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Course File
Course File
R-16
IV B. Tech
2019-20
Course File Prepared by
P.USHA
Assistant Professor
DIGITAL SIGNAL PROCESSING(EE721PE)
a) Course Objectives
1. To provide background and fundamental material for the analysis and processing of digital
signals.
2. To familiarize the relationships between continuous time and discrete time signals and systems.
3. To study fundamentals of time, frequency and Z-plane analysis and to discuss the inter-
relationships of these analytic method.
4. To study the designs and structures of digital (IIR and FIR) filters from analysis to synthesis for a
given specifications.
5. The impetus is to introduce a few real-world signal processing applications.
6. To acquaint in FFT algorithms, Multi-rate signal processing technique and finite word length
effects.
b) Course Outcomes
c) Topic outcomes
S.No Topic Topic outcome At the end of the topic the
student will be able to
1 UNIT-1 Introduction to DSP Explain a digital processing system.
2 Discrete time signals and sequences Compare discrete time signal and
sequence.
3 Linear shift invariant systems Describe a LTI system.
4 problems
5 Stability and Casuality Describe stability and causality of
system.
6 Linear constant coefficient difference Solve the difference equations.
equation
7 Frequency domain representation of DTS Demonstrate the frequency domain
representation of DTS.
8 Application of Z transform List the applications of Z transform.
9 Solution of difference equation of digital Solve the difference equation.
filters
10 System function, Explain stability criterion and its
frequency response.
11 Stability criterion Explain stability criterion and its
frequency response.
12 Frequency response of stable systems Explain stability criterion and its
frequency response.
13 Realization of digital filters-direct and
canonic form Design a digital filter using different
methods.
14 Cascade and parallel form Design a digital filter using different
methods.
15 problems Summarize the important concepts of
16 Revision, Discussion of previous question this unit.
papers
17 UNIT II-Discrete Fourier Series: DFS Describe the representation of DFS.
representation
18 Properties of DFS list the properties of DFS
19 DFT and Properties of DFT list the properties of DFT
20 Linear convolution using DFT Solve linear convolution using
different methods.
21 Overlap Add method Solve linear convolution using different
methods.
22 Overlap Save method Solve linear convolution using different
methods.
23 Relation between DTFT and DFS Analyze the relation between different
transforms.
4) COURSE PRE–REQUISITES
1. Basics of mathematics
2. Different types of signals and systems
3. Difference between continuous and discrete signals
CO PO1 PO2 PO3 PO4 PO5 PO6 PO7 PO8 PO9 PO10 PO11 PO12
CO1 3 3 1 3 3
CO2 3 3 1 3 3
CO3 3 3 3
CO4 3 3 3
CO5 2 2 1 2 1
Legends: 1 – Low
2 – Medium
3 – High
5.b). SYLLABUS:
Discrete Fourier series: DFS Representation of Periodic Sequences, properties of Discrete Fourier
Series, Discrete Fourier Transforms: Properties of DFT, Linear Convolution of Sequences using
DFT, Computation of DFT: Over-Lap Add Method, Over-Lap Save Method, Relation
II between DTFT, DFS, DFT and Z-Transform. 17
Fast Fourier Transforms: Fast Fourier Transforms (FFT) – Radix-2 Decimation-in-Time and
Decimation-in-Frequency FFT Algorithms, Inverse FFT, and FFT with General Radix-N.
IIR Digital Filters: Analog filter approximations – Butter worth and Chebyshev, Design of IIR
III Digital Filters from Analog Filters, Step and Impulse Invariant Techniques, Bilinear 10
Transformation Method, Spectral Transformations.
FIR Digital Filters: Characteristics of FIR Digital Filters, Frequency Response, Design of FIR
IV Filters: Fourier Method, Digital Filters using Window Techniques, Frequency Sampling Technique, 08
Comparison of IIR & FIR filters
Multi rate Digital Signal Processing: Introduction, Down Sampling Decimation, Up sampling,
Interpolation, Sampling Rate Conversion.Finite Word Length Effects: Limit cycles, Overflow
V Oscillations, Round-oft Noise in IIR Digital Filters, Computational Output Round Off Noise, 16
Methods to Prevent Overflow, Trade Off Between Round Off and Overflow Noise, Dead Band
Effects.
Contact classes for syllabus coverage 64
Lectures beyond syllabus 02
Tutorial classes 00
Classes for gaps&Add-on classes 02
Total No. of classes 68
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
2 Wavelet Transforms 1
b. http://nptel.ac.in/courses/117104070/9
c. http://nptel.ac.in/courses/117104070/6
d. http://nptel.ac.in/courses/117102060/39
☐ ADD-ON ☐ OTHERS
COURSES
Text Book Discrete Time Signal Processing — A. V. Oppenheim and R.W Schaffer,
PHI, 2009 Fundamentals of Digital Signal Processing — Loney Ludeman,
John Wiley, 2009
Unit-I
1 UNIT-1 Introduction to DSP
7 Application of Z transform
23 FFT
26 Inverse FFT
31 Revision
38 Spectral transformation
43 Frequency response
52 Revision
54 Decimation
55 Interpolation
59 Overflow oscillations
65 Problems
66 Revision
68 Wavelet Transforms
Book 1 Digital Signal Processing – Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G.Manolakis, Pearson Education / PHI, 2007.
Book 2 Discrete Time Signal Processing – A. V. Oppenheim and R.W.Schaffer, PHI, 2009.
Reference Books
Book 3 Digital Signal Processing – Fundamentals and Applications – Li Tan, Elsevier, 2008.
Book 4 Digital Signal Processing – S.Salivahanam, A.Vallavaraj and C.Gnanapriya, TMH, 2009.
Chapters Nos No of
classes
Unit Topic Book 1 Book 2 Book 3 Book 4
1 4
I Introduction to DSP
Tutorial classes 00
College Code
K. G. Reddy College of Engineering &Technology
Date &
Subject: DSP
Session
1. Find the total response of the system described by difference equation y(n)-4y(n- 1) +4y(n-2)
= x(n)-x(n- 1) when the input is x(n)= (-1)n u(n) with the initial condition y (-1) =y (-2) =1.
2. A) Find x(k) of the given sequence x(n) = {1,2,3,4,4,3,2,1} using DIT-FFT algorithm.
B) Determine the number of additions & multiplications required to implement 16-point DFT&
16 point FFT
3.A) Find the IDFT of the given sequence X (k) = {2, 2-3j, 2+3j,-2}
4.Derive the transfer function and order for the analog low pass butterworth filter
_____________________________________________________________________________
Bloom’s Course
Q.NO QUESTION
level outcome
Design a Chebyshev filter for the following
specifications using impulse invariance method
0.8 ≤ |H(ejw) | ≤ 1 , 0 ≤ w ≤ 0.2π
1 L6,L3 CO5
|H(ejw) | ≤ 0.2 , 0.6π≤ w ≤ π
2. The necessary and sufficient condition for causality of an LTI system is, [ ]
its unit sample response
a) h(n) = 0 for negative values of n b) h(n) = 0 for positive values of n
c) h(n) = 0 for integer values of n d) h(n) = 0 for complex values of n
6. Linear convolution of two real sequences with P and Q points, result for zero appended real sequence
after executing circular convolution using FFT will go wrong if FFT of length N is used such that
[ ]
a) N < P+Q-1 b) N=P+Q-1 c) N>P+Q-1 d) N<P+Q
7 he fundamental period of
𝜋𝑛 2𝑓0
x(n)=cos( ) is
8
12. A ___________ signal exhibits no un certainity of value at any given instant of time
14. The _____________ theorem states that the fourier transform of autocorrelation sequence is the
energy density spectrum
16. When a sequence is circularly shifted in time by 5 units, the magnitude response __________
18. In DIF the complex multiplication takes place after the ___________ operation
19. The transition band is ________ in butter worth filter compared to Chebyshev filter
2. a
3. c
4. a
5. a
6. b
7. b
8. c
9. c
10. a
12. Deterministic
13. Superposition
15. Zero
17. Sampling
18. Add-subtract
19. Less
20. Z-transform
Code No: 126VK Set No. 1
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD
B.Tech. III Year, II Sem., II Mid-Term Examinations, April-2018
DIGITAL SIGNAL PROCESSING
Objective Exam
Name: ______________________________ Hall Ticket No. A
Answer All Questions. All Questions Carry Equal Marks. Time: 20 Min. Marks: 10.
4.The impulse response, which is symmetric having odd number of samples can be used to design
[ ]
a) low pass b) High pass c) Band pass d) All of the above
8. Limit cycle is [ ]
a) Zero input limit cycle b) Overflow limit cycle
c) Both zero input and overflow limit cycle d) None of the above
21. The anti- symmetrical impulse response can be used to design ______________
22. For ___________ realization the current output y(n) is a function of only past and present inputs
14 The direct form FIR filter needs ................ between the adders to reduce the delay of the
adder tree and to achieve high throughput.
17. __________ filter is used specifically after upsampling process for removal of unwanted images
18. A __________is formed by an interconnection of the up-sampler, the down-sampler, and the
components of an LTI digital filter
20. The frequency response of the system with input h(n) and window length M is given by________
1. D
2. A
3. A
4. D
5. A
6. B
7. C
8. C
9. A
10. C
12. Non-recursive
13. FIR
15. -21 dB
19. Error
UNIT-I
1. Classification of systems
A discrete-time system can be thought of as a transformation or operator that maps an input sequence
{x[n]} to an output sequence
By placing various conditions on T(·) we can define different classes of systems. We give some properties
of systems. Basic System Properties are as given below
1) Systems with or without memory: A system is said to be memoryless if the out put for each value of the
independent variable at a given time n depends only on the input value at time n. For example system specified
by the relationship y[n] = cos(x[n]) + z is memoryless. A particularly simple memoryless system is the identity
system defined by y[n] = x[n]
In general we can write input-output relationship for memoryless system as
y[n] = g(x[n])
Not all systems are memoryless. A simple example of system with memory is a delay defined by
y[n] = x[n − 1]
A system with memory retains or stores information about input values at times other than the current input
value.
