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Dspimp
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COURSE FILE
Contents
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2. SYLLABUS
GEETHANJALI COLLEGE OF ENGINEERING & TECHNOLOGY
Introduction to Digital Signal Processing, Discrete time signals and sequences, Linear shift
invariant Systems, Stability and Causality, Linear constant coefficient difference equations,
Frequency domain representation of discrete time signals and systems.
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 between DTFT, DFS, DFT and Z
transform.
FAST FOURIER TRANSFORMS
UNIT III:
IIR DIGITAL FILTERS
Analog filter approximations - Butter worth and Chebyshev, design IIR Digital Filters from Analog
filters, Step and Impulse InvariantTechniques,Bilinear Transformation Method, Spectral
transformations.
Characteristics of FIR digital filters, Frequency response, Design of FIR digital filters:Fourier
Method, Digital Filters using Window Techniques, Frequency Sampling Technique, Comparison
of IIR & FIR filters.
UNIT V: MULTIRATE DIGITAL SIGNAL PROCESSING
Finite word length effects: Limit cycles, Overflow oscillations, Round off Noise in IIR Digital Filters,
Computational output Round off noise, Methods to prevent Overflow, Tradeoff between Round off and
Over flow Noise, Dead Band Effects.
This course will introduce the basic concepts and techniques for processing signals on a computer.
By the end of the course, students will be familiar with the most important methods in DSP,
including digital filter design, transform-domain processing and importance of Signal Processors.
The course emphasizes intuitive understanding and practical implementations of the theoretical
concepts.
To produce graduates who understand how to analyze and manipulate digital signals and have the
fundamental Mat lab programming knowledge to do so.
Course Outcomes:
CO 2: Able to calculate Z-transforms for discrete time signals and system functions.
CO 3: Ability to calculate discrete time domain and frequency domain of signals using discrete Fourier
series and Fourier transform.
CO 4:.Ability to develop Fast Fourier Transform (FFT) algorithms for faster realization of signals and
systems.
CO 5: Able to design Digital IIR filters from Analog filters using various techniques (Butterworth and
Chebyshev).
CO 6: Able to design Digital FIR filters using window techniques,Fouriour methods and frequency
sampling technique..
CO 8: Ability to demonstrate the impacts of finite word length effects in filter design.
8. PREREQUISITES, IF ANY
Laplace Transforms
Fourier Transforms
UNIT-I (INTRODUCTION)
1) Students can understand the concept of discrete time signals & sequences.
2) Analyze and implement digital signal processing systems in time domain.
3) They can solve linear constant coefficient difference equations.
4) They can understand Frequency domain representation of discrete time signals and systems.
5) They can understand the practical purpose of stability and causality.
6) To determine stability, causality for a given impulse response.
7) Understand how analog signals are represented by their discrete-time samples, and in what ways
digital filtering is equivalent to analog filtering.
8) The basics of Z-transforms and its applications are studied.
9) Digital filters are realized using difference equations.
10) Calculate the response of applying a given input signal to a system described by a linear constant
coefficient differential equation.
1) Ability to understand discrete time domain and frequency domain representation of signals and
systems.
2) Compute convolution and the discrete Fourier transform (DFT) of discrete-time signals.
3) Analyze and implement digital systems using the DFT.
4) Ability to understand Discrete Fourier Series and Transforms and comparison with other
transforms like Z transforms.
5) Ability to represent discrete-time signals in the frequency domain.
6) Calculate exponential Fourier series coefficients using properties of Fourier series
7) Graphically portray the magnitude and phase of the Fourier series coefficients versus ω.
Fast Fourier Transforms
8) Ability to develop Fast Fourier Transform algorithms for faster realization of signals and systems.
9) Ability to understand discrete time domain and frequency domain representation of signals and
systems.
10) Analyze and implement digital systems using the FFT.
11) Describe how and why Fourier Transforms and Fourier series are related.
1) Ability to understand the characteristics of linear-phase finite impulse response (FIR) filters
2) Ability to understand Digital Filters with special emphasis on realization of FIR and IIR filters.
3) Ability to design linear-phase FIR filters according to predefined specifications using the window
and frequency sampling methods
4) Ability to understand the concepts of Digital FIR filters.
5) The student is able to specify and design respective frequency selective FIR and IIR filters using
the most common methods.
6) The student is able to solve for the impulse and frequency responses of FIR and IIR filters given
as difference equations, transfer functions, or realization diagrams, and can present analyses of the
aliasing and imaging effects based on the responses of the filters.
7) Measure the effectiveness of FIR filters.
1) Ability to understand the concepts of sampling rate conversions, Decimation and Interpolation as
part of Signal Processing techniques.
2) Able to explain how the multirate implementation of ADC and DAC converters works.
3) Able to describe basic sampling rate conversion algorithms.
4) Able to draw and describe different kinds of interpolator and decimator.
5) Able to analyze how the interpolated FIR filter works.
6) Able to do sampling rate conversion.
*When the course outcome weightage is < 40%, it will be given as moderately correlated (1).
*When the course outcome weightage is >40%, it will be given as strongly correlated (2).
POs 1 2 3 4 5 6 7 8 9 10 11 12 13
Digital signal
Digital Signal
Processing
2 2 2 2 2 2
Processing
CO 1: Able to 2 2
s
demonstrate
different Analog and
Discrete time signals.
CO 2: Ability to 2 2 2 2
calculate discrete
time domain and
frequency domain of
signals using discrete
Fourier Series and
Fourier transform.
CO 3: Ability to 2 2 2 2 2
develop Fast Fourier
Transform (FFT)
algorithms for faster
realization of signals
and systems.
CO 4: Able to 2 2 2 2
calculate Z-
transforms for
discrete time signals
and system
functions.
