meets the desired specifications, including ripple
and transition bandwidth.

2.The filter's length and coefficients.

Advantages of FIR filter:

 They can easily be designed to be “linear phase”

 They are simple to implement
 They are suited to multi-rate applications.
 They have desirable numeric properties.

Disadvantages of FIR filter:

Compared to IIR filters, FIR filters sometimes have the disadvantage that they require more memory and/or
calculation to achieve a given filter response characteristic. Also, certain responses are not practical to
implement with FIR filters.

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L-20
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof/ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Linear phase Finite Impulse Response filter

Introduction :

 Linear phase is a property of a filter, where the phase response of the filter is a linear function of
frequency.
 The result is that all frequency components of the input signal are shifted in time (usually delayed) by
the same constant amount (the slope of the linear function), which is referred to as the group delay.
 There is no phase distortion due to the time delay of frequencies relative to one another.
 For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter by having coefficients which are symmetric or anti-symmetric
Prerequisite knowledge for Complete understanding and learning of Topic:

 A filter is called a linear phase filter if the phase component of the frequency response is a linear
function of frequency.
 For a continuous-time application, the frequency response of the filter is the Fourier transform of the
filter's impulse response, and a linear phase version has the form:
Detailed content of the Lecture:

Linear-phase FIR filter can be divided into four basic types.

Type Impulse response

I symmetric length is odd
II symmetric length is even
III anti-symmetric length is odd
IV anti-symmetric length is even

When h(n) is nonzero for 0 ≤ n ≤ N −1 (the length of the impulse response h(n) is N), then the symmetry of
the impulse response can be written as h(n) = h(N − 1 − n) and the anti-symmetry can be written as h(n) =
−h(N − 1 − n).

TYPE I: ODD-LENGTH SYMMETRIC

The frequency response of a length N = 5 FIR Type I filter can be written as follows
TYPE II: EVEN-LENGTH SYMMETRIC

The frequency response of a length N = 4 FIR Type II filter can be written as follows.
TYPE III: ODD-LENGTH ANTI-SYMMETRIC
TYPE IV: EVEN-LENGTH ANTI-SYMMETRIC

The frequency response of a length N = 4 FIR Type IV filter can be written as follows.

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L-21
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Finite Impulse Response filter

Introduction :

 Linear phase is a property of a filter, where the phase response of the filter is a linear function of
frequency.
 The result is that all frequency components of the input signal are shifted in time (usually delayed) by
the same constant amount (the slope of the linear function), which is referred to as the group delay.
 There is no phase distortion due to the time delay of frequencies relative to one another.
 For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter by having coefficients which are symmetric or anti-symmetric
Prerequisite knowledge for Complete understanding and learning of Topic:

 A filter is called a linear phase filter if the phase component of the frequency response is a linear
function of frequency.
 For a continuous-time application, the frequency response of the filter is the Fourier transform of the
filter's impulse response, and a linear phase version has the form:
Detailed content of the Lecture:
TYPE I: ODD-LENGTH SYMMETRIC

The frequency response of a length N = 5 FIR Type I filter can be written as follows

TYPE II: EVEN-LENGTH SYMMETRIC

The frequency response of a length N = 4 FIR Type II filter can be written as follows.

TYPE III: ODD-LENGTH ANTI-SYMMETRIC

TYPE IV: EVEN-LENGTH ANTI-SYMMETRIC

The frequency response of a length N = 4 FIR Type IV filter can be written as follows.
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L-22
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Fourier Series

Introduction :

 Fourier series is, in some way a combination of the Fourier sine and Fourier cosine series.
 The Fourier sine/cosine series are not the series will actually converge to f(x) or not at this point.
 The process of deriving the weights that describe a given function is a form of Fourier analysis. For
functions on unbounded intervals, the analysis and synthesis analogies are Fourier transform and
inverse transform.
Prerequisite knowledge for Complete understanding and learning of Topic:

A Fourier series is an expansion of a periodic function f(x) in terms of an infinite sum of sines and cosines.
Fourier series make use of the orthogonality relationships of the sine and cosine functions. The computation
and study of Fourier series is known as harmonic analysis and is extremely useful as a way to break up an
arbitrary periodic function into a set of simple terms that can be plugged in, solved individually, and then
recombined to obtain the solution to the original problem or an approximation to it to whatever accuracy is
desired or practical.The mathematical expression

is called fourier series.