2) Invertibility: A system is said to be invertible if the input signal {x[n]} can be recovered from the output
signal {y[n]}. For this to be true two different input signals should produce two different outputs. If some
different input signal produce same output signal then by processing output we cannot say which input
produced the output.
Example of an invertible system is
then x[n] = y[n] − y[n − 1] Example if a non-invertible system is y[n]=0. That is the system produces an all zero
sequence for any input sequence. Since every input sequence gives all zero sequence, we can not find out which
input produced the output. The system which produces the sequence {x[n]} from sequence {y[n]} is called the
inverse system. In communication system, decoder is an inverse of the encoder.
3) Causality: A system is causal if the output at any time depends only on values of the input at the present
time and in the past.
is noncausal.
Note that operators of this type satisfy the linearity conditions, and c0,...,cn are real constants.
However, Equation 2 can easily be written as a linear constant coefficient recurrence equation without
difference operators. Conversely, linear constant coefficient recurrence equations can also be written in
the form of a difference equation, so the two types of equations are different representations of the same
relationship. Although we will still call them linear constant coefficient difference equations in this
course, we typically will not write them using difference operators. Instead, we will write them in the
simpler recurrence relation form
∑k=0Naky(n−k)=∑k=0Mbkx(n−k)
where x is the input to the system and y is the output. This can be rearranged to find y(n) as
y(n)=1a0(−∑k=1Naky(n−k)+∑k=0Mbkx(n−k))
The forms provided by equations will be used in the remainder of this course.
A similar concept for continuous time setting, differential equations, is discussed in the chapter on time
domain analysis of continuous time systems. There are many parallels between the discussion of linear
constant coefficient ordinary differential equations and linear constant coefficient differece equations.
If x[n] e j 2f nd
then
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
y[n] h[k ]x[n k ] h[k ]e j 2f d ( nk )
k k
e j 2f n h[k ]e j 2f k
d d
k
e j 2f d n
H (e jw )
Eigenfunction eigenvalue
Example:
Let x[ n ] A cos 2fn A
2
e
j 2fn e j 2fn
y[ n ] h[ k ] x[ n k ]
k
A
j 2f n k
j 2f n k
2 k
h[ k ]e h[ k ]e
k
A
2
H e
jw
e
j 2fn
H e jw e j 2fn
A special case of this problem exist when h[n] is real
H e jw H e jw
In this case
y[n] A H e jw cos2fn where angH e jw
In other words a sinusoidal input to a discrete time LTI system provides a sinusoidal output.
Discrete-time Fourier transform X e jw of a sequence x[n] is defined as:
X e jw
x[n]e
n
jwn
(1)
In general X e jw is a complex function of the real variable w and can be written as:
X e jw X re e jw jX im e jw
X e jw can alternatively be expressed in polar form as:
X e jw X e jw e j ( w) where,
( w) arg X e jw (2)
In many applications the Fourier transform is called the Fourier Spectrum and likewise X e jw and (w) are
referred to as the “magnitude spectrum” and “phase spectrum” respectively.
Note from eq.(2) that if we replace (w) with (w) + 2k , where k is an integer, X e jw remains unchanged
implying that the phase function cannot be uniquely specified for any Fourier Transform.
(6.1)
where is the input signal, is the output signal, and the constants
specifies a digital filtering operation, and the coefficient sets and fully characterize
the filter. In this example, we have .
When the coefficients are real numbers, as in the above example, the filter is said to be real. Otherwise, it
may be complex.
Notice that a filter of the form of Eq. (5.1) can use ``past'' output samples (such as ) in the
calculation of the ``present'' output . This use of past output samples is calledfeedback. Any filter
having one or more feedback paths ( ) is called recursive. (By the way, the minus signs for the
feedback in Eq. (5.1) will be explained when we get to transfer functions in §6.1.)
More specifically, the coefficients are called the feedforward coefficients and the coefficients are
called the feedback coefficients.
A filter is said to be recursive if and only if for some . Recursive filters are also called infinite-
When used for discrete-time physical modeling, the difference equation may be referred to as
an explicit finite difference scheme.6.2
Showing that a recursive filter is LTI (Chapter 4) is easy by considering its impulse-response
representation (discussed in §5.6). For example, the recursive filter
UNIT-II
1. Properties of DFT
As a special case of general Fourier transform, the discrete time transform shares all properties (and their
proofs) of the Fourier transform discussed above, except now some of these properties may take different
Time Shifting
Proof:
Time Reversal
Frequency Shifting
Differencing
Differencing is the discrete-time counterpart of differentiation.
Proof:
Differentiation in frequency
proof: Differentiating the definition of discrete Fourier transform with respect to , we get
Proof of (a):
Proof of (b):
Parseval's Relation
2. Computation of DFT using Over-Lap Add Method and Over-Lap Save Method
There are many DSP applications where a long signal must be filtered insegments. For instance,
high fidelity digital audio requires a data rate of about 5 Mbytes/min, while digital video requires
about 500 Mbytes/min. With data rates this high, it is common for computers to have insufficient
memory to simultaneously hold the entire signal to be processed. There are also systems that process
segment-by-segment because they operate in real time. For example, telephone signals cannot be
delayed by more than a few hundred milliseconds, limiting the amount of data that are available for
processing at any one instant. In still other applications, the processing may require that the signal be
segmented. An example is FFT convolution, the main topic of this chapter.
The overlap-add method is based on the fundamental technique in DSP:
(1) Decompose the signal into simple components,
(2) Process each of the components in some useful way, and
(3) Recombine the processed components into the final signal.
When an N sample signal is convolved with an M sample filter kernel, the output signal is N + M - 1
sample long. For instance, the input signal,
(a), is 300 samples (running from 0 to 299), the filter kernel,
(b), is 101 samples (running from 0 to 100), and the output signal, (i), is 400 samples.
When an N sample signal is filtered, it will be expanded by M - 1 point to the right. (This is
assuming that the filter kernel runs from index 0 to M. If negative indexes are used in the filter kernel,
the expansion will also be to the left). In (a), zeros have been added to the signal between sample 300
and 399 to illustrate where this expansion will occur. Don't be confused by the small values at the ends
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
of the output signal, (i). This is simply a result of the windowed-sinc filter kernel having small values
near its ends. All 400 samples in (i) are nonzero, even though some of them are too small to be seen in
the graph.
Below Figures show the decomposition used in the overlap-add method. The signal is broken into
segments, with each segment having 100 samples from the original signal. In addition, 100 zeros are
added to the right of each segment. In the next step, each segment is individually filtered by convolving it
with the filter kernel. This produces the output segments shown in figures. Since each input segment is
100 samples long, and the filter kernel is 101 samples long, each output segment will be 200 samples long.
The important point to understand is that the 100 zeros were added to each input segment to allow for the
expansion during the convolution.
Notice that the expansion results in the output segments overlapping each other. These overlapping output
segments are added to give the output signal, (i). For instance, samples 200 to 299 in (i) are found by
adding the corresponding samples in (g) and (h). The overlap-add method produces exactly the same
output signal as direct convolution. The disadvantage is a much greater program complexity to keep track
of the overlapping samples.
The FFT is a complicated algorithm, and its details are usually left to those that specialize in such things.
This section describes the general operation of the FFT, but skirts a key issue: the use of complex numbers.
In complex notation, the time and frequency domains each contain one signal made up of N complex
points. Each of these complex points is composed of two numbers, the real part and the imaginary part.
For example, when we talk about complex sample X[42], it refers to the combination of ReX[42]
and ImX[42]. In other words, each complex variable holds two numbers. When two complex variables are
multiplied, the four individual components must be combined to form the two components of the product
The following discussion on "How the FFT works" uses this jargon of complex notation. That is, the
singular terms: signal, point, sample, and value, refer to the combination of the real part and the imaginary
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
part.
The FFT operates by decomposing an N point time domain signal into N time domain signals each
composed of a single point. The second step is to calculate the N frequency spectra corresponding to
these N time domain signals. Lastly, the N spectra are synthesized into a single frequency spectrum.
Below figure shows an example of the time domain decomposition used in the FFT. In this example, a 16
point signal is decomposed through four
Separate stages. The first stage breaks the 16 point signal into two signals each consisting of 8 points. The
second stage decomposes the data into four signals of 4 points. This pattern continues until there
are N signals composed of a single point. An interlaced decomposition is used each time a signal is broken
in two, that is, the signal is separated into its even and odd numbered samples.
There are Log2N stages required in this decomposition, i.e., a 16 point signal (2 4) requires 4 stages, a 512
point signal (27) requires 7 stages, a 4096 point signal (212) requires 12 stages, etc. Remember this
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
value, Log2N.
5. Inverse FFT
If you need to compute inverse fast Fourier transforms (inverse FFTs) but you only have forward FFT
software (or forward FFT FPGA cores) available to you, below are four ways to solve your problem.
Preliminaries
To define what we're thinking about here, an N-point forward FFT and an N-point inverse FFT are
described by:
Forward FFT→X(m)=N−1∑n=0x(n)e−j2πnm/N(1)(1)Forward FFT→X(m)=∑n=0N−1x(n)e−j2πnm/NInv
erse FFT→x(n)=1NN−1∑m=0X(m)ej2πmn/NInverse FFT→x(n)=1N∑m=0N−1X(m)ej2πmn/N
=1NN−1∑m=0[Xreal(m)+jXimag(m)]ej2πmn/N(2)(2)=1N∑m=0N−1[Xreal(m)+jXimag(m)]ej2πmn/N
Inverse FFT Method# 1
The first method of computing inverse FFTs using the forward FFT was proposed as a "novel" technique
in 1988 [1]. That method is shown in Figure 1.