CO 5: Able to 2 1 2 2 2 2
design Digital IIR
filters from Analog
filters using various
techniques
(Butterworth and
Chebyshev).
CO 6: Able to 2 1 2 2 2 2
design Digital FIR
filters using
window
techniques,Fouriour
methods and
frequency sampling
technique.
CO 7: Ability to 2 2 2 2 2
design different
kinds of interpolator
and decimator.
CO 8: Ability to 2 2 2 2 2
demonstrate the
impacts of finite
word length effects
in filter design.
To be attached
Day 1 2 3 4 5 6 7
Mon DSP LUNCH DSP lab
TUE DSP lab DSP
WED DSP
THUR
FRI DSP
SAT DSP(T) Technical seminars
Sl. Unit Total Date Topic to be covered in One Regular/ Teaching Remarks
No. number lecture Additional/ Aids used
No. of Missing LCD
periods /OHP/BB
sequences
3 31/12/14 Linear shift invariant systems, Regular BB
stability and causality
4 01/01/15 Linear constant coefficient Regular BB
16 difference equations
5 02/01/15 Frequency domain Regular BB/OHP
representation of discrete time
signals and systems
Wavelet Transforms Missing BB/OHP
7 Review of Z transforms
8 Applications of Z transforms
,solution of difference equations
9 Block diagram
of digital filtersrepresentation of
linear constant coefficient
10 Basic structures
difference of IIR systems,
equations
Transposed forms
11 Basic structure of FIR systems,
System function
12 Tutorial class-2
16 MID TEST 1
30 Tutorial class-4
- Butter worth
III
37 Spectral Transformations
38 Solving university question
paper
39 Revision
40 Tutorial class-5
50 Assignment test-4
51 10/02/15 Concept of Multirate signal Regular BB/OHP
Processing
52 11/02/15 Decimation Regular BB/OHP
53 12/02/15 interpolation BB
16/03/15 BB
MID TEST II
GUIDELINES:
Distribution of periods:
In the class, starting from the basic definitions of a discrete-time signal, we will work our way
through Fourier analysis, filter design, sampling, interpolation and quantization to build a DSP
toolset complete enough to analyze a practical communication system in detail. Hands-on examples
and demonstration will be routinely used to close the gap between theory and practice.
The digital signal processor can be programmed to perform a variety of signal processing
operations, such as filtering, spectrum estimation, and other DSP algorithms. Depending on the
speed and computational requirements of the application, the digital signal processor may be
realized by a general purpose computer, minicomputer, special purpose DSP chip, or other digital
hardware dedicated to performing a particular signal processing task.
For this reason IIR filters have much better frequency response than FIR filters of the same order. Unlike
FIR filters, their phase characteristic is not linear which can cause a problem to the systems which need
phase linearity. For this reason, it is not preferable to use IIR filters in digital signal processing when the
phase is of the essence.
Otherwise, when the linear phase characteristic is not important, the use of IIR filters is an excellent
solution.
There is one problem known as a potential instability that is typical of IIR filters only. FIR filters do not
have such a problem as they do not have the feedback. For this reason, it is always necessary to check
after the design process whether the resulting IIR filter is stable or not.
IIR filters can be designed using different methods. One of the most commonly used is via the reference
analog prototype filter. This method is the best for designing all standard types of filters such as low-pass,
high-pass, band-pass and band-stop filters.
This book describes the design method using reference analog prototype filter. Figure 3-1-2 illustrates the
block diagram of this method.
UNIT V - Multirate Digital Signal Processing
The process of converting a signal from a given rate to a different rate is called sampling
rate conversion. In turn,
systems that employ multiple sampling rates in the processing of digital signals are called
multirat~digital signal processing systems.
Sampling rate conversion of a digital signal can be accomplished in one of two general methods.
One method is to pass the digital signal through a DIA converter, filter it if necessary, and then
to resample the resulting analog signal at the desired rate (i.e., to pass the analog signal through
an AID converter). The second method is to perform the sampling rate conversion entirely in the
digital domain.One apparent advantage of the first method is that the new sampling rate can be
arbitrarily selected and need not have any special relationship to the old sampling rate. A major
disadvantage, however, is the signal distortion, introduced by the DIA converter in the signal
reconstruction, and by the quantization effects in the AD conversion. Sampling rate conversion
performed in the digital domain avoids this major disadvantage.Here we describe sampling rate
conversion and multirate signal processing in the digital domain. First we describe sampling rate
conversion by a rational factor and present several methods for implementing the rate converter,
including single-stage and multistage implementations. Then, we describe a method for sampling
rate conversion by an arbitrary factor and discuss its implementation.Finally, we present several
applications of sampling rate conversion in multirate signal processing systems, which include
the implementation of narrowband filters,digital filter banks, and quadrature mirror filters. We
also discuss the use of quadrature mirror filters in subband coding. transmultiplexers. and finally
oversampling A/D and D/A converters.
INTRODUCTION
The process of sampling rate conversion in the digital domain can be viewed asa linear
filtering operation, as illustrated . The input signal x ( n )is characterized by the sampling rate F,
= 1/T, and the output signal y(m) is characterized by the sampling rate F! = l/T,., where T, and
7j, are the correspondingsampling intervals. In the main part of our treatment, the ratio F,
/Flisconstrained to be rational,where D and I are relatively prime integers. We shall show that
the linear filteris characterized by a time-variant impulse response. denoted as h(tl. m). Hencethe
input x(?r) and the output y(m) are related by the convolution summation fortime-variant
systems.The sampling rate conversion process can also be understood from the pointof view of
digital resampling of the same analog signal. Let x ( r ) be the analogsignal that is sampled at the
first rate F, to generate x(t1). The goal ofrate conversion is to obtain another sequence j ~ ( n t )
directly from x ( n ) . whichis equal to the sampled values of x t t ) at a second rate F!. As is
depicted .!*(m) is a time-shifted version of x ( n ) . Such a time shift can be
UNIT V - Multirate Digital Signal Processing
The process of converting a signal from a given rate to a different rate is called sampling
rate conversion. In turn,
systems that employ multiple sampling rates in the processing of digital signals are called
multirat~digital signal processing systems.