Detailed content of the Lecture:

 One of the successful methods, for the design of FIR filters is based on the application of the Fourier
series.
 In this method, it is observed that the frequency of an FIR filter is a periodic function of frequency
with a period equal to the sampling frequency and consequently, it can be expressed in terms of the
Fourier series.
 The Fourier series by itself does not lead to satisfactory results but by using the Fourier series in
conjunction with a special class of functions known as window functions, good results can be
obtained.
 Approximation obtained by this method is suboptimal but the amounts of design effort and
computation required are relatively insignificant.
The frequency response of an FIR digital filter can be represented by the Fourier series as,

where the Fourier coefficients h(n) are the desired impulse response sequence of the filter, which can be
determined from

If we substitute e)“ = z, we obtain the transfer function ofthe digital filter, that is,

Therefore, if an expression is available for the frequency response, a transfer function can be obtained, which
happens to be a non causal and of infinite order. A finite order transfer 35 function can be obtained by
truncating the Fourier series. This can be accomplished by letting,
  This modification does not change the amplitude response of the filter, however the abrupt truncation
of the Fourier series results in oscillations in the passband and stopband.
 These oscillations are due to slow convergence of the Fourier series, particularly near points of
discontinuity. This effect is known as the Gibbs phenomenon

The amplitude of Gibbs oscillations can be reduced using discrete window functions.

A window function, represented by w(n) has the following time-domain properties:

1. w(n) = 0 for | n j> (N-1)/2

2. It is symmetrical, i.e., w(-n) = w(n).

The application of a window function consists of multiplying the impulse response obtained by applying the
Fourier series by the window function to obtain a modified impulse response as

hw(n) = w(n) h(n)

Since the window function is of finite duration, a finite-order transfer function given by
Through the application of the complex convolution, the frequency response of the modified filter can be
expressed as,

The application of the window function has two effects on the amplitude response of the filter. First, the
amplitudes of Gibbs’ oscillations in the pass bands and stop bands are directly related to the ripple ratio of the
window. Second, transition bands are introduced between pass bands and stop bands whose width is directly
related to the main-lobe width of the window.

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L-23
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 The window method for digital filter design is fast, convenient, and robust, but generally suboptimal.
 It is easily understood in terms of the convolution theorem for Fourier transforms, making it
instructive to study after the Fourier theorems and windows for spectrum analysis.
 It can be effectively combined with the frequency sampling method.
 The window method consists of simply ``windowing'' a theoretically ideal filter impulse
 response by some suitably chosen window function ,

 Window functions are always time limited. This means there is always a finite integer such

that for all .

The final windowed impulse response is thus always time-limited

Detailed content of the Lecture:

The unit sample response hd(n) obtained from the above relation is infinite in duration, so it must be
truncated at some point say n= M-1 to yield an FIR filter of length M (i.e. 0 to M-1). This truncation of hd(n)
to length M-1 is same as multiplying hd(n) by the rectangular window defined as

Thus the unit sample response of the FIR filter becomes

The multiplication of the window function w(n) with hd(n) is equivalent to convolution of Hd(w) with W(w),
where W(w) is the frequency domain representation of the window function

Thus the convolution of Hd(w) with W(w) yields the frequency response of the truncated FIR filter
But direct truncation of hd(n) to M terms to obtain h(n) leads to the Gibbs phenomenon effect which
manifests itself as a fixed percentage overshoot and ripple before and after an approximated discontinuity in
the frequency response due to the non-uniform convergence of the fourier series at a discontinuity. Thus the
frequency response obtained by using (8) contains 3 ripples in the frequency domain. In order to reduce the
ripples, instead of multiplying hd(n) with a rectangular window w(n), hd(n) is multiplied with a window
function.

The several effects of windowing the Fourier coefficients of the filter on the result of the frequency response
of the filter are as follows:

(i) A major effect is that discontinuities in H(w) become transition bands between values on either side of the
discontinuity.

(ii) The width of the transition bands depends on the width of the main lobe of the frequency response of the
window function, w(n) i.e. W(w).

(iii) Since the filter frequency response is obtained via a convolution relation, it is clear that the resulting
filters are never optimal in any sense.

(iv) As M (the length of the window function) increases, the main lobe width of W(w) is reduced which
reduces the width of the transition band, but this also introduces more ripple in the frequency response.

(v) The window function eliminates the ringing effects at the band edge and does result in lower side lobes at
the expense of an increase in the width of the transition band of the filter.
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L-24
LECTURE HANDOUTS

ECE III/V
Course Name with Code : 16ECD09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : FIR FILTER DESIGN Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Rectangular window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample response
hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function to
be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
 These two requirements are interrelated .
  Window Technique implicates a function called window Function.
 It is also known as Tapering Function.

Detailed content of the Lecture:

Rectangular Window

The rectangular window (sometimes known as the Boxer or Dirichlet window) is the simplest window,
equivalent to replacing all but N values of a data sequence by zeros, making it appear as though the waveform
suddenly turns on and off.

where is the window length in samples (assumed odd for now). A plot of the rectangular window appears
in Fig.3.1 for length . It is sometimes convenient to define windows so that their dc gain is 1, in

which case we would multiply the definition above by

also called the Dirichlet function or periodic sinc function. This (real) result is for the zero-centered rectangular
window. For the causal case, a linear phase term appears:

The term ``aliased sinc function'' refers to the fact that it may be simply obtained by sampling the length-

continuous-time rectangular window, which has Fourier transform sinc (given

amplitude in the time domain). Sampling at intervals of seconds in the time domain corresponds to

aliasing in the frequency domain over the interval Hz, and by direct derivation. The window duration
increases continuously in the time domain: the magnitude spectrum can only change in discrete jumps as new
samples are included, even though it is continuously parametrized in .As the sampling rate goes to infinity,
the aliased sinc function therefore approaches the sinc function.