UNIT-III
1. Analog filter approximations -Butter worth
The classical method of analog filters design is Butterworth approximation. The Butterworth filters are also
known as maximally flat filters. Squared magnitude response of a Butterworth low-pass filter is defined as
follows
gain at DC is equal to 1;
has a maximum at
The first derivatives of (3.1) are equal to zero at . This is why Butterworth filters are
known as maximally flat filters.
219]. Thus, if denotes the impulse-response of an analog (continuous-time) filter, then the digital
(discrete-time) filter given by the impulse-invariant method will have impulse response ,
where denotes the sampling interval in seconds. Moreover, the order of the filter is preserved, and
To derive the impulse-invariant method, we begin with the analog transfer function
(9.1)
3. and perform a partial fraction expansion (PFE) down to first-order terms [452]:9.4
where is the th pole of the analog system, and is its residue [452]. Assume that the system is at
least marginally stable [452] so that there are no poles in the right-half plane ( ). Such a PFE is
always possible when is a strictly proper transfer function (more poles than zeros
[452]).9.5 Performing the inverse Laplace transform on the partial fraction expansion we obtain the impulse
response in terms of the system poles and residues:
Taking the z transform gives the digital filter transfer function designed by the impulse-invariant method:
We see that the -plane poles have mapped to the -plane poles
(9.2)
and the residues have remained unchanged. Clearly we must have , i.e., the analog
poles must lie within the bandwidth spanned by the digital sampling rate . Otherwise, the pole
since
The Chebyshev polynomial of degree n is denoted Tn(x), and is given by the explicit formula Tn(x) = cos(n arccos x) This
may look trigonometric at first glance (and there is in fact a close relation between the Chebyshev polynomials and the
discrete Fourier transform); however ( can be combined with trigonometric identities to yield explicit expressions for
Tn(x) ,
T0(x)=1
T1(x) = x
T2(x)=2x2 − 1
T3(x)=4x3 − 3x
Tn+1(x)=2xTn(x) − Tn−1(x) n ≥ 1.
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 filteralthough there are analog filters constructed
with switched capacitors that are discrete-time filters). It maps positions on the axis, , 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
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.
5. Spectral Transformations.
When designing a particular filter, it is common practice to first design a Low-pass filter, and then using
a spectral transform to convert that lowpass filter equation into the equation of a different type of filter.
This is done because many common values for butterworth, cheybyshev and elliptical low-pass filters are
already extensively tabulated, and designing a filter in this manner rarely requires more effort then
looking values up in a table, and applying the correct transformations. This page will enumerate some of
the common transformations for filter design.
It is important to note that spectral transformations are not exclusively used for analog filters. There are
digital variants of these transformations that can be applied directly to digital filters to transform them into a
different type. This page will only talk about the analog filter transforms, and the digital filter transforms will
be discussed elsewhere.
UNIT-IV
1. Characteristics of FIR Digital Filters
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, which is referred to as the phase delay. And
consequently, there is no phase distortion due to the time delay of frequencies relative to one another.
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)
(a) Rectangular
(c) Hanning
(d) Harming
(e) Blackman
(f) Kaiser
The main lobe width and first side lobe attenuation increase as we proceed down the window listed above.
An ideal lowpass filter with linear phase and cut off is characterized by
Since this is symmetric about , if we change and use one of the windows listed above the will get linear
phase FIR filter. Transition width and minimum stopped attenuation are listed in the Table 9.3.
Bartlett -25dB
Hanning -44dB
Hamming -53dB
Blackman -74dB
In designing FIR filter using Fourier series method the infinite duration impulse response
is truncated at n= ± (N-1/2).Direct truncation of the series will lead to fixed percentage
overshoots and undershoots before and after an approximated discontinuity in the
frequency response .
where X(z) is the z transform of the filter input signal , is the z transform of the output signal , and is the
filter transfer function.
A basic property of the z transform is that, over the unit circle , we find the spectrum[84].8.1To
show this, we set in the definition of the z transform, Eq. (6.1), to obtain
which may be recognized as the definition of the bilateral discrete time Fourier
transform(DTFT) when is normalized to 1 When is causal, this definition reduces to the usual
(unilateral) DTFT definition:
(8.1)
DTFT
UNIT-V
1. Decimation
Decimation can be regarded as the discrete-time counterpart of sampling. Whereas in sampling we start with a
continuous-time signal x(t) and convert it into a sequence of samples x[n], in decimation we start with a discrete-
time signal x[n] and convert it into another discrete-time signal y[n], which consists of sub-samples of x[n]. Thus,
the formal definition of M-fold decimation, or down-sampling, is defined by Equation. In decimation, the sampling
rate is reduced from Fs to Fs/M by discarding M – 1 samples for every M samples in the original sequence
∞
y[ n]= v[nM ]=∑ h[k] x[nM -k]
k=-∞
Interpolation is the exact opposite of decimation. It is an information preserving operation, in that all samples of
x[n] are present in the expanded signal y[n]. The mathematical definition of L-fold interpolation is defined by
Equation and the block diagram notation is depicted in below Figure Interpolation works by inserting (L–1) zero-
valued samples for each input sample. The sampling rate therefore increases from Fs to LFs. With reference to
Figure, the expansion process is followed by a unique digital low-pass filter called an anti-imaging filter. Although
the expansion process does not cause aliasing in the interpolated signal, it does however yield undesirable replicas
in the signal’s frequency spectrum.
y[n] =L h[k]w[n-k]
4. Limit cycles
When a stable IIR filter digital filter is excited by a finite sequence, that is constant, the output will
ideally decay to zero. However, the non-linearity due to finite precision arithmetic operations often
causes periodic oscillations to occur in the output. Such oscillations occur in the recursive systems are
called Zero input Limit Cycle Oscillation. Normally oscillations in the absence of output u (k) =0 by
equation given below is called Limit cycle oscillations
Discrete time signal: A discrete time signal is defined only at discrete instants of time.
The independent variable has discrete values only, which are uniformly spaced. A discrete time signal is
often derived from the continuous time signal by sampling it at a uniform rate.
By placing various conditions on T(·) we can define different classes of systems. We give some properties
of systems. Basic System Properties are as given below
1) Systems with or without memory: A system is said to be memoryless if the out put for each value of the
independent variable at a given time n depends only on the input value at time n. For example system specified
by the relationship y[n] = cos(x[n]) + z is memoryless. A particularly simple memoryless system is the identity
system defined by y[n] = x[n]
In general we can write input-output relationship for memoryless system as
y[n] = g(x[n])
Not all systems are memoryless. A simple example of system with memory is a delay defined by
y[n] = x[n − 1]
A system with memory retains or stores information about input values at times other than the current input
value.
2) Invertibility: A system is said to be invertible if the input signal {x[n]} can be recovered from the output
signal {y[n]}. For this to be true two different input signals should produce two different outputs. If some
different input signal produce same output signal then by processing output we cannot say which input
produced the output.
Example of an invertible system is
then x[n] = y[n] − y[n − 1] Example if a non-invertible system is y[n]=0. That is the system produces an all zero
sequence for any input sequence. Since every input sequence gives all zero sequence, we can not find out which
input produced the output. The system which produces the sequence {x[n]} from sequence {y[n]} is called the
inverse system. In communication system, decoder is an inverse of the encoder.
3) Causality: A system is causal if the output at any time depends only on values of the input at the present
time and in the past.
is noncausal.
4) Stability
There are several definitions for stability. Here we will consider bounded input bonded output(BIBO) stability. A
system is said to be BIBO stable if every bounded input produces a bounded output. We say that a signal {x[n]}
is bounded if |x[n]|<M < ∞ for all n
is stable as y[n] is sum of finite numbers and so it is bounded. The accumulator system defined by
is unstable. If we take {x[n]} = {u[n]}, the unit step then y[0] = 1, y[1] =2, y[2] = 3, are y[n] = n +1, n ≥ 0 so y[n]
grows without bound.
5) Time invariance
A system is said to be time invariant if the behavior and characteristics of the system do not change with time.
Thus a system is said to be time invariant if a time delay or time advance in the input signal leads to identical
delay or advance in the output signal. Mathematically if {y[n]} = T ({x[n]}), Then {y[n – no]} = T({x[n – no]}) for
any no
The two expression look different, but infact they are equal. Let us change the index of summation by l = k – no
in the first sum then we see that
Hence, {y[n]} = {y[n – no]} and the system is time-invariant. As a second example consider the system defined
by y[n] = nx[n]
if
while y[n – no] = (n – no)x[n – no] and so the system is not time-invariant. It is time varying. We can also see this
by giving a counter example. Suppose input is {x[n]} = {δ[n]} then output is all zero sequence. If the input is {δ[n
−1]} then output is {δ[n−1]} which is definitely not a shifted version version of all zero sequence.
6) Linearity
This is an important property of the system. We will see later that if we have system which is linear and time
invariant then it has a very compact representation. A linear system possesses the important property of
superposition: If an input consists of weighted sum of several signals, the and the output is also weighted sum
of the responses of the system to each of those input signals. Mathematically let {y1[n]} be the response of the
2) Prove that the impulse response of an LTI system is absolutely summable for stability of the system
Due to its convolution property, the z-transform is a powerful tool to analyze LTI systems
When the input is the eigenfunction of all LTI system, i.e., , the operation on this input by
the system can be found by multiplying the system's eigenvalue H(z) to the input.