Sampling rate conversion of a digital signal can be accomplished in one of two general methods.
One method is to pass the digital signal through a DIA converter, filter it if necessary, and then
to resample the resulting analog signal at the desired rate (i.e., to pass the analog signal through
an AID converter). The second method is to perform the sampling rate conversion entirely in the
digital domain.One apparent advantage of the first method is that the new sampling rate can be
arbitrarily selected and need not have any special relationship to the old sampling rate. A major
disadvantage, however, is the signal distortion, introduced by the DIA converter in the signal
reconstruction, and by the quantization effects in the AD conversion. Sampling rate conversion
performed in the digital domain avoids this major disadvantage.Here we describe sampling rate
conversion and multirate signal processing in the digital domain. First we describe sampling rate
conversion by a rational factor and present several methods for implementing the rate converter,
including single-stage and multistage implementations. Then, we describe a method for sampling
rate conversion by an arbitrary factor and discuss its implementation.Finally, we present several
applications of sampling rate conversion in multirate signal processing systems, which include
the implementation of narrowband filters,digital filter banks, and quadrature mirror filters. We
also discuss the use of quadrature mirror filters in subband coding. transmultiplexers. and finally
oversampling A/D and D/A converters.
INTRODUCTION
The process of sampling rate conversion in the digital domain can be viewed asa linear
filtering operation, as illustrated . The input signal x ( n )is characterized by the sampling rate F,
= 1/T, and the output signal y(m) is characterized by the sampling rate F! = l/T,., where T, and
7j, are the correspondingsampling intervals. In the main part of our treatment, the ratio F,
/Flisconstrained to be rational,where D and I are relatively prime integers. We shall show that
the linear filteris characterized by a time-variant impulse response. denoted as h(tl. m). Hencethe
input x(?r) and the output y(m) are related by the convolution summation fortime-variant
systems.The sampling rate conversion process can also be understood from the pointof view of
digital resampling of the same analog signal. Let x ( r ) be the analogsignal that is sampled at the
first rate F, to generate x(t1). The goal ofrate conversion is to obtain another sequence j ~ ( n t )
directly from x ( n ) . whichis equal to the sampled values of x t t ) at a second rate F!. As is
depicted .!*(m) is a time-shifted version of x ( n ) . Such a time shift can be
4.1. Introduction
An FIR digital filter of order M may be implemented by programming the signal-flow-graph shown
below. Its difference equation is:
y[n] = a0x[n] + a1x[n-1] + a2x[n-2] + ... + aMx[n-M]
x[n]
z-1 z-1 .. z-1 z-1
.
aM
aM-1
a0 a1
y[n]
Fig. 4.1
Its impulse-response is {..., 0, ..., a0, a1, a2,..., aM, 0, ...} and its frequency-response is the DTFT of
the impulse-response, i.e.
M
H ( e j ) h[n]e jn
n
a e
n 0
n
jn
j
Now consider the problem of choosing the multiplier coefficients. a0, a1,..., aM such that H( e ) is
close to some desired or target frequency-response H(ej) say. The inverse DTFT of H’(ej) gives
the required impulse-response :
1
h[n] H (e
j
)e jn d
2
The methodology is to use the inverse DTFT to get an impulse-response {h[n]} & then realise some
approximation to it Note that the DTFT formula is an integral, it has complex numbers and the range
of integration is from - to , so it involves negative frequencies.
Reminders about integration
dx
(1) If x(t ) e at then aeat
dt
x(t )dt
e
at
dt
1
e at
1 a
e e a
a a
FINITE WORD LENGTH EFFECTS
Practical digital filters must be implemented with finite precision numbers and arithmetic. As a
result, both the filter coefficients and the filter input and output signals are in discrete form. This
leads to four types of finite wordlength effects. Discretization (quantization) of the filter
coefficients has the effect of perturbing the location of the filter poles and zeroes. As a result, the
actual filter response differs slightly from the ideal response. This deterministic frequency
response error is referred to as coefficient quantization error. The use of finite precision
arithmetic makes it necessary to quantize filter calculations by rounding or truncation.
Roundoffnoise is that errorin the filter output that resultsfrom rounding or truncating calculations
within the filter. As the name implies, this error looks like low-level noise at the filter output.
Quantization of the filter calculations also renders the filter slightly nonlinear. For large signals
this nonlinearity is negligible and roundoff noise is the major concern. However, for recursive
filters with a zero or constant input, this nonlinearity can cause spurious oscillations called limit
cycles. With fixed-point arithmetic it is possible for filter calculations to overflow. The term
overflow oscillation, sometimes also called adder overflow limit cycle, refers to a high-level
oscillation that can exist in an otherwise stable filter due to the nonlinearity associated with the
overflow of internal filter calculations. In this chapter, we examine each of these finite length
effects. Both fixed-point and floatingpoint number representations are consider
Limit Cycles
Additional/missing topics
Speech processing
Radar Signal Processing
DSP Processors
Pulse Code Modulation
Correlation
Geortzel algorithm
FIR Least square design
methods
Multi stage implementation of sampling rate
conversion
1. Correlation
The concept of correlation can best be presented with an example. Figure 7-13 shows the key
elements of a radar system. A specially designed antenna transmits a short burst of radio wave
energy in a selected direction. If the propagating wave strikes an object, such as the helicopter in
this illustration, a small fraction of the energy is reflected back toward a radio receiver located
near the transmitter. The transmitted pulse is a specific shape that we have selected, such as the
triangle shown in this example. The received signal will consist of two parts: (1) a shifted and
scaled version of the transmitted pulse, and (2) random noise, resulting from interfering radio
waves, thermal noise in the electronics, etc. Since radio signals travel at a known rate, the speed
of light, the shift between the transmitted and received pulse is a direct measure of the distance to
the object being detected. This is the problem: given a signal of some known shape, what is the
best way to determine where (or if) the signal occurs in another signal. Correlation is the answer.