Figure: Fourier transform of the rectangular window.

The phase of rectangular-window transform is zero for , which is the width of
the main lobe. This is why zero-centered windows are often called zero-phase windows; while the phase
actually alternates between 0 and radians, the values occur only within side-lobes which are routinely
neglected.

Figure: Magnitude (dB) of the rectangular-window transform

Figure: Magnitude of the rectangular-window Fourier transform.

 Figure: Phase of the rectangular-window Fourier transform.

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L-25
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Hamming window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function
to be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
 These two requirements are interrelated .
 Window Technique implicates a function called window Function.
  It is also known as Tapering Function.

Detailed content of the Lecture:

Hamming Window

The Hamming window is determined by choosing (with ) to cancel the largest side
lobe. Doing this results in the values

The peak side-lobe level is approximately dB for the Hamming window . It happens that this choice

is very close to that which minimizes peak side-lobe level (down to dB--the lowest possible within
the generalized Hamming family).

Since rounding the optimal to two significant digits gives , the Hamming window can be
considered the ``Chebyshev Generalized Hamming Window'' (approximately). Chebyshev-type designs
normally exhibit equiripple error behavior, because the worst-case error (side-lobe level in this case) is
minimized. Generalized Hamming windows can have a step discontinuity at their endpoints, but no impulsive
points.

 The Hamming window and its DTFT magnitude are shown in Figure. The Hamming window is also
one period of a raised cosine.
 However, the cosine is raised so high that its negative peaks are above zero, and the window has
a discontinuity in amplitude at its endpoints (stepping discontinuously from 0.08 to 0).
 This is 10 dB better than the Hann case of Figure and 28 dB better than the rectangular window.
 The main lobe is approximately wide, as is the case for all members of the generalized Hamming
family
For the Hamming window, the side-lobes nearest the main lobe have been strongly shaped by the

optimization. As a result, the nearly dB per octave roll-off occurs only over an interior interval of
the spectrum, well between the main lobe and half the sampling rate. The optimized side-lobes nearest the
main lobe occupy a smaller frequency interval about the main lobe.

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L-26
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Filter design using Windowing technique-Hanning window

Introduction :

 There are essentially three well-known methods for FIR filter design namely:
(1) The window method
(2) The frequency sampling technique
(3) Optimal filter design methods
The Window Method:
 In this method the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

Prerequisite knowledge for Complete understanding and learning of Topic:

 A window is a finite array consisting of coefficients selected to satisfy the desirable requirements.
 While designing digital FIR filter using window function it is necessary to specify a window function
to be used and the filter order according to the required specifications (selectivity and stop band
attenuation).
  These two requirements are interrelated .
 Window Technique implicates a function called window Function.
 It is also known as Tapering Function.
Detailed content of the Lecture:

Hanning Window

 Hanning window is the shape of one cycle of a cosine wave with 1 added to it so it is always positive.
 The sampled signal values are multiplied by the Hanning function.
 The ends of the time record are forced to zero regardless of what the input signal is doing.
 While the Hanning window does a good job of forcing the ends to zero, it also adds distortion to the
wave form being analyzed in the form of amplitude modulation; i.e., the variation in amplitude of the
signal over the time record.
 Amplitude Modulation in a wave form results in sidebands in its spectrum, and in the case of the
Hanning window, these sidebands, or side lobes as they are called, effectively reduce the frequency
resolution of the analyzer by 50%.
 It is as if the analyzer frequency "lines" are made wider.
 The highest-level side lobes are about 32 dB down from the main lobe.

 The Hanning window should always be used with continuous signals, but must never be used with
transients. The reason is that the window shape will distort the shape of the transient, and
the frequency and phase content of a transient is intimately connected with its shape.
 The measured level will also be greatly distorted. Even if the transient were in the center of
the Hanning window, the measured level would be twice as great as the actual level because of
the amplitude correction the analyzer applies when using the Hanning weighting.
 A Hanning weighted signal actually is only half there, the other half of it having been removed by the
windowing. This is not a problem with a perfectly smooth and continuous signal like a sinusoid, but
most signals we want to analyze, such as machine vibration signatures are not perfectly smooth.
 If a small change occurs in the signal near the beginning or end of the time record, it will either be
analyzed at a much lower level than its true level, or it may be missed altogether.
  For this reason, it is a good idea to employ overlap processing. To do this, two time buffers are
required in the analyzer.
 For 50% overlap, the sequence of events is as follows:
 When the first buffer is half full, i.e., it contains half the samples of a time record, the second buffer is
connected to the data stream and also begins to collect samples. As soon as the first buffer is full,
the FFT is calculated, and the buffer begins to take data again.
 When the second buffer is filled, the FFT is again calculated on its contents, and the result sent to
the spectrum-averaging buffer. This process continues on until the desired number of averages is
collected.