Assuming the system is initially at rest with zero output , then its response to an
input is:
Due to the properties of the ROC, we know that If an LTI system is causal (with a right sided impulse response
function for ), then the ROC of its transfer function is the exterior of a circle
including infinity. In particular, when is rational, then the system is causal if and only if its ROC is the
exterior of a circle outside the out-most pole, and the order of numerator is no greater than the order of the
denominator.
Note the requirement for the orders of the numerator and denominator guarantees the existence of even
when .
Stable LTI systems
In other words, if the impulse response function of an LTI system is absolutely integrable, then the
system is stable. We can show that this condition is also necessary, i.e., all stable LTI systems' impulse
response functions are absolutely integrable. Now we have:
An LTI system is stable if and only if its impulse response is absolutely summable, i.e., the frequency response
function exits, i.e. the ROC of its transfer function includes the unit circle
2) Explain the principle of operation of analog to digital conversion with a neat diagram
Digital signal processing (DSP) is the use of digital processing, such as by computers, to perform a wide variety
of signal processing operations. The signals processed in this manner are a sequence of numbers that
represent samples of a continuous variable in a domain such as time, space, or frequency. Digital signal
processing and analog signal processing are subfields of signal processing. DSP applications
include audio and speech signal processing, sonar, radar and other sensor array processing, spectral
estimation, statistical signal processing, digital image processing, signal processing
for telecommunications, control of systems, biomedical engineering, seismic data processing, among others.
Digital signal processing can involve linear or nonlinear operations. Nonlinear signal processing is closely
related to nonlinear system identification and can be implemented in the time, frequency, and spatial-temporal
domains.
Signal sampling
The increasing use of computers has resulted in the increased use of, and need for, digital signal processing. To
digitally analyze and manipulate an analog signal, it must be digitized with an analog-to-digital converter.
Sampling is usually carried out in two stages, discretization and quantization.
Discretization means that the signal is divided into equal intervals of time, and each interval is represented by a
The main advantage of digital signals over analog signals is that the precise signal level of the digital
signal is not vital. This means that digital signals are fairly immune to the imperfections of real
electronic systems which tend to spoil analog signals. As a result, digital CD's are much more robust
than analog LP's.
Codes are often used in the transmission of information. These codes can be used either as a means of
keeping the information secret or as a means of breaking the information into pieces that are
manageable by the technology used to transmit the code, e.g. The letters and numbers to be sent by a
Morse code are coded into dots and dashes.
Digital signals can convey information with greater noise immunity, because each information
component (byte etc) is determined by the presence or absence of a data bit (0 or one). Analog signals
vary continuously and their value is affected by all levels of noise.
Digital signals can be processed by digital circuit components, which are cheap and easily produced in
many components on a single chip. Again, noise propagation through the demodulation system is
minimized with digital techniques.
Digital signals do not get corrupted by noise etc. You are sending a series of numbers that represent the
signal of interest (i.e. audio, video etc.)
Digital signals typically use less bandwidth. This is just another way to say you can cram more
information (audio, video) into the same space.
Digital can be encrypted so that only the intended receiver can decode it (like pay per view video,
secure telephone etc.)
Enables transmission of signals over a long distance.
Transmission is at a higher rate and with a wider broadband width.
It is more secure.
It is also easier to translate human audio and video signals and other messages into machine language.
There is minimal electromagnetic interference in digital technology.
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
It enables multi-directional transmission simultaneously.
4) Discuss the steps involved in calculating convolution sum by taking simple practical example
The steps involved in calculating sum are
1) Folding
2) Shifting
3) Multiplication
4) Summation
.Let's say a chef decides to offer private cooking lessons. A starting pupil receives a 1-hour lesson to explore
interests and background. Next day, and all the days following, the private lessons are 3 hours long. If the
function h is the hours per day for one pupil, the graph would look like:
The λ = 0 at the bottom of the Σ is our start value for λ. We will have d multiplies to add: one for λ = 0, another
for λ = 1, and we keep going until λ = d. See the 2 above the Σ in the next equation? The number 2 tells us the
end value for λ. To clarify, let's work out the number of hours of instruction for d = 2.
1) _______ is a System which carries out mathematical operations on a sequences of samples related to a
signal.
2) _______ is defined as any physical quantity that varies with time, space, or any other independent
variable
3) A discrete time signal having a set of discrete values is ____________
4) The Process of converting a continuous-valued signal into discrete-valued signal is called
__________________.
5) Any signal that can be uniquely described by an explicit mathematical expression, a table of data, or a
well-defined rule is ______________
6) A system is said to be _________ if the output of the system at any time n depends only on present and
past inputs
7) ____________ is a constant function that describes the magnitude and phase shift of a filter over a range
of frequencies
8) ___________ is the property that states the Z-transforms of a sum of signals is the sum of individual Z-
transforms.
9) __________ of time signals corresponds to the multiplication of Z-transform.
10) If the ROC extends outward from the outermost pole, then the system is _________.
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
1) DSP
2) Signal
3) Digital Signal
4) Quantization
5) Deterministic
6) Causal
7) Frequency Response
8) Linearity
9) Convolution
10) Causal
MULTIPLE CHOICE QUESTIONS
UNIT 2
2 MARKS QUESTIONS WITH ANSWERS
Periodicity
Linearity and symmetry Multiplication of two DFTs
Circular convolution
Time reversal
Circular time shift and frequency shift Complex conjugate
Circular correlation
Let x1(n) and x2(n) are finite duration sequences both of length N with DFTs X1 (k) and X2 (k). If
X3(k) = X1(k) X2(k) then the sequence x3(n) can be obtained by circular.
If the data sequence x (n) is of long duration it is very difficult to obtain the output sequence y(n) due
to limited memory of a digital computer. Therefore, the data sequence is divided up into smaller sections.
4. What is FFT?
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of the symmetry and
periodicity properties of twiddle factor to effectively reduce the DFT computation time. It is based on the
fundamental principle of decomposing the computation of DFT of a sequence of length N into successively
smaller DFTs.
5. How many multiplications and additions are required to compute N point DFT using radix- 2FFT?
The number of multiplications and additions required to compute N point DFT using radix-2 FFT are N log2 N
and N/2 log2 N respectively
It can be used to find the response of a linear It cannot be used to find the response of a filter
filter
Zero padding is not necessary to find the Zero padding is necessary to find the response
response of a linear filter.
5. What are the differences and similarities between DIF and DITalgorithms?
Differences:
1) The input is bit reversed while the output is in natural order for DIT, whereas for DIF the output is
bit reversed while the input is in naturalorder.
2) The DIF butterfly is slightly different from the DIT butterfly, the difference being that the complex
multiplication takes place after the add-subtract operation inDIF.
Similarities:
Both algorithms require same number of operations to compute the DFT. Both algorithms can be done in
place and both need to perform bit reversal at some place during the computation.
Time Shifting
Proof:
Time Reversal
Frequency Shifting
Differencing
Differencing is the discrete-time counterpart of differentiation.
Proof:
Differentiation in frequency
proof: Differentiating the definition of discrete Fourier transform with respect to , we get
Convolution Theorems
The convolution theorem states that convolution in time domain corresponds to multiplication in frequency
domain and vice versa:
Proof of (a):
Proof of (b):
Parseval's Relation
DFT.
This algorithm is very similar in concept to the Decimation in Frequency (DIF) Algorithm discussed earlier,
so the presentation will be a little less detailed. Have a look at the section describing the DIF algorithm first.
We defined the FFT as:
If N is even, the above sum can be split into 'even' (n=2n') and 'odd' (n=2n'+1) halves, where n'=0..N/2-1,
and re-arranged as follows:
This process of splitting the 'time domain' sequence into even an odd samples is what gives the algorithm its
name, 'Decimation In Time'. As with the DIF algorithm, we have succeeded in expressing an N point
transform as 2 (N/2) point sub-transforms. The principal difference here is that the order we do things has
changed. In the DIF algorithm the time domain data was 'twiddled' before the two sub-transforms were
performed. Here the two sub-transforms are performed first. The final result is obtained by 'twiddling' the
resulting frequency domain data. There is a slight problem here, because the two sub-transforms only give
values for k=0..N/2-1. We also need values for k=N/2..N-1. But from the periodicity of the DFT we know:
and..
where:
This all we need to produce a simple recursive DIT FFT routine for any N which is a regular power of 2
(N=2p).
When an N sample signal is filtered, it will be expanded by M - 1 point to the right. (This is assuming
that the filter kernel runs from index 0 to M. If negative indexes are used in the filter kernel, the expansion
will also be to the left). In (a), zeros have been added to the signal between sample 300 and 399 to illustrate
where this expansion will occur. Don't be confused by the small values at the ends of the output signal, (i).
This is simply a result of the windowed-sinc filter kernel having small values near its ends. All 400 samples
in (i) are nonzero, even though some of them are too small to be seen in the graph.
Below Figures show the decomposition used in the overlap-add method. The signal is broken into
segments, with each segment having 100 samples from the original signal. In addition, 100 zeros are added
to the right of each segment. In the next step, each segment is individually filtered by convolving it with the
filter kernel. This produces the output segments shown in figures. Since each input segment is 100 samples
long, and the filter kernel is 101 samples long, each output segment will be 200 samples long. The important
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
point to understand is that the 100 zeros were added to each input segment to allow for the expansion during
the convolution.
Notice that the expansion results in the output segments overlapping each other. These overlapping output
segments are added to give the output signal, (i). For instance, samples 200 to 299 in (i) are found by adding
the corresponding samples in (g) and (h). The overlap-add method produces exactly the same output signal
as direct convolution. The disadvantage is a much greater program complexity to keep track of the
overlapping samples.
4) Explain FFT
The FFT is a complicated algorithm, and its details are usually left to those that specialize in such things.
This section describes the general operation of the FFT, but skirts a key issue: the use of complex numbers.
In complex notation, the time and frequency domains each contain one signal made up of N complex points.