What if the target signal contains samples with a negative value? Nothing changes. Imagine that
the correlation machine is positioned such that the target signal is perfectly aligned with the
matching waveform in the received signal. As samples from the received signal fall into the
correlation machine, they are multiplied by their matching samples in the target signal.
Neglecting noise, a positive sample will be multiplied by itself, resulting in a positive number.
Likewise, a negative sample will be multiplied by itself, also resulting in a positive number.
Even if the target signal is completely negative, the peak in the cross -correlation will still be
positive.
If there is noise on the received signal, there will also be noise on the cross-correlation signal. It
is an unavoidable fact that random noise looks a certain amount like any target signal you can
choose. The noise on the cross-correlation signal is simply measuring this similarity. Except for
this noise, the peak generated in the cross-correlation signal is symmetrical between its left and
right. This is true even if the target signal isn't symmetrical. In addition, the width of the peak is
twice the width of the target signal. Remember, the cross-correlation is trying to detect the target
signal, not recreate it. There is no reason to expect that the peak will even look like the target
signal.
Correlation is the optimal technique for detecting a known waveform in random noise. That is,
the peak is higher above the noise using correlation than can be produced by any other linear
system. (To be perfectly correct, it is only optimal for random white noise). Using correlation to
detect a known waveform is frequently called matched filtering.
The correlation machine and convolution machine are identical, except for one small difference.
As discussed in the last chapter, the signal inside of the convolution machine is flipped left-for-
right. This means that samples numbers: 1, 2, 3 … run from the right to the left. In the
correlation machine this flip doesn't take place, and the samples run in the normal direction.
Geortzel algorithm
The Goertzel algorithm is a digital signal processing (DSP) technique for identifying frequency
components of a signal, published by Gerald Goertzel in 1958. While the general Fast Fourier
transform (FFT) algorithm computes evenly across the bandwidth of the incoming signal, the
Goertzel algorithm looks at specific, predetermined frequencies.
A practical application of this algorithm is recognition of the DTMF tones produced by the
buttons pushed on a telephone keypad
It can also be used "in reverse" as a sinusoid synthesis function, which requires only 1
multiplication and 1 subtraction per sample.
Explanation of algorithm
The Goertzel algorithm computes a sequence, s(n), given an input sequence, x(n):
,
which becomes, assuming x(k) = 0 for all k < 0
or, the equation for the (n + 1)-sample DFT of x, evaluated for ω and
multiplied by the scale factor e + 2πiωn.
Note that applying the additional transform Y(z)/S(z) only requires the
last two samples of the s sequence. Consequently, upon processing N
samples x(0)...x(N − 1), the last two samples from the s sequence can be
used to compute the value of a DFT bin, which corresponds to the chosen
frequency ω as
For the special case often found when computing DFT bins, where
ωN = k for some integer, k, this simplifies to
Let the FIR filter length be L+1 samples, with even, and suppose we'll initially design it to
be centered about the time origin. Then the frequency response is given on our frequency grid
by
Enforcing even symmetry in the impulse response, i.e., , gives a zero phase
FIR filter which we can later right-shift samples to make a causal, linear phase
filter. In this case, the frequency response reduces to a sum of cosines:
or in matrix
form:
(Note that Remez exchange algorithms are also based on this formulation
internally.)
where
This is quadratic in , hence it has a global minimum which we can find by taking the
derivative, setting it to zero, and solving for . Doing this yields:
These are the famous normal equations whose solution is given by:
The matrix
Typically, the number of frequency constraints is much greater than the number of
design variables (filter taps). In these cases, we have an overdetermined system of
equations (more equations than unknowns). Therefore, we cannot generally satisfy
all the equations, and we are left with minimizing some error criterion to find the
``optimal compromise'' solution.
In practice, the least-squares solution can be found by minimizing the sum of squared
errors:
Figure 4.28 suggests that the error vector is orthogonal to the column space of
the matrix , hence it must be orthogonal to each column in :
This is how the orthogonality principle can be used to derive the fact that the best least
squares solution is given by
Note that the pseudo-inverse projects the vector onto the column space of .
(Note: To obtain the best numerical algorithms for least-squares solution in Matlab, it is
usually better to use ``x = A b'' rather than explicitly computing the pseudo-inverse as in
``x = pinv(A) * b''.)
The decimator and interpolator discussed so far are of a single-stage structure. When
large changes in sampling rate are required, multiple stages of sample rate conversion
are found
(decimation)
realization. The decimation in Figure 3.23 can be realized in two stages if the
decimation factor
D can be expressed as a product of two integers, D1 and D2. Referring to Figure 3.24,
in the first stage, the signal x(n) is decimated by a factor of D1. The output, v(p) is
by
D = (D1D2). The filters H1(z) and H2(z) are so designed that the aliasing in the band of
interest is belowa prescribed level and that the overall passband and stopband tolerances are
met. This multi-stage sampling rate conversion system offers less computation and more
flexibility in filter design. An example is given below to illustrate the idea of multi -stage
We have a discrete time signal with a sampling rate of 90 kHz. The signal has the
desired information in the frequency band from 0 to 450 Hz (passband), and the band from
450 to 500 Hz is the transition band. The signal is to be decimated by a factor of ninety.