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L-27
LECTURE HANDOUTS

ECE II/IV
Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : DESIGN OF FIR FILTER Date of Lecture :

Topic of Lecture: Frequency sampling method

Introduction :

 The frequency-sampling method for FIR filter design is perhaps the simplest and most direct
technique imaginable when a desired frequency response has been specified.
 It consists simply of uniformly sampling the desired frequency response, and performing an
inverse DFT to obtain the corresponding (finite) impulse response.
 The results are not optimal, however, because the response generally deviates from what is
desired between the samples.
 When the desired frequency-response is undersampled, which is typical, the
resulting impulse response will be time aliased to some extent.
Prerequisite knowledge for Complete understanding and learning of Topic:

 The main idea of the frequency sampling design method is that a desired frequency response can be
approximated by sampling it of N evenly spaced points and then obtaining an interpolated frequency
response that passes through the frequency samples.
 For filters with reasonably smooth frequency responses,the interpolation error is generally small.
 In the case of band select filters, where the desired frequency response changes radically across
bands, the frequency samples which occur in transition bands are made to be unspecified variables
whose values are chosen by an optimization algorithm which minimizes some function of the
 approximation error of the filter finally, it was shown that there were two distinct types of frequency
sampling filters, depending on where the initial frequency sample.

Detailed content of the Lecture:

The frequency sampling method allows us to design recursive and nonrecursive FIR filters for both standard
frequency selective and filters with arbitrary frequency response.

No recursive frequency sampling filters :

The problem of FIR filter design is to find a finite–length impulse response h (n) that corresponds to desired
frequency response. In this method h (n) can be determined by uniformly sampling, the desired frequency
response HD (ω) at the N points and finding its inverse DFT of the frequency samples

where H (k), k = 0, 1, 2,……., N-1, are samples of the HD (ω) . For linear phase filters, with positive
symmetrical impulse response, we can write

where  = (N-1)/2. For N odd, the upper limit in the summation is (N - 1)/2, to obtain a good approximation
to the desired frequency
Recursive frequency sampling filter :
In recursive frequency sampling method the DFT samples H (k) for an FIR sequence can be regarded as
samples of the filters z– transform, evaluated at N points equally spaced around the unit circle.

thus the z–transform of an FIR filter can easily be expressed in terms of its DFT coefficients,
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L-37
LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./EC

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Multirate signal processing

Introduction :
The sampling rate of a signal is changed in order to increase the efficiency of various
signal processing operations. Decimation, or down-sampling, reduces the sampling
rate, whereas expansion, or up-sampling, followed by interpolation increases the
sampling rate.
Prerequisite knowledge for Complete understanding and learning of Topic:

 Signal Processing

Detailed content of the Lecture:

Up-sampling
 Increasing the sampling frequency, before D/A conversion in order to relax the
requirements of the analog lowpass antialiasing filter.
 This technique is used in audio CD, where the sampling frequency 44.1 kHz is increased
fourfold to 176.4 kHz before D/A conversion.
Decomposition
 Decomposition of a signal into M components containing various frequency bands. If the
original signal is sampled at the sampling frequency fs (with a frequency band of width
fs/2, or half the sampling frequency), every component then contains a frequency band of
width 1 2 fs/M only, and can be represented using the sampling rate fs/M.
 This allows for efficient parallel signal processing with processors operating at lower
sampling rates. The technique is also applied to data compression in subband coding, for
example in speech processing, where the various frequency band components are
represented with different word lengths.
 In the implementation of high-performance filtering operations, where a very narrow
transition band is required. The requirement of narrow transition bands leads to very high
filter orders.
 However, by decomposing the signal into a number of subbands containing the passband,
stopband and transition bands, each component can be processed at a lower rate, and the
transition band will be less narrow. Hence the required filter complexity may be reduced
significantly.
Subband decomposition
 In order to present the basic techniques involved in decomposing a signal into subband
components, let’s consider a simple case where a signal is decomposed into two components: a
low-frequency component and a high-frequency component.
 The purpose of the filters H1 and H2 is to extract the low- and high-frequency components of
the signal x. For perfect signal decomposition, H1 should be an ideal low-pass filter with the
passband [0, π/2], and H2 should be an ideal high-pass filter with the passband [π/2, π], cf.
XD1(ω) = 1 / 2 X(ω/2), 0 ≤ ω < π
XD2(ω) = 1/ 2 X(π − ω/2)∗ , 0 ≤ ω < π

 Real filters characteristics resemble more the general form. It follows that it is not possible to
separate the frequency bands exactly, but instead either some aliasing between the frequency
bands is unavoidable, or, if the frequencies at the band edges are attenuated completely, some
frequencies are lost altogether.
 The standard solution to the aliasing problem is to design the filters H1 and H2 in such a way
that despite aliasing, it is still possible to reconstruct the original signal from the components.
This can be achieved with quadrature mirror filters.
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L-38
LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Decimation