Each of these complex points is composed of two numbers, the real part and the imaginary part. For
example, when we talk about complex sample X[42], it refers to the combination of ReX[42] and ImX[42]. In
other words, each complex variable holds two numbers. When two complex variables are multiplied, the
four individual components must be combined to form the two components of the product
The following discussion on "How the FFT works" uses this jargon of complex notation. That is, the
singular terms: signal, point, sample, and value, refer to the combination of the real part and the imaginary
part.
The FFT operates by decomposing an N point time domain signal into N time domain signals each
composed of a single point. The second step is to calculate the N frequency spectra corresponding to
these N time domain signals. Lastly, the N spectra are synthesized into a single frequency spectrum.
Below figure shows an example of the time domain decomposition used in the FFT. In this example, a 16
point signal is decomposed through four
Separate stages. The first stage breaks the 16 point signal into two signals each consisting of 8 points. The
second stage decomposes the data into four signals of 4 points. This pattern continues until there
are N signals composed of a single point. An interlaced decomposition is used each time a signal is broken in
two, that is, the signal is separated into its even and odd numbered samples.
There are Log2N stages required in this decomposition, i.e., a 16 point signal (2 4) requires 4 stages, a 512
point signal (27) requires 7 stages, a 4096 point signal (212) requires 12 stages, etc. Remember this
value, Log2N.
1) The important tools used in frequency analysis of signals are _____________and __________ ___
2) __________ can be used to find the response of a linear filter.
3) _________ is not necessary to find the response of a linear filter in linear convolution.
4) The linear convolution of 2 finite duration sequences x(n) and h(n) of lengths L samples and M samples
will result in a output sequence of duration ___________ samples
5) By using with Zero padding, DFT can be used in ____________
2𝜋
1. Given that , W = 𝑒 −𝑖( 𝑁 ) ,where N = 3. Then F = 𝑊 𝑁 can be computed as F =
(A) 0 (B) 1
(C)- 1 (D) 5
Solution
(A) 0 (B) 1
(C) -1 (D) e
Solution
4 − 6𝑖
3.Given that N=2, {f} = { }. The valuesfor vector shown in
−2 + 4𝑖
Can be computed as
−2 −2
(A) { } (B){ }
−6 6
2 2
(C) { } (D) { }
−6 6
Solution
can be computed as
−2
(A) { }
−10
−1
(B) { }
−10
−2
(C) { }
−5
−1
(D) { }
−5
Solution
5. If the forcing function F(t) is given as F(t) = ∑7𝑛=0 10 sin(2𝜋𝑛𝑡).Then, to avoid aliasing phenomenon, the
minimum number of sample data points Nminshould be
(A)8 (B)16
(C)24 (D) 32
Solution
6. Based on the figure below, aliasing phenomena will not occur because there were
Solution
7. Using the definition E= 𝑒 −𝑖2𝜋/𝑁 , and the Euler identity𝑒 ±𝑖𝜃 = 𝑐𝑜𝑠𝜃 ± 𝑠𝑖𝑛𝜃, the value E N/6 can be computed
as
Solution
8. Using the definition E = 𝑒 −𝑖2𝜋/𝑁 , and the Euler identity 𝑒 ±𝑖𝜃 = 𝑐𝑜𝑠𝜃 ± 𝑠𝑖𝑛𝜃, the value E 6N can be computed
as
(A) i+1
(B) i−1
(C) 1
(D) -1
Solution
Solution
(10) For N=24=16, levelL=2and referring to the figure shown below, the only terms of vector f2(−)which only
need to compute are:
Solution
UNIT III
2 MARKS QUESTIONS WITH ANSWERS
Disadvantages:
1. For the same filter specifications the order of FIR filter design can be as high as 5 to 10 times that
in an IIRdesign.
2. Large storage requirement isrequirement
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
3. Powerful computational facilities required for theimplementation.
2. What is the mapping procedure between S-plane & Z-plane in the method of mapping differentials? What
are its characteristics?
The mapping procedure between S-plane & Z-plane in the method of mapping of differentials is
given by
H(Z) =H(S)|S=(1-Z-1)/T
The effect of the non-linear compression at high frequencies can be compensated. When the desired
magnitude response is piece-wise constant over frequency, this compression can be compensated by
introducing a suitable pre-scaling, or pre-warping the critical frequencies by using the formula.
Disadvantage:
· The mapping is highly non-linear producing frequency, compression at highfrequencies.
· Neither the impulse response nor the phase response of the analog filter is preserved in a digital
filter obtained by bilineartransformation.
The classical method of analog filters design is Butterworth approximation. The Butterworth filters are also
known as maximally flat filters. Squared magnitude response of a Butterworth low-pass filter is defined as
follows
gain at DC is equal to 1;
has a maximum at
The first derivatives of (3.1) are equal to zero at . This is why Butterworth filters are
known as maximally flat filters.
Function has poles and doesn't have any finite zeros. It is easy to see that if is a pole of
(3.2), then is also a pole of (3.2). In order to find the poles of transfer function that satisfy (3.2), we
have to select one pole from each pair of the poles of expression (3.2). As it was mentioned
before, the poles of a valid filter have to have negative real parts.
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
The poles of (3.2) can be found as roots of equation
Observing that , where stands for the odd number, roots of can be obtained as
solutions to the equation
All poles lie on a circle of radius in the complex s-plane. Since the difference has the
same value for all roots, it can be concluded that the poles are equally spaced on the circumference.
The poles of Butterworth filter lie on left half of the s-plane, and they can be given as follows
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
Advantages
In reality, the signal bandwidth of the digital sequence is much lower than the analog sequence. Signal
processing speed is one of the key factors of calculating the total device performance. Actually the
speed operation totally depends on the number of the arithmetic operation in the processor.
Finite word-length effect, which results quantizing noise and round-off noise, is another major
drawback during computation.
It needs much longer time to design and develop the digital sequences though it can be used on other
tasks or applications once developed. Ordinarily good support of computer aided design can convert
them into a enjoyable tasks.
3) Discuss magnitude characteristics of an analog Butterworth filter and give its pole locations
Note that the higher the Butterworth filter order, the higher the number of cascaded stages there are within
the filter design, and the closer the filter becomes to the ideal “brick wall” response.
In practice however, Butterworth’s ideal frequency response is unattainable as it produces excessive passband
ripple.
Where the generalised equation representing a “nth” Order Butterworth filter, the frequency response is given
as:
Where: n represents the filter order, Omega ω is equal to 2πƒ and Epsilon ε is the maximum pass band gain,
(Amax). If Amax is defined at a frequency equal to the cut-off -3dB corner point (ƒc), ε will then be equal to one
and therefore ε2 will also be one. However, if you now wish to define Amax at a different voltage gain value, for
example 1dB, or 1.1220 (1dB = 20logAmax) then the new value of epsilon, ε is found by:
Where:
H0 = the Maximum Pass band Gain, Amax.
H1 = the Minimum Pass band Gain.
The Frequency Response of a filter can be defined mathematically by its Transfer Function with the standard
Voltage Transfer Function H(jω) written as:
Where:
Vout = the output signal voltage.
Vin = the input signal voltage.
j = to the square root of -1 (√-1)
ω = the radian frequency (2πƒ)
Note: ( jω ) can also be written as ( s ) to denote the S-domain. and the resultant transfer function for a second-
order low pass filter is given as:
To help in the design of his low pass filters, Butterworth produced standard tables of normalized second-order
low pass polynomials given the values of coefficient that correspond to a cut-off corner frequency of 1
radian/sec.
1 (1+s)
2 (1+1.414s+s2)
4 (1+0.765s+s2)(1+1.848s+s2)
5 (1+s)(1+0.618s+s2)(1+1.618s+s2)
6 (1+0.518s+s2)(1+1.414s+s2)(1+1.932s+s2)
7 (1+s)(1+0.445s+s2)(1+1.247s+s2)(1+1.802s+s2)
8 (1+0.390s+s2)(1+1.111s+s2)(1+1.663s+s2)(1+1.962s+s2)
9 (1+s)(1+0.347s+s2)(1+s+s2)(1+1.532s+s2)(1+1.879s+s2)
10 (1+0.313s+s2)(1+0.908s+s2)(1+1.414s+s2)(1+1.782s+s2)(1+1.975s+s2)
Find the order of an active low pass Butterworth filter whose specifications are given as: Amax = 0.5dB at a pass
band frequency (ωp) of 200 radian/sec (31.8Hz), and Amin = 20dBat a stop band frequency (ωs) of 800
radian/sec. Also design a suitable Butterworth filter circuit to match these requirements.
Firstly, the maximum pass band gain Amax = 0.5dB which is equal to a gain of 1.0593(0.5dB = 20log A) at a
frequency (ωp) of 200 rads/s, so the value of epsilon ε is found by:
Secondly, the minimum stop band gain Amin = 20dB which is equal to a gain of 10 (20dB = 20log A) at a stop
band frequency (ωs) of 800 rads/s or 127.3Hz.
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
Substituting the values into the general equation for a Butterworth filters frequency response gives us the
following:
Since n must always be an integer ( whole number ) then the next highest value to 2.42 is n = 3, therefore a “a
third-order filter is required” and to produce a third-order Butterworth filter, a second-order filter stage
cascaded together with a first-order filter stage is required.
From the normalised low pass Butterworth Polynomials table above, the coefficient for a third-order filter is
given as (1+s)(1+s+s2) and this gives us a gain of 3-A = 1, or A = 2. As A = 1 + (Rf/R1), choosing a value for both
the feedback resistor Rf and resistor R1 gives us values of 1kΩ and 1kΩ respectively, ( 1kΩ/1kΩ + 1 = 2 ).