The required tolerances are a passband ripple of 0.002 and a stopband ripple of 0.001.
According to the formula by Kaiser, the approximate length of an FIR filter is given
by
where peak passband ripple (linear) δp = 0.002, peak stopband ripple (linear) δs = 0.001,
normalized transition bandwidth passband edge frequency fp = 450 Hz,
stopband edge frequency fs = 500 Hz, and sampling frequency Fs = 90 kHz.
From Equation 3.37, the lowpass FIR filter H(z) has a length of N ≈ 5424. Therefore,
the number of multiplications per second, Msec, needed for this single-stage decimator is
Since only one out of ninety samples is actually used, the computation rate is based on the
Let us now consider the two-stage implementation of the decimation process as shown in
Figure 3.26.
Figure 3.25: (a) Block Diagram for Single-Stage Decimation, (b) The Filter Specification.
Figure 3.26: Block Diagram for Multi-Stage Decimation.
Due to the cascade decomposition, each of the two filters, H1(z) and H2(z), must have
a linear passband ripple specification half of that specified for the single-stage filter, H(z).
The stopband ripple specifications for these two filters can be the same as that o f H(z)
since the cascade connection will only reduce the stopband ripple.
Stage One
The first stage will decimate the input signal x(n) by a factor of forty-five. The filter
specifications for the first-stage LPF H1(z) are
The reason for choosing this value of the stopband edge is that, after decim ation by a
factor of forty-five, the residual energy of the signal in the band from 1000 to 2000 Hz will
be aliased back to the band from 0 to 1000 Hz. Due to the attenuation in the stopband, the
energy of the signal in the band from 1500 to 2000 Hz is very small compared to that in
1000 to 1500 Hz. So the amount of aliasing in the desired band of interest (0 to 450 Hz)
According to Equation 3.37, the approximate length of the FIR filter, H1(z) is N1 =
276. The number of multiplications per second for the first stage is
Stage Two
Figure 3.28 shows the characteristics of H2(z). This stage will perform a decimation
of factor two on the output signal of the first stage. So, the total decimation of x(n) is by a
For the second stage, the length of the filter, as calculated from Equation 3.37, is N2 =
The total number of multiplications per second required for the two-stage implementation
of the decimator is
So, the two-stage implementation requires only of the operation required of the
single-stage implementation
Stage One
In this stage, decimation by fifteen is performed on the input signal x(n). The
characteristics of the LPF, H1(z), are shown in Figure 3.30. The filter specifications
are
Figure 3.30: Decimation Filter Design for Stage One.
As in the two-stage case, the choice of stopband edge frequency can be extended to the
point for which negligible aliasing occurs in the passband (band of interest).
The approximate length of the filter as given by Equation 3.37 is N1 = 60. The number
Stage Two
In this stage, a decimation by a factor of three is done. The specifications of the LPF in this
As before, the stopband edge frequency can be stretched out to 1500 Hz. The filter
characteristics are shown in Figure 3.31.
The length of filter required for this stage is N2 = 20 and the number of multiplications
per second is
Stage Three
The third stage performs a decimation of factor two on the output of the second stage. The
Figure 3.32 shows the specifications for H3(z). As before, the approximate length of
the filter, as calculated from Equation 3.37, is N3 = 134. The number of multiplications
From this example, we can see that a significant saving in computation as well as in storage can
be achieved by a multi-stage decimator and interpolator design. These savings depend on the
optimum design of the number of stages and the choice of decimation factor for the individual
stages.
The examples illustrate the many different combinations and ordering possible. One approach is
to determine the sets of I and D factors that satisfy the filtering requirements and then estimate the
storage and computational costs for each set. The lowest cost solution is then selected.
16. UNIVERSITY QUESTON PAPERS OF PRUVIOIUS YEARS
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
]
2.a) Define DFT and IDFT. State any Four properties of DFT.
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.
b) Compare the computational complexity of FFT and DFT. [8+8]
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).
5.a) Compare Butterworth and Chebyshev approximation techniques.
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]
6.a) Derive the conditions to achieve Linear Phase characteristics of FIR filters
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]
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) Bring out the relationship between DFT and Z-transform. [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
(b) Compare FIR and IIR filters. [10+6]
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)] = ex (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(n-
1). Assuming that the system is initially relaxed, determine its unit
sample response h(n). [8+8]
2. (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]
3. (a) Describe how targets can be decided using RADAR
(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]
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)] = ex (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(n-
1). Assuming that the system is initially relaxed, determine its unit
sample response h(n). [8+8]
7. (a) Design a high pass filter using hamming window with a cut-off
frequency of
1.2 radians/second and N=9
(b) Compare FIR and IIR filters.
[10+6]
1. (a) Design a high pass filter using hamming window with a cut-off
frequency of
1.2 radians/second and N=9
(b) Compare FIR and IIR filters.
[10+6]
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)] = ex (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(n-
1). Assuming that the system is initially relaxed, determine its unit
sample response h(n). [8+8]
A. T [x(n)] = x(n
− n0 ) B. T
[x(n)] = ex (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(n-
1). Assuming that the system is initially relaxed, determine its unit
sample response h(n). [8+8]
[6+6+4]
4. (a) Design a high pass filter using hamming window with a cut-off
frequency of
1.2 radians/second and N=9
(b) Compare FIR and IIR filters.
[10+6]
5. (a) An LTI system is described by the equation y(n)=x(n)+0.81x(n-1)-
0.81x(n-
2)-0.45y(n-2). Determine the transfer function of the system. Sketch the poles and
zeroes on the Z-plane
(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(n-
1). [8+8]
TWO MARKS:
1. Define about DFT and IDFT?
2. Find the values of WN
k ,When N=8, k=2 and also for k=3.