Introduction :
 Decimation is the process of reducing the sampling rate of a signal.
 The term downsampling usually refers to one step of the process, but sometimes the
terms are used interchangeably. Complementary to upsampling, which increases
sampling rate, decimation is a specific case of sample rate conversion in a multi-rate
digital signal processing system. A system component that performs decimation is
called a decimator
Prerequisite knowledge for Complete understanding and learning of Topic:

 Down Sampling
 Up Samling
Detailed content of the Lecture:
 When decimation is performed on a sequence of samples of a signal or other continuous
function, it produces an approximation of the sequence that would have been obtained by
sampling the signal at a lower rate (or density, as in the case of a photograph).
 The decimation factor is usually an integer or a rational fraction greater than one. This factor
multiplies the sampling interval or, equivalently, divides the sampling rate. For example,
if compact disc audio at 44,100 samples/second is decimated by a factor of 5/4, the
resulting sample rate is 35,280.
Decimation by an integer factor
 Decimation by an integer factor, M, can be explained as a two-step process, with an equiv-
alent implementation that is more efficient:
 Reduce high-frequency signal components with a digital lowpass filter.
 Downsample the filtered signal by M; that is, keep only every Mth sample.
 Downsampling alone causes high-frequency signal components to be misinterpreted by
subsequent users of the data, which is a form of distortion called aliasing. The first step, if
necessary, is to suppress aliasing to an acceptable level. In this application, the filter is
called an anti-aliasing filter, and its design is discussed below. Also see undersampling for
information about downsampling bandpass functions and signals.
 When the anti-aliasing filter is an IIR design, it relies on feedback from output to input,
prior to the downsampling step.
  With FIR filtering, it is an easy matter to compute only every Mth output. The calculation
performed by a decimating FIR filter for the nth output sample is a dot product:
Y(n) = ∑x[nM-k].h(k)
 where the h[k] sequence is the impulse response, and K is its length. x[k] represents the
input sequence being downsampled. In a general purpose processor, after
computing y[n], the easiest way to compute y[n+1] is to advance the starting index in
the x[k] array by M, and recompute the dot product. In the case M=2, h[k] can be
designed as a half-band filter, where almost half of the coefficients are zero and need not
be included in the dot products.
 Impulse response coefficients taken at intervals of M form a subsequence, and there
are M such subsequences (phases) multiplexed together. The dot product is the sum of
the dot products of each subsequence with the corresponding samples of the x[k]
sequence.
 Furthermore, because of downsampling by M, the stream of x[k] samples involved in any
one of the M dot products is never involved in the other dot products. Thus M low-order
FIR filters are each filtering one of M multiplexed phases of the input stream, and
the M outputs are being summed.
 In other words, the input stream is demultiplexed and sent through a bank of M filters
whose outputs are summed. When implemented that way, it is called a polyphase filter.
 For completeness, we now mention that a possible, but unlikely, implementation of each
phase is to replace the coefficients of the other phases with zeros in a copy of the h[k]
array, process the original x[k] sequence at the input rate, and decimate the output by a
factor of M.
 The equivalence of this inefficient method and the implementation described above is
known as the first Noble identity
Anti-aliasing filter
 The requirements of the anti-aliasing filter can be deduced from any of the three pairs of
graphs.
 Note that all three pairs are identical, except for the units of the abscissa variables. The
upper graph of each pair is an example of the periodic frequency distribution of a sampled
function, x(t), with Fourier transform, X(f).
 The lower graph is the new distribution that results when x(t) is sampled three times
slower, or (equivalently) when the original sample sequence is decimated by a factor
of M=3.

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LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Interpolation

Introduction :
 Interpolation is a type of estimation, a method of constructing new data points within
the range of a discrete set of known data points.
 It is often required to interpolate, i.e., estimate the value of that function for an
intermediate value of the independent variable.
Prerequisite knowledge for Complete understanding and learning of Topic:

 Down Sampling
 Up Samling
Detailed content of the Lecture:

Linear interpolation
 Generally, linear interpolation takes two data points, say (xa,ya) and (xb,yb), and the
interpolant is given by:

 Linear interpolation is quick and easy, but it is not very precise. Another disadvantage is
that the interpolant is not differentiable at the point xk.
 The following error estimate shows that linear interpolation is not very precise. Denote the
function which we want to interpolate by g, and suppose that x lies between xa and xb and
that g is twice continuously differentiable.
  Then the linear interpolation error is

Interpolation of Finite Set Points

Polynomial interpolation
 Polynomial interpolation is a generalization of linear interpolation. Note that the linear
interpolant is a linear function. We now replace this interpolant with a polynomial of
higher degree.