We know that the cut-off corner frequency, the -3dB point (ωo) can be found using the formula 1/CR, but we
need to find ωo from the pass band frequency ωp then,
So, the cut-off corner frequency is given as 284 rads/s or 45.2Hz, (284/2π) and using the familiar
formula 1/CR we can find the values of the resistors and capacitors for our third-order circuit.
Note that the nearest preferred value to 0.352uF would be 0.36uF, or 360nF.
and finally our circuit of the third-order low pass Butterworth Filter with a cut-off corner frequency of 284
rads/s or 45.2Hz, a maximum pass band gain of 0.5dB and a minimum stop band gain of 20dB is constructed as
follows.
4.
5.
a. Recursive
b. Non Recursive
c. Reversive
d. Non Reversive
ans-a
ans-d
a. Direct Method
b. In direct method
c. Recursive method
ans-a
4. In the design a IIR Digital filter for the conversion of analog filter in to Digital domain the desirable property
is
a. The axis in the s - plane should map outside the unit circle in the z - Plane
b. The Left Half Plane(LHP) of the s - plane should map in to the unit circle in the Z -
Plane
c. The Left Half Plane(LHP) of the s-plane should map outside the unit circle in the z-
Plane
d. The Right Half Plane(RHP) of the s-plane should map in to the unit circle in the Z -
Plane
Ans-d
9.The filter that may not be realized by approximation of derivatives techniques are
1) Band pass filters
2) High pass filters
3) Low pass filters
4) Band reject filters
10. Which among the following represent/s the characteristic/s of an ideal filter?
a. Constant gain in pass band
b. Zero gain in stop band
c. Linear Phase Response
d. All of the above
P.USHA, Assistant. Professor,dept of ECE,KGRCET
DIGITAL SIGNAL PROCESSING(EE721PE)
ANSWER: (d) All of the above
1. Filters exhibit their dependency upon the system design for the stability purpose
3. The transformation technique in which there is one to one mapping from s-domain to z-domain is
6. LTI IIR systems are stable if all the poles of the system function lie the unit circle
9. IIR filter can have linear phase , if and only if ,it is both and
10. The advantage of direct form-II over direct form-I realization is needs
ANSWERS
1. IIR
5. ellipse
6. inside
8. infinite
UNIT IV
1. 2 MARKS QUESTIONS WITH ANSWERS
2. How phase distortion and delay distortion are introduced?
The phase distortion is introduced when the phase characteristics of a filter is nonlinear within the
desired frequency band. The delay distortion is introduced when the delay is not constant within the desired
frequency band.
6. Define necessary and sufficient condition for the linear phase characteristic of a FIR filter?
The phase function should be a linear function of w, which in turn requires constant group delay
and phase delay.
1. List the well-known design technique for linear phase FIR filter design?
Fourier series method and window method Frequency
sampling method
Optimal filter design method
Because of the non-linear mapping: the amplitude response of digital IIR filter is expand
at lower frequencies and compressed at higher frequencies in comparison to the
analog filter.
1. The impulse response of this filter is restricted The impulse response extends to infinite
to finite number of samples duration
2. FIR Filters have linear phase IIR filter don’t have linear phase
3. Always stable Not always stable
4. Greater flexibility Less flexibility
In linear phase filter ( ) α, the linear phase filter does not alter the shape of the
original signal. If phase response of the filter is nonlinear the output signal is
distorted one. In many cases Linear Phase filter is required throughout the pass
band of the filter to preserve the shape of the given signal within the pass band.
An IIR filter cannot produce a linear phase. The FIR filter can give linear phase,
when the impulse response of the filter is symmetric about its midpoint.
(a) Rectangular
(c) Hanning
(d) Harming
(e) Blackman
(f) Kaiser
where is modified zero-order Bessel function of the first kind given by
The main lobe width and first side lobe attenuation increase as we proceed down the window listed above.
An ideal lowpass filter with linear phase and cut off is characterized by
Since this is symmetric about , if we change and use one of the windows listed above the will get linear
phase FIR filter. Transition width and minimum stopped attenuation are listed in the Table 9.3.
Bartlett -25dB
Hanning -44dB
Hamming -53dB
Blackman -74dB
For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter. Approximations can be achieved with infinite impulse response (IIR) designs, which are more
computationally efficient.
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)
5.Why are FIR filters generally preferred over IIR filters in multi rate (decimating and interpolating)
systems?
Because only a fraction of the calculations that would be required to implement a decimating or interpolating
FIR in a literal way actually needs to be done.
Since FIR filters do not use feedback, only those outputs which are actually going to be used have to be
calculated. Therefore, in the case of decimating FIRs (in which only 1 of N outputs will be used), the other
N-1 outputs don’t have to be calculated. Similarly, for interpolating filters (in which zeroes are inserted
between the input samples to raise the sampling rate) you don’t actually have to multiply the inserted zeroes
with their corresponding FIR coefficients and sum the result; you just omit the multiplication-additions that
are associated with the zeroes (because they don’t change the result anyway.)
In contrast, since IIR filters use feedback, every input must be used, and every input must be calculated
because all inputs and outputs contribute to the feedback in the filter.
a. A & B
b. C & D
c. A & D
d. B & C
ANSWER:(a) A & B
5. In FIR filter design, which among the following parameters is/are separately controlled by using Kaiser
window?
a. Hamming window
b. Hanning window
c. Barlett window
d. Blackman window
a. linearly
b. elliptically
c. hyperbolically
d. parabolically
ANSWER: linearly
8. In FIR filters, which among the following parameters remains unaffected by the quantization effect?
a. Magnitude Response
b. Phase Characteristics
c. Both a and b
d. None of the above
9. In the frequency response characteristics of FIR filter, the number of bits per coefficient should be
_________in order to maintain the same error.
a. Increased
b. Constant
c. Decreased
d. None of the above
ANSWER: Increased
10. A filter is said to be linear phase filter if the phase delay and group delay are _______
a. High
b. Moderate
c. Low
d. Constant
ANSWER: Constant
Answers
1. Type-I
2. Origin
3. Linear
4. Transition bandwidth
5. N=M-1
6. FIR
m N
7. ∑bk z-k/1+∑ak z-k
K=0 k=1
8. 4π/M
9. Width
10. 12π/M
UNIT V
1.Give the different quantization errors occur due to finite word length
registers in digital filters?
2. Mention the different quantization methods available for Finite Word Length Effects?
1. Truncation
2. Rounding
3. State truncation?
Truncation is a process of discarding all bits less significant than LSB that is retained
4. Define Rounding?
Rounding a number to b bits is accomplished by choosing a rounded result as the b
bit number closest number being unrounded.
When a stable IIR filter digital filter is excited by a finite sequence, that is constant, the
output will ideally decay to zero. However, the non-linearity due to finite precision
arithmetic operations often causes periodic oscillations to occur in the output. Such
oscillations occur in the recursive systems are called Zero input Limit Cycle Oscillation.
Normally oscillations in the absence of output u (k) =0 by equation given below is called
Limit cycle oscillations
4. The error in the filter output that results from rounding or truncating calculations within the filter
is called
5. To change the sampling rate for better efficiency in two or multiple stages, The decimation and
interpolation factors must be _________unity.
a. Less than
b. Equal to
c. Greater than
d. None of the above
6. How is/are the roundoff errors reduced in the digital FIR filter?
7. Consider the assertions (steps) given below. Which among the following is a correct sequence of
designing steps for the sampling rate converters?
a. A, B, C, D
b. C, A, D, B
c. D, A, B, C
d. B, D, A, C
ANSWER: B, D, A, C
9. Which is/are the correct way/s for the result quantization of an arithmetic operation?
a. Result Truncation
b. Result Rounding
c. Both a and b
d. None of the above
ANSWER: b
Answers
1. Limit cycle
2. Input quantization error
3. Overflow
4. Interpolation
5. Dead band
6. Anti-aliasing
7. Addition
8. Product quantization error
9. Truncation
10. decrease
Part – 2
S.NO TOPICS
1 Attendance Register/Teacher Log Book
2 Time Table
3 Academic calendar
8 Assignment Evaluation-marks/Grades
VISION
MISSION
After 3-5 years from the year of graduation, our graduates will,
PEO 1: To inculcate the adaptability skills into the students for design and use of
analog and digital circuits and communications and any other allied fields of
Electronics.
PEO 3: To develop professional skills in students that prepares them for immediate
employment and for lifelong learning in advanced areas of Electronics and
communications and related fields.
PEO 4: To equip with skills for solving complex real-world problems related to VLSI,
Embedded Systems, Signal/Image processing, Communications, and Wave
theory.
PEO 5: Graduates will make valid judgment, synthesize information from a range of
sources and communicate them in sound ways in order to find an
economically viable solution.
PEO 6: To develop overall personality and character with team spirit, professionalism,
integrity, and moral values with the support of humanities, social sciences and
physical educational courses.
PO2: Problem analysis: Identify, formulate, review research literature, and analyze complex
engineering problems reaching substantiated conclusions using first principles of
mathematics, natural sciences, and engineering sciences.
PO5: Modern tool usage: Create, select, and apply appropriate techniques, resources, and
modern engineering and IT tools including prediction and modeling to complex
engineering activities with an understanding of the limitations.
PO6: The engineer and society: Apply reasoning informed by the contextual knowledge to
assess societal, health, safety, legal and cultural issues and the consequent responsibilities
relevant to the professional engineering practice.
PO7: Environment and sustainability: Understand the impact of the professional engineering
solutions in societal and environmental contexts, and demonstrate the knowledge of, and
need for sustainable development.
PO8: Ethics: Apply ethical principles and commit to professional ethics and responsibilities and
norms of the engineering practice.
PO9: Individual and team work: Function effectively as an individual, and as a member or
leader in diverse teams, and in multidisciplinary settings.