3. Compare DIT radix-2 FFT and DIF radix -2 FFT.
4. Draw the radix-2 FFT–DIF butterfly diagram.
5. Draw the radix-2 FFT–DIT butterfly diagram.
6. What is the necessity of sectioned convolution in signal processing?
7. Define Correlation of the sequence.
8. State any two DFT properties.
9. Why impulse invariant transformation is not a one-to-one mapping?
PART - B
1. a) Compute 4- point DFT of casual three sample sequence is given by,
x(n) = 1/3, 0_n_2
= 0, else (10)
b) State and prove shifting property of DFT. (6)
2. Derive and draw the radix -2 DIT algorithms for FFT of 8 points. (16)
3. Compute the DFT for the sequence {1, 2, 0, 0, 0, 2, 1, 1}. Using radix -2 DIF FFT
and
radix -2 DIT- FFT algorithm. (16)
4. Find the output y(n) of a filter whose impulse response is h(n) = {1, 1, 1} and input
signal
x(n) = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1}. Using Overlap add overlap save method. (16)
5. In an LTI system the input x(n) = {1, 1, 1}and the impulse response h(n) = {-1, -
}Determine
the response of LTI system by radix -2 DIT FFT (16)
6. Find the output y(n) of a filter whose impulse response is h(n) = {1, 1, 1} and input
signal
x(n) = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1}. Using Overlap save method (16)
TWO MARKS:
1. Differentiate IIR filters and FIR filters.
2. Write the characteristics features of Hanning window
3. Define pre-warping effect? Why it is employed?
4. Give any two properties of Butterworth filter.
5. When a FIR filter is said to be a linear phase FIR filter
6. Write the characteristics features of rectangular window.
7. Write the expression for Kaiser window function..
8. What are the advantages and disadvantages of FIR filters?
9. Write the characteristics features of Hamming window
10. Why mapping is needed in the design of digital filters?
PART - B
1 . With a neat sketch explain the design of IIR filter using impulse invariant
transformation. (16)
2. Apply impulse invariant transformation to H(S) =
(S +1) (S + 2)
with T =1sec and find H(Z). (16)
3. For a given specifications of the desired low pass filter is
0.707 _ |H(_)| _1.0, 0 _ _ _ 0.2_
|H(_)| _ 0.08, 0.4 _ _ _ _ _
Design a Butterworth filter using bilinear transformation. (16)
4. Explain the procedural steps the design of low pass digital Butterworth filter and
list its
properties. (16)
5. The normalized transfer function of an analog filter is given by,
1
Ha(Sn)= Sn
2 + 1.414Sn +1
with a cutoff frequency of 0.4 _, using bilinear transformation. (16)
.
6. List the three well known methods of design technique for IIR filters and explain
any one.(16)
7. Design a low pass filter using rectangular window by taking 9 samples of w(n)
and with a cutoff frequency of 1.2 radians/sec.
Using frequency sampling method, design a band pass FIR filter with the following
specification. Sampling frequency Fs =8000 Hz, Cutoff frequency fc1 =1000Hz, fc2
=3000Hz.Determine the filter coefficients for N =7. (16)
8. Design an ideal high pass filter with Hd(ej _) = 1 ; _/4 _ | _| _ _
= 0 ; | _| _ _/4 Using Hamming window with N =11 (16)
9. Determine the coefficients of a linear phase FIR filter of length N =15 which has a
symmetric
unit sample response and a frequency response that satisfies the conditions
H (2 _k /15) = 1 ; for k = 0, 1, 2, 3
0.4 ; for k = 4
0 ; for k = 5, 6, 7 (16)
10. Design and implement linear phase FIR filter of length N =15 which has following
unit sample
sequence H(k) = 1 ; for k = 0, 1, 2, 3
0 ; for k =4, 5, 6, 7 (16)
11. Convert the analog filter in to a digital filter whose system function is
S + 0.2 H(s)= ---------------- .Use Impulse Invariant Transformation .Assume T=1sec
(16)
(S + 0.2)2 + 9
1
12. The Analog Transfer function H(s)= ----------------.Determine H(Z) .Using
Impulse (S+1) (S+2) Invariant Transformation .Assume T=1sec . (8)
13. Apply Bilinear Transformation to H(s)= ------------- with T=0.1 sec. (8)
(S+2)(S+3)
TWO MARKS:
1. What are the effects of finite word length in digital filters?
2. List the errors which arise due to quantization process.
3. Discuss the truncation error in quantization process.
4. Write expression for variance of round-off quantization noise.
5. What is sampling?
6. Define limit cycle Oscillations, and list out the types.
7. When zero limit cycle oscillation and Over flow limit cycle oscillation has occur?
8. Why? Scaling is important in Finite word length effect.
9. What are the differences between Fixed and Binary floating point number
representation?
10. What is the error range for Truncation and round-off process?
PART - B
1. The output of an A/D is fed through a digital system whose system function is
H(Z)=(1-_)z /(z-_), 0<_<1.Find the output noise power of the digital system. (8)
2. The output of an A/D is fed through a digital system whose system function is
H(Z)=0.6z/z-0.6. Find the output noise power of the digital system=8 bits (8)
3. Discuss in detail about quantization effect in ADC of signals. Derive the expression
for Pe(n) and SNR. (16)
4 a. Write a short notes on limit cycle oscillation (8)
b. Explain in detail about signal scaling (8)
5. A digital system is characterized by the difference equation
Y(n)=0.95y(n-1)+x(n).determine the dead band of the system when x(n)=0 and y(-
1)=13. (16)
6. Two first order filters are connected in cascaded whose system functions of
theIndividual sections are H1(z)=1/(1-0.8z-¹ ) and H2(z)=1/(1-0.9z -¹ ).Determine
theOver all output noise power. (16)
Assignment Questions
UNIT-I
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
onequati is an LTI system y(n) = x(n+1)-3x(n)+x(n-1) ; n≥0.