 Polynomial interpolation also has some disadvantages. Calculating the interpolating

polynomial is computationally expensive (see computational complexity) compared to
linear interpolation. Furthermore, polynomial interpolation may exhibit oscillatory
artifacts, especially at the end points.
 Polynomial interpolation can estimate local maxima and minima that are outside the
range of the samples, unlike linear interpolation.

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LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Cascading Sample Rate Converters

Introduction :
 Sample-rate conversion is the process of changing the sampling rate of a discrete signal to
obtain a new discrete representation of the underlying continuous signal.

Prerequisite knowledge for Complete understanding and learning of Topic:

 Down Sampling
 Up Samling
Detailed content of the Lecture:

Sample-rate conversion is the process of changing the sampling rate of a discrete signal to
obtain a new discrete representation of the underlying continuous signal. Application
areas include image scaling and audio/visual systems, where different sampling rates
may be used for engineering, economic, or historical reasons.
1. Conceptual approaches to sample-rate conversion include: converting to an analog
continuous signal, then re-sampling at the new rate, or calculating the values of the new
samples directly from the old samples.
2. If the ratio of the two sample rates is (or can be approximated by) a fixed rational
number L/M: generate an intermediate signal by inserting L − 1 0s between each of the
original samples. Low-pass filter this signal at half of the lower of the two rates. Select
every M-th sample from the filtered output, to obtain the result.
3. Treat the samples as geometric points and create any needed new points by interpolation.
Choosing an interpolation method is a trade-off between implementation complexity and
conversion quality (according to application requirements). Commonly used
are: ZOH (for film/video frames), cubic (for image processing) and windowed sinc
function (for audio).
Anti-aliasing
 Oversampling can make it easier to realize analog anti-aliasing filters. Without
oversampling, it is very difficult to implement filters with the sharp cutoff necessary to
maximize use of the available bandwidth without exceeding the Nyquist limit.
 By increasing the bandwidth of the sampling system, design constraints for the anti-
aliasing filter may be relaxed.
  Once sampled, the signal can be digitally filtered and downsampled to the desired
sampling frequency. In modern integrated circuit technology, the digital filter associated
with this downsampling are easier to implement than a comparable analog filter required
by a non-oversampled system.
Resolution
 In practice, oversampling is implemented in order to reduce cost and improve
performance of an analog-to-digital converter (ADC) or digital-to-analog
converter (DAC).[1] When oversampling by a factor of N, the dynamic range also increases
a factor of N because there are N times as many possible values for the sum.
Noise
 If multiple samples are taken of the same quantity with uncorrelated noise added to each
sample, then because, as discussed above, uncorrelated signals combine more weakly
than correlated ones, averaging N samples reduces the noise power by a factor of N.
 If, for example, we oversample by a factor of 4, the signal-to-noise ratio in terms of power
improves by factor of 4 which corresponds to a factor of 2 improvement in terms of
voltage.
Oversampling in reconstruction
 The term oversampling is also used to denote a process used in the reconstruction phase of
digital-to-analog conversion, in which an intermediate high sampling rate is used between
the digital input and the analogue output.
 Digital interpolation is used to add additional samples between recorded samples, thereby
converting the data to a higher sample rate, a form of upsampling.
 When the resulting higher-rate samples are converted to analog, a less complex and less
expensive analog reconstruction filter is required. Essentially, this is a way to shift some of
the complexity of reconstruction from analog to the digital domain.
 Oversampling in the ADC can achieve some of the same benefits as using a higher sample
rate at the DAC.
 In signal processing, undersampling or bandpass sampling is a technique where
one samples a bandpass-filtered signal at a sample rate below its Nyquist rate (twice the
upper cutoff frequency), but is still able to reconstruct the signal.
 When one undersamples a bandpass signal, the samples are indistinguishable from the
samples of a low-frequency alias of the high-frequency signal. Such sampling is also
known as bandpass sampling, harmonic sampling, IF sampling, and direct IF-to-digital
conversion

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L-41
LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Efficient Transversal Structure for Decimator

Introduction :
 Decimation and interpolation are the two basic building blocks in the multirate digital signal
processing systems. As the linear canonical transform (LCT) has been shown to be a
powerful tool for optics and signal processing, it is worthwhile and interesting to analyze
the decimation and interpolation in the LCT domain. .
Prerequisite knowledge for Complete understanding and learning of Topic:

 Decimator

Detailed content of the Lecture:


Important Books/Journals for further learning including the page nos.:
5. Digital Signal Processing Principles Algorithms & Applications, John G. Proakis &
Dimitris G.Manolakis, 4th Edition. Pg.No(784-789)

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LECTURE HANDOUTS

ECE
./ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Efficient Transversal Structure for Interpolator

Introduction :
 Decimation and interpolation are the two basic building blocks in the multirate digital
signal processing systems. As the linear canonical transform (LCT) has been shown to be
a powerful tool for optics and signal processing, it is worthwhile and interesting to
analyze the decimation and interpolation in the LCT domain. .
Prerequisite knowledge for Complete understanding and learning of Topic:

 Interpolation

Detailed content of the Lecture:

Important Books/Journals for further learning including the page nos.:
6. Digital Signal Processing Principles Algorithms & Applications, John G. Proakis &
Dimitris G.Manolakis, 4th Edition. Pg.No(784-789)

Course Teacher

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 MUTHAYAMMAL ENGINEERING COLLEGE
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L-43
LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Adaptive Filters: Introduction, Applications of adaptive filtering to

equalization

Introduction :
 An adaptive filter is a system with a linear filter that has a transfer function controlled by
variable parameters and a means to adjust those parameters according to an optimization
algorithm. Because of the complexity of the optimization algorithms, almost all adaptive
filters are digital filters.
 Adaptive filters are required for some applications because some parameters of the
desired processing operation (for instance, the locations of reflective surfaces in
a reverberant space) are not known in advance or are changing. The closed loop adaptive
filter uses feedback in the form of an error signal to refine its transfer function.
Prerequisite knowledge for Complete understanding and learning of Topic:

 Ideal Filters
 Linear Filtering
Detailed content of the Lecture:

 There are two input signals to the adaptive filter: and which are sometimes
called the primary input and the reference input respectively.
 The adaption algorithm attempts to filter the reference input into a replica of the desired
input by minimizing the residual signal, When the adaption is successful, the output of the
filter is effectively an estimate of the desired signal.
 which includes the desired signal plus undesired interference and are correlated to some
of the undesired interference in k represents the discrete sample number. The filter is
controlled by a set of L+1 coefficients or weights.

Tapped delay line FIR filter

 If the variable filter has a tapped delay line Finite Impulse Response (FIR) structure, then
the impulse response is equal to the filter coefficients. The output of the filter is given by

Yk = ∑ wlk x(k-l)
Adaptive Linear Combiner
 The adaptive linear combiner (ALC) resembles the adaptive tapped delay line FIR filter
except that there is no assumed relationship between the X values.

 If the X values were from the outputs of a tapped delay line, then the combination of
tapped delay line and ALC would comprise an adaptive filter. However, the X values
could be the values of an array of pixels. Or they could be the outputs of multiple tapped
delay lines. The ALC finds use as an adaptive beam former for arrays of hydrophones or
antennas.
Yk = ∑ Wlk Xlk
Convergence
 Here μ(convergence) controls how fast and how well the algorithm converges to the
optimum filter coefficients. If μ is too large, the algorithm will not converge.
 If μ is too small the algorithm converges slowly and may not be able to track changing
conditions. If μ is large but not too large to prevent convergence, the algorithm reaches
steady state rapidly but continuously overshoots the optimum weight vector.
 Sometimes, μ is made large at first for rapid convergence and then decreased to minimize
overshoot.
Nonlinear Adaptive Filters
 The goal of nonlinear filters is to overcome limitation of linear models. There are some
commonly used approaches: Volterra LMS, Kernel adaptive filter, Spline Adaptive
Filter and Urysohn Adaptive Filter. The general idea behind Volterra LMS and Kernel
LMS is to replace data samples by different nonlinear algebraic expressions.
Yi = ∑ Wj Xij
Applications of Adaptive Filtering
 Linear Predictor
 Inverse Modeling
 Echo Cancellation
 Foetal Monitoring
 Noise Cancellation Filters

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L-44
LECTURE HANDOUTS

ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture: Subband Coding

Introduction :
 Sub-band coding (SBC) is any form of transform coding that breaks a signal into a
number of different frequency bands, typically by using a fast Fourier transform, and
encodes each one independently. This decomposition is often the first step in data
compression for audio and video signals.
Prerequisite knowledge for Complete understanding and learning of Topic:

 Linear Prediction

Detailed content of the Lecture:

Encoding audio signals

 The simplest way to digitally encode audio signals is pulse-code modulation (PCM),
which is used on audio CDs, DAT recordings, and so on.
 Digitization transforms continuous signals into discrete ones by sampling a signal's
amplitude at uniform intervals and rounding to the nearest value representable with the
available number of bits. This process is fundamentally inexact, and involves two
errors: discretization error, from sampling at intervals, and quantization error, from
rounding.
 The more bits used to represent each sample, the finer the granularity in the digital
representation, and thus the smaller the quantization error. Such quantization errors may
be thought of as a type of noise, because they are effectively the difference between the
original source and its binary representation.
 With PCM, the audible effects of these errors can be mitigated with dither and by using
enough bits to ensure that the noise is low enough to be masked either by the signal itself
or by other sources of noise.
 A high quality signal is possible, but at the cost of a high bitrate (e.g., over 700 kbit/s for
one channel of CD audio). In effect, many bits are wasted in encoding masked portions of
the signal because PCM makes no assumptions about how the human ear hears.
 Coding techniques reduce bitrate by exploiting known characteristics of the auditory
system. A classic method is nonlinear PCM, such as the μ-law algorithm. Small signals are
digitized with finer granularity than are large ones; the effect is to add noise that is
 proportional to the signal strength.
Basic principles
 The utility of SBC is perhaps best illustrated with a specific example. When used for audio
compression, SBC exploits auditory masking in the auditory system.
 Human ears are normally sensitive to a wide range of frequencies, but when a sufficiently
loud signal is present at one frequency, the ear will not hear weaker signals.
 The basic idea of SBC is to enable a data reduction by discarding information about
frequencies which are masked. The result differs from the original signal, but if the
discarded information is chosen carefully, the difference will not be noticeable, or more
importantly, objectionable.
 A digital filter bank divides the input signal spectrum into some number of subbands. The
psychoacoustic model looks at the energy in each of these subbands, as well as in the
original signal, and computes masking thresholds using psychoacoustic information.
 Each of the subband samples is quantized and encoded so as to keep the quantization
noise below the dynamically computed masking threshold.
 The final step is to format all these quantized samples into groups of data called frames, to
facilitate eventual playback by a decoder.
 Decoding is much easier than encoding, since no psychoacoustic model is involved. The
frames are unpacked, subband samples are decoded, and a frequency-time mapping
reconstructs an output audio signal.
Applications
 Beginning in the late 1980s, a standardization body, the Moving Picture Experts
Group (MPEG), developed standards for coding of both audio and video. Subband coding
resides at the heart of the popular MP3 format (more properly known as MPEG-1 Audio
Layer III), for example.
 Sub-band coding is used in the G.722 codec which uses sub-band adaptive differential
pulse code modulation (SB-ADPCM) within a bit rate of 64 kbit/s.
 In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and
lower) and the signals in each sub-band are encoded using ADPCM.

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ECE II/IV

Course Name with Code : 19ECC09/ DIGITAL SIGNAL PROCESSING

Course Teacher : Dr.T.R.Ganesh Babu. Prof./ECE

Unit : MULTIRATE AND DIGITAL SIGNAL PROCESSORS

Date of Lecture :

Topic of Lecture : Channel Vocoders

Introduction :
 Channel vocoder is a device for compressing, or encoding, the data needed to represent a
speech waveform, while still retaining the intelligibilty of the original waveform.

Prerequisite knowledge for Complete understanding and learning of Topic:

 Sub Band Coding

Detailed content of the Lecture:

 The energy of each filter's output was then measured, or sampled, at regular time intervals
and stored. This collection of filter energy samples then comprised the "coding" of the
speech signal.
 This code could then be transmitted over a communication channel of lower bandwidth
than would be neccessary for the raw speech signal. At the receiving end, the speech
signal is reconstructed from this code by using the time sequence of filter energy samples
to modulate the amplitude of a pulse signal being fed into a bank of filters similar to the
ones used for the encoding.
 The channel vocoder, then, first analyzes the speech signal to estimate this time-varying
spectral variation. To do this it uses the filters in the filter banks to determine how a
particular frequency component of the speech signal is changing with time.
 On the output end, this analysis of the spectral variations are used to synthesize the
speech signal by using another filter bank to apply these same spectral variations to an
artificial periodic pulse like signal. The output filter bank acts as an artificial vocal track
and the pulse signal acts as a set of artificial vocal chords.
 Some sounds produced during speech do not arise from the vocal chords, but are
produced by turbulent air flow near constrictions in the vocal tract such as may occur
between the tongue and the teeth. For example, such sounds as "SSS", "K", "SSHH", "P",
and so forth arise in this manner.
 These sounds would be poorly reconstructed using a pulse excitation source, and so most
channel vocoders also have a noise signal that can be used as an excitation source as well.
A "Voiced/Unvoiced" detector circuit is used to detect whether the speech signal is arising
from vocal chord excitation (Voiced speech) or is arising from noise excitation (Unvoiced
 speech), and the appropriate excitation source is then selected at the output end.
 Channel vocoders were originally developed for signal coding purposes, with an eye
(ear?) towards reducing the amount of data that would be needed to be transmitted over
communication channels. In fact, speech coding system development continues to this day
to be a vigorous area of research and development.
 These systems have far outstripped the basic channel vocoder idea in complexity, coding
efficiency, and intelligibilty, however. So why do we still care about channel vocoders?
The reason is that channel vocoders (and the functionally equivalent, but computationally
quite different, phase vocoder) have found application to music production.

 The vowel sounds are pitched and have a definite spectral structure. The vocal filter
module is designed to produce the spectrum of vowel sounds. Vowel sounds are created
by passing a periodic pulse waveform (which sets the basic pitch of the voice) through a
complex filter with multiple resonances.
 The pulse waveform models the vibration of the vocal cords. In the patches below we use
a narrow pulse wave, but half-wave rectified sine-waves and sawtooth waves could also
be used.
 The complex filter models the effect of the vocal cavity, formed by the mouth, throat, and
nasal passages. Because of the complicated shape of these passages, some frequencies are
enhanced while others are diminished in strength. This results in resonances and anti-
resonances.

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