PO11: Project management and finance: Demonstrate knowledge and understanding of the
engineering and management principles and apply these to one’s own work, as a member
and leader in a team, to manage projects and in multidisciplinary environments.
PO12: Life-long learning: Recognize the need for, and have the preparation and ability to engage
in independent and life-long learning in the broadest context of technological change.
PSO 1: Problem Solving Skills – Graduate will be able to apply latest electronics
techniques and communications principles for designing of communications
systems.
PSO 2: Professional Skills – Graduate will be able to develop efficient and effective
Communications systems using modern Electronics and Communications
engineering techniques.
PSO 4: The Engineer and Society– Ability to apply the acquired knowledge for the
advancement of society and self.
UNIT -I
Introduction: Introduction to Digital Signal Processing: Discrete Tim signals &
Sequences, Linear Shift Invariant Systems, Stability, and Causality, Linear Constant
Coefficient Difference Equations, Frequency Domain Representation of Discrete Time
Signals and Systems
Realization of Digital Filters: Applications of Z — Transforms, Solution Difference
Equations of Digital Filters, System Function, Stability Criterion frequency Response of
Stable Systems, Realization of Digital Filters — Direct, Canonic, Cascade and Parallel
Forms.
UNIT -II
Discrete Fourier series: DFS Representation of Periodic Sequence properties of Discrete
Fourier Series, Discrete Fourier Transforms: Properties of DFT, Linear Convolution of
Sequences using DFT, Computation of D Over-Lap Add Method, Over-Lap Save Method,
Relation between DTFT, DFS, DFT and Z-Transform.
Fast Fourier Transforms: Fast Fourier Transforms (FFT) – Radix-2 Decimation-in-Time and
Decimation-in-Frequency FFT Algorithms, Inverse FFT, and FFT with General Radix-N.
UNIT- III
IIR Digital Filters: Analog filter approximations – Butter worth and Chebyshev, Design of IIR
Digital Filters from Analog Filters, Step and Impulse Invariant Techniques, Bilinear
Transformation Method, Spectral Transformations.
UNIT-IV
FIR Digital Filters: Characteristics of FIR Digital Filters, Frequency Response, Design of FIR
Filters: Fourier Method, Digital Filters using Window Techniques, Frequency Sampling
Technique, Comparison of IIR & FIR filters
UNIT-V
Multi rate Digital Signal Processing: Introduction, Down Sampling Decimation, Up sampling,
Interpolation, Sampling Rate Conversion. Finite Word Length Effects: Limit cycles, Overflow
Oscillations, Round-off Noise in IIR Digital Filters, Computational Output Round Off Noise,
Methods to Prevent Overflow, Trade Off Between Round Off and Overflow Noise, Dead
Band Effects.
TEXT BOOKS
Digital Signal Processing, Principles, Algorithms, and Applications John G. Proakis,
Dimitris G. Manolakis, Pearson Education / PHI, 2007.
Discrete Time Signal Processing — A. V. Oppenheim and R.W Schaffer, PHI,
2009 Fundamentals of Digital Signal Processing — Loney Ludeman, John Wiley, 2009
REFERENCE BOOKS
Digital Signal Processing — Fundamentals and Applications — Li Tan, Elsevier, 2008
Given a general aperiodic signal of finite duration, that is; for some integer N,
. From this aperiodic signal we can construct a periodic signal for which
is one period. As we chose period N to be larger than the duration of , is identical
to . As the period , for any finite value of n.
Since over a period that includes the interval , it is convenient to choose the
interval of summation to be this period, so that can be replaced by in the summation.
Therefore,
time ,for signal the discrete time fourier transform is periodic in with period
Consider a periodic sequence x[n] with period N and with fourier series representation
Then discrete time Fourier Transform of a periodic signal x[n] with period N can be written as :
2.WAVELET TRANSFORM
A wavelet is a wave-like oscillation with an amplitude that begins at zero, increases, and then
decreases back to zero. It can typically be visualized as a "brief oscillation" like one recorded by
a seismograph or heart monitor. Generally, wavelets are intentionally crafted to have specific
properties that make them useful for signal processing. Using a "reverse, shift, multiply and
integrate" technique called convolution, wavelets can be combined with known portions of a
damaged signal to extract information from the unknown portions.
All wavelet transforms may be considered forms of time-frequency representation for continuous-
time (analog) signals and so are related to harmonic analysis. Almost all practically useful discrete
wavelet transforms use discrete-time filterbanks. These filter banks are called the wavelet and
scaling coefficients in wavelets nomenclature. These filterbanks may contain either finite impulse
response (FIR) or infinite impulse response (IIR) filters. The wavelets forming a continuous
wavelet transform (CWT) are subject to the uncertainty principle of Fourier analysis respective
sampling theory: Given a signal with some event in it, one cannot assign simultaneously an exact
time and frequency response scale to that event. The product of the uncertainties of time and
frequency response scale has a lower bound. Thus, in the scaleogram of a continuous wavelet
transform of this signal, such an event marks an entire region in the time-scale plane, instead of
just one point. Also, discrete wavelet bases may be considered in the context of other forms of the
uncertainty principle.
Wavelet transforms are broadly divided into three classes: continuous, discrete and
multiresolution-based.
Continuous wavelet transforms (continuous shift and scale parameters)
In continuous wavelet transforms, a given signal of finite energy is projected on a continuous
family of frequency bands (or similar subspaces of the Lp function space L2(R)). For instance, the
signal may be represented on every frequency band of the form [f, 2f] for all positive frequencies
f > 0. Then, the original signal can be reconstructed by a suitable integration over all the resulting
frequency components.
The frequency bands or subspaces (sub-bands) are scaled versions of a subspace at scale 1. This
subspace in turn is in most situations generated by the shifts of one generating function ψ in
L2(R), the mother wavelet.
Discrete wavelet transforms (discrete shift and scale parameters)
It is computationally impossible to analyze a signal using all wavelet coefficients, so one may
wonder if it is sufficient to pick a discrete subset of the upper halfplane to be able to reconstruct
a signal from the corresponding wavelet coefficients. One such system is the affine system for
some real parameters a > 1, b > 0. The corresponding discrete subset of the halfplane consists of
all the points (am, namb) with m, n in Z.
Fig- D4 wavelet
In any discretised wavelet transform, there are only a finite number of wavelet coefficients for
each bounded rectangular region in the upper halfplane. Still, each coefficient requires the
evaluation of an integral. In special situations this numerical complexity can be avoided if the
scaled and shifted wavelets form a multiresolution analysis. This means that there has to exist an
auxiliary function, the father wavelet φ in L2(R), and that a is an integer. A typical choice is a = 2
and b = 1. The most famous pair of father and mother wavelets is the Daubechies 4-tap wavelet.
UNIT-I
Learning Objectives
• Why process signals digitally?
• Definition of a real-time application.
• Why use Digital Signal Processing
processors?
• What are the typical DSP algorithms?
• Parameters to consider when choosing
a DSP processor.
Why go digital?
• Digital signal processing techniques are now so
powerful that sometimes it is extremely
difficult, if not impossible, for analogue signal
processing to achieve similar performance.
• Examples:
– FIR filter with linear phase.
– Adaptive filters.
Why go digital?
• Analogue signal processing is achieved
by using analogue components such as:
– Resistors.
– Capacitors.
– Inductors.
• The inherent tolerances associated with
these components, temperature,
voltage changes and mechanical
vibrations can dramatically affect the
effectiveness of the analogue circuitry.
Why go digital?
• With DSP it is easy to:
– Change applications.
– Correct applications.
– Update applications.
• Additionally DSP reduces:
– Noise susceptibility.
– Chip count.
– Development time.
– Cost.
– Power consumption.
Why NOT go digital?
• High frequency signals cannot be
processed digitally because of two
reasons:
–Analog to Digital Converters, ADC
cannot work fast enough.
–The application can be too
complex to be performed in real-
time.
Real-time processing
• DSP processors have to perform tasks in
real-time, so how do we define real-time?
• The definition of real-time depends on the
application.
• Example: a 100-tap FIR filter is performed
in real-time if the DSP can perform and
complete the following operation between
two samples:
Real-time processing
Waiting Time
Processing Time
n n+1
Sample Time
• Voice transmission
• Transmission systems
Cascade form
Parallel form
UNIT-II
23
Discrete Fourier Series
~
x[n] ~
x[n rN]
~
• Given a periodic sequence x[n] with period N so that
~ 1 ~
x[n] Xk e j2 / Nkn
N k
• The Fourier series representation can be written as
e j2 / Nk mNn e j2 / Nkne j2 mn e j2 / Nkn
X k e
~ 1 ~ j 2 / N kn
x [ n]
N k 0
W N e j 2 / N
• For convenience we sometimes use
~ N 1
~
Xk x[n]Wkn
N
• Analysis equation n0
~ 1 N 1 ~
x[n] Xk WNkn
• Synthesis equation
N k 0
25
Example 1
~
1 n rN
x[n] n rN
r 0 else
• DFS of a periodic impulse train
~ N 1 N 1
Xk ~x[n]e j2 / Nkn
n0
[n]e
n0
j2 / N kn
e j2 / Nk 0 1
~ ~
x n DFS
Xk
~ j2 km / N~
• Shift of a Sequence x n m DFS
e Xk
~
e j2 nm / N~
x n DFS
Xk m
~ ~
• Duality x n DFS
Xk
~
Xn DFS
N~
x k
Symmetry Properties
Symmetry Properties Cont’d
Periodic Convolution
• Take two periodic sequences
~ ~
x1 n DFS
X1 k
~ ~
x2 n DFS
X2 k
• Let’s form the product
X 3 k X 1 k X 2 k
~ ~ ~
N X 2 k
km ~
2
~
x [
n 0
n m ]
W kn
N W
1 2 ~ j jn 1 2 2 ~
2 X e e d
2 Xk
2k jn
e d
k N N
0 0
2 k
1 ~ 2 2k jn 1 N1 ~
Xk e d Xk e
j n
N k 0
N N k 0
N
Relation between Finite-length and Periodic Signals
n 0 N n 0
x[ n
n even
]W N
nk
x[
n odd
n ]W nk
N
N 1 N 1
x[2r ](WN2 ) rk WNk x[2r 1](WN2 ) rk
2 2
r 0 r 0
N 1 N 1
x[2r ]W
2 2
rk
N /2 W k
N x[ 2 r 1]W rk
N /2
r 0 r 0
DIT Algorithm (cont.)