2.a) Write short notes on classification of systems.
b) Derive BIBO stability criteria to achieve stability of a system.
3.a) Discuss various discrete time sequences.
b) Give the Basic block diagram of Digital Signal Processor.
5. (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).
UNIT-II
1.a) Define DFS. State any Four properties of DFS.
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}.
3.(a) Design a high pass filter using hamming window with a cut-off frequency
1.2 radians/second and N=9
(b) Compare FIR and IIR filters.
4.a) Define DFS. State any Four properties of DFS.
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}.
5.a) Define DFT and IDFT. State any Four properties of DFT.
b) Find 8-Point DFT of the given time domain sequence x(n) = {1, 2, 3,
4}.
5.a) Find IFFT of the given X(K) = { 1,2,3,4,4,3,2,1}using DIF algorithm
b) Bring out the relationship between DFT and Z-transform.
6.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) Bring out the relationship between DFT and Z-transform.
7. (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)] = ex (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(n-
1). Assuming that the system is initially relaxed, determine its unit
sample response h(n).
8.a) Derive the expressions for computing the FFT using DIT algorithm
and hence draw the standard butterfly structure.
b) Compare the computational complexity of FFT and DFT.
UNIT-IV
UNIT-IV
1.a) Derive the conditions to achieve Linear Phase characteristics of FIR filters
b) Design an FIR Digital Low pass filter using Hanning window whose cut
1ff freq is 2 rad/s and length of window N=9.
UNIT-V
1. a) Discuss the implementation of Polyphase filters for Interpolators with an example
b) Discuss the sampling rate conversion by a factor I/D with the help of a Neat block Diagram.
2. a) Define Interpolation and Decimation.
b) Discuss the sampling rate conversion by a factor I/D with the help of aNeat block
Diagram.
3.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.
4. a) Define Multirate systems and Sampling rate conversion
b) Discuss the process of n Decimation by a factor D and explain how the aliasing
effect can be eliminated
5.(a) An LTI system is described by the equation
y(n)=x(n)+0.81x(n-1)-0.81x(n-2)-0.45y(n-2).
Determine the transfer function of the system. Sketch the poles and zeroes on the Z-plane.
(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(n-1).
UNIT-I: INTRODUCTION
1. Sketch the discrete time signal x(n) =4 _ (n+4) + _(n)+ 2 _ (n-1) + _ (n-2) -5 _ (n-3).
2. What is the causality condition for an LTI system?
3. Define stable and unstable systems.
4. Test the condition for stability of the first-order IIR filter governed by the equation
y(n)=x(n)+bx(n-1).
5. Define Linearity, Time Invariant, Stability and Causality.
REALIZATION OF DIGITAL FILTERS
1. What are two different types of structures for realization of IIR systems?
1. Direct-form1 structure
2.Direct-form2 structure
6.Lattice-ladder structure
3. Define signal flow graph.
4. A signal flow graph is a graphical representation of the relationship between the variables of
6. How Quantization errors can be minimized if we realize an LTI system in cascade form
2. Find the values of WNk ,When N=8, k=2 and also for k=3.
2. Derive and draw the radix -2 DIT algorithms for FFT of 8 points.
3. Compute the DFT for the sequence {1, 2, 0, 0, 0, 2, 1, 1}. Using radix -2 DIF FFT and
4. Find the output y(n) of a filter whose impulse response is h(n) = {1, 1, 1} and input
signal x(n) = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1}. Using Overlap add overlap save
method.
5. In an LTI system the input x(n) = {1, 1, 1}and the impulse response h(n) = {-1, -
6. Find the output y(n) of a filter whose impulse response is h(n) = {1, 1, 1} and input
transformation.
2. Apply impulse invariant transformation to H(S) = (S +1) (S + 2) with T =1sec and find
H(Z).
4. Explain the procedural steps the design of low pass digital Butterworth filter and list its
properties.
5. The normalized transfer function of an analog filter is given by,
1Ha(Sn)= Sn2 + 1.414Sn +1 with a cutoff frequency of 0.4 _, using bilinear
transformation. .
6. List the three well known methods of design technique for IIR filters and explain any
one.
7. Design a low pass filter using rectangular window by taking 9 samples of w(n)
and with a cutoff frequency of 1.2 radians/sec.Using frequency sampling method, design a
band pass FIR filter with the following specification. Sampling frequency F s =8000 Hz,
=7.
8. Design an ideal high pass filter with Hd(ej _) = 1 ; _/4 _ | _| _ _= 0 ; | _| _ _/4 Using
9. Determine the coefficients of a linear phase FIR filter of length N =15 which has a
symmetric unit sample response and a frequency response that satisfies the conditions H
(2 _k /15) = 1; for k = 0, 1, 2, 3
= 0.4 ; for k = 4
= 0; for k = 5, 6, 7
10. Design and implement linear phase FIR filter of length N =15 which has following unit
11. Convert the analog filter in to a digital filter whose system function is
T=1sec
12. The Analog Transfer function H(s)= ----------------.Determine H(Z) .Using Impulse (S+1)
3.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.
4. a) Define Multirate systems and Sampling rate conversion
b) Discuss the process of n Decimation by a factor D and explain how the aliasing
effect can be eliminated
5.(a) An LTI system is described by the equation
y(n)=x(n)+0.81x(n-1)-0.81x(n-2)-0.45y(n-2).