• Result is the sum of two N/2 length DFTs
X [k ] G
[k ] WNk H
[k ]
N/2 DFT N/2 DFT
of even samples of odd samples
x[0,2,4,6] N/2
DFT
X[0…7]
x[1,3,5,7] N/2
DFT
WN0... 7
Detail of “Butterfly”
• Cross feed of G[k] and H[k] in flow diagram is
called a “butterfly”, due to shape
W Nr
or simplify:
WN( r N 2 ) W Nr -1
( WNr )
8-point DIT-FFT Diagram
x[0,4,2,6,
X[0…7]
1,5,3,7]
W N0 1
W N0 1
W N0 1 WN2 1
W N0 1
W N0 1 W N1 1
W N0 1 WN2 1
W N0 1 WN2 W N3
1 1
8-point DIF-FFT Diagram
UNIT-III
Chebyshev 1 Approximation:
The Chebyshev 1 approximation for an ideal lowpass filter
has equal-valued ripples in the passband . It is known as
minimax approximation and also known as the equiripple
approximation.
Chebyshev II
UNIT-IV
Frequency response of FIR digital filters
we consider the FIR filters in which the impulse
response h[n] are assumed to be symmetric or
antisymmetric.
Since the order of the polynomial in each of these two
1 N 1 1 zN
H ( z ) H (k ). 2
N k 0 j k
1 e N .z 1
Comparison of FIR & IIR digital filters
UNIT-IV
Frequency response of FIR digital filters
we consider the FIR filters in which the impulse
response h[n] are assumed to be symmetric or
antisymmetric.
Since the order of the polynomial in each of these two
1 N 1 1 zN
H ( z ) H (k ). 2
N k 0 j k
1 e N .z 1
Comparison of FIR & IIR digital filters
Thank you
SET-1
1. a) Define an LTI System and show that the output of an LTI system is given by the convolution of
Input sequence and impulse response.
b) Prove that the system defined by the following difference equation is an LTI system y(n) =
x(n+1)- 3x(n)+x(n-1) ; n≥0. [8+8 ]
b) Find 8-Point DFT of the given time domain sequence x(n) = {1, 2, 3, 4}. [8+8]
3. a) Derive the expressions for computing the FFT using DIT algorithm and hence draw the standard
butterfly structure.
4. Discuss and draw various IIR realization structures like Direct form – I, Direct form-II, Parallel and
cascade forms for the difference equation given y(n) = - 3/8 Y(n-1) + 3/32 y(n-2) + 1/64 y(n-3) + x(n)
+ 3 x(n-1) + 2 x(n-2).
b) Design a Digital Butterworth LPF using Bilinear transformation technique for the following
specifications 0.707 ≤ | H(w) | ≤ 1 ; 0 ≤ w ≤ 0.2π | H(w) | ≤ 0.08 ; 0.4 π ≤ w ≤ [ 8+8]
b) Design an FIR Digital High pass filter using Hamming window whose cut off freq is 1.2 rad/s
and length of window N=9. [8+8]
b) Discuss the process of n Decimation by a factor D and explain how the aliasing effect can be
eliminated. [8+8]
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}. [8+8]
3.a) Find X(K) of the given sequence x(n) = { 1,2,3,4,4,3,2,1}using DIT- FFT algorithm
4. What are the various basic building blocks in realization of Digital Systems and hence discuss
transposed form realization structures.
b) Compute the poles of an Analog Chebyshev filter TF that satisfies the Constraints 0.707 ≤ |
H(jΩ)| ≤ 1 ; 0 ≤ Ω≤ 2 | H(jΩ)| ≤ 0.1 ; Ω ≥ 4 and determine Ha(s) and hence obtain H(z) using Bilinear
transformation. [16]
b) Design an FIR Digital Low pass filter using Hanning window whose cut off freq is 2 rad/s and
length of window N=9. [8+8]
b) Discuss the sampling rate conversion by a factor I/D with the help of a Neat block Diagram.
[8+8]
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}. [8+8]
b) Discuss Direct form, Cascade and Linear phase realization structures of FIR filters. [8+8 ]
b) Discuss IIR filter design using Bilinear transformation and hence discuss frequency warping
effect. [8+8]
b) Design an FIR Digital Low pass filter using rectangular window whose cut off freq is 2 rad/s and
length of window N=9. [8+8]
7. a) Define Interpolation and Decimation. List out the advantages of Sampling rate conversion.
b) Discuss the sampling rate conversion by a factor I with the help of a Neat block Diagram.[8+8]
b) Give the Internal Architecture of TMS320C5X 16 bit fixed point processor.[ 8+8]
SET-4
b) The discrete time system is represented by the following difference equations in which x(n) is
input and y(n) is output. Y(n) = 3y 2 (n-1)- nx(n)+4x(n-1)-2x(n-1). [8+8]
b) Find the Linear convolution of the given two sequences x(n)={1,2} and h(n) ={1,2,3} using
DFT and IDFT. [8 +8]
3. a) Develop DIT-FFT algorithm and draw signal flow graphs for decomposing the DFT for N=6 by
considering the factors for N = 6 = 2.3.
b) Discuss IIR filter design using Impulse Invariant transformation and list out its advantages and
Limitations. [8+8]
b) Design an FIR Digital Band pass filter using rectangular window whose upper and lower cut off
freq.’s are 1 & 2 rad/s and length of window N = 9. [8+8]
b) Discuss the sampling rate conversion by a factor I/D with the help of a Neat block Diagram. [8
+8]
b)Give the Internal Architecture of TMS320C5X 16 bit fixed point processor. [8+8]
SET-1
1. (a) Discuss impulse invariance method of deriving IIR digital filter from corre- sponding analog
filter.
(b) Use the Bilinear transformation to convert the analog filter with system func- tion H(S) = S +
0.1/(S + 0.1)2 + 9 into a digital IIR filters. Select T = 0.1 and compare the location of the zeros in
H(Z) with the locations of the zeros obtained by applying the impulse invariance method in the
conversion of H(S). [8+8]
2. (a) Design a high pass filter using hamming window with a cut-off frequency of 1.2 radians/second
and N=9
3. (a) For each of the following systems, determine whether or not the system is i. stable ii. causal iii.
linear iv. shift-invariant. A. T[x(n)] = x(n − n0 ) B. T [x(n)] = e x (n) C. T[x(n)] = a x(n) + b. Justify
your answer.
(b) A system is described by the difference equation y(n)-y(n-1)-y(n-2) = x(n1). Assuming that the
system is initially relaxed, determine its unit sample response h(n). [8+8]
(b) In the above Question how many non - trivial multiplications are R e q u i r e d .
5. (a) Discuss the frequency-domain representation of discrete-time systems and sig- nals by deriving
the necessary relation.
(b) Draw the frequency response of LSI system with impulse response h(n) = a n u(−n) (|a| < 1)
(b) Give an expression for the following parameters relative to RADAR i. Beam width ii.
Maximum unambiguous range
8. (a) Compute Discrete Fourier transform of the following finite length sequence considered to be of
length N. i. x(n) = δ(n + n0 ) where 0 < n0 < N ii. x(n) = a n where 0 < a < 1.
(b) If x(n) denotes a finite length sequence of length N, show that x((−n))N =x((N − n))N . [8+8]
SET-2
1. (a) Design a high pass filter using hamming window with a cut-off frequency of 1.2 radians/second
and N=9
(b) Give an expression for the following parameters relative to RADAR i. Beam width ii.
Maximum unambiguous range (c) Discuss signal processing in a RADAR system. [6 +6+4]
(b) Define stable and unstable systems. Test the condition for stability of the first-order IIR filter
governed by the equation y(n)=x(n)+bx(n1). [8+8]
4. (a) Compute Discrete Fourier transform of the following finite length sequence considered to be of
length N. i. x(n) = δ(n + n0 ) where 0 < n0 < N ii. x(n) = a n where 0 < a < 1.
(b) If x(n) denotes a finite length sequence of length N, show that x((−n))N = x((N − n))N . [8+8]
5. (a) For each of the following systems, determine whether or not the system is i. stable ii. causal iii.
linear iv. shift-invariant. A. T[x(n)] = x(n − n0 ) B. T [x(n)] = e x (n) C. T[x(n)] = a x(n) + b. Justify
your answer
(b) A system is described by the difference equation y(n)-y(n-1)-y(n-2) = x(n1). Assuming that the
system is initially relaxed, determine its unit sample response h(n). [8+8]
6. (a) Discuss the frequency-domain representation of discrete-time systems and signals by deriving
the necessary relation.
(b) Draw the frequency response of LSI system with impulse response h(n) = a n u(−n) (|a| < 1)
[8+8]
(b) In the above Question how many non - trivial multiplications are required.
8. (a) Discuss impulse invariance method of deriving IIR digital filter from corre- sponding analog
filter. (b) Use the Bilinear transformation to convert the analog filter with system func- tion H(S) =
S + 0.1/(S + 0.1)2 + 9 into a digital IIR filters. Select T = 0.1 and compare the location of the zeros in
H(Z) with the locations of the zeros obtained by applying the impulse invariance method in the
conversion of H(S). [8+8]