Determine the transfer function of the system. Sketch the poles and zeroes on the Z-plane.
(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(n-1).
5. What is sampling?
7. When zero limit cycle oscillation and Over flow limit cycle oscillation has occur?
9. What are the differences between Fixed and Binary floating point number
representation?
10. What is the error range for Truncation and round-off process?
PART - B
1. The output of an A/D is fed through a digital system whose system function is
H(Z)=(1-_)z /(z-_), 0<_<1.Find the output noise power of the digital system.
2. The output of an A/D is fed through a digital system whose system function is
H(Z)=0.6z/z-0.6. Find the output noise power of the digital system=8 bits
3. Discuss in detail about quantization effect in ADC of signals. Derive the expression for
Y(n)=0.95y(n-1)+x(n).determine the dead band of the system when x(n)=0 and y(-
1)=13.
6. Two first order filters are connected in cascaded whose system functions of
theIndividual sections are H1(z)=1/(1-0.8z-¹ ) and
H2(z)=1/(1-0.9z¹ ).Determine the Over all output noise power.
DIGITAL SIGNAL PROCESSORS
TWO MARKS:
4. What are the different buses of TMS 320C5x processor and list their functions
PART - B
2. Explain briefly:
(iii).VLIW architecture
(ii). Pipelining
4. Draw and explain the architecture of TMS 320C5x processor
ASSIGNMENT QUESTIONS:
UNIT-1
1. Determine the energy of the discrete time sequence (2)
x(n) = (½)n, n_0 =3 n, n<0
2. Define multi channel and multi dimensional signals. (2)
3. Define symmetric and anti symmetric signals. (2)
4. Differentiate recursive and non recursive difference equations. (2)
5. What is meant by impulse response? (2)
6. What is meant by LTI system? (2)
7. What are the basic steps involved in convolution? (2)
8. Define the Auto correlation and Cross correlation? (2)
9. What is the causality condition for an LTI system? (2)
10. What are the different methods of evaluating inverse z transform? (2)
11. What is meant by ROC? (2)
12. What are the properties of ROC? (2)
13. What is zero padding? What are it uses? (2)
14. What is an anti imaging and anti aliasing filter? (2)
15. State the Sampling Theorem. (2)
16. Determine the signals are periodic and find the fundamental period (2)
i) sin_ 2 _t
ii) sin 20_t+ sin5_t
17. Give the mathematical and graphical representations of a unit sample, unit step sequence.
(2)
18. Sketch the discrete time signal x(n) =4 _ (n+4) + _(n)+ 2 _ (n-1) + _ (n-2) -5 _ (n-3) (2)
19. Find the periodicity of x(n) =cost(2_n / 7) (2)
20. What is inverse system? (2)
21. Write the relationship between system function and the frequency response. (2)
22. Define commutative and associative law of convolutions. (2)
23. What is meant by Nyquist rate and Nyquist interval? (2)
24. What is an aliasing? How to overcome this effect? (2)
25. What are the disadvantages of DSP? (2)
26.Compare linear and circular convolution.(2)
27.What is meant by section convolution? (2)
28.Compare over lap add and save method. (2)
29. Define system function. (2)
30.State Parseval’s relation in z - transform. (2)
PART B
1. Determine whether the following system are linear, time-invariant (16)
(a)y(n) = Ax(n) +B. (4)
i(a)y(n) =x(2n). (4)
ii(a)y(n) =n x2 (n). (4)
iv)y(n) = a x(n) (4)
2. Check for following systems are linear, causal, time in variant, stable, static (16)
i) y(n) =x(2n). (4)
ii) y(n) = cos (x(n)). (4)
iii) y(n) = x(n) cos (x(n) (4)
iv) y(n) =x(-n+2) (4)
3. (a)For each impulse response determine the system is (a) stable i(a) causal
i) h(n)= sin (_ n / 2) . (4)
ii) h(n) = _(n) + sin _ n (4)
(b)Find the periodicity of the signal x(n) =sin (2_n / 3)+ cos (_ n / 2) (8)
4. (a)Find the periodicity of the signal
i) x(n) = cos (_ /4) cos(_n /4). (4)
ii) x(n) = cos (_ n 2 / 8) (4)
(b) State and proof of sampling theorem. (8)
5. Explain in detail about A to D conversion with suitable block diagram and to
reconstruct the signal. (16)
6. What are the advantages of DSP over analog signal processing? (16)
CONVOLUTION:
8. Find the output of an LTI system if the input is x(n) =(n+2) for 0_ n_ 3
and h(n) =an u(n) for all n (16)
9. Find the convolution sum of x(n) =1 n = -2,0,1
= 2 n= -1
= 0 elsewhere
and h(n) = _ (n) – _ (n-1) + _( n-2) - _ (n-3). (16)
10. (a)Find the convolution of the following sequence x(n) = u(n) ; h(n) =u(n-3). (8)
(b)Find the convolution of the following sequence x(n) =(1,2,-1,1) , h(n) =(1, 0 ,1,1). (8 )
12.Find the output sequence y(n) if h(n) =(1,1,1) and x(n) =(1,2,3,1) using a circular
Convolution. (16)
13. Find the convolution y(n) of the signals (16)
x(n) ={ _ n, -3 _ n _ 5 and h(n) ={ 1, 0 _ n _ 4
0, elsewhere } 0, elsewhere }
UN IT-II :
20. The impulse response of LTI system is h(n)=(1,2,1,-1).Find the response of the system to
the input x(n)=(2,1,0,2) (16)
21. Determine the magnitude and phase response of the given equation
y(n) =x(n)+x(n-2) (16)
22. Determine the response of the causal system y(n) – y(n-1) =x(n) + x(n-1) to inputs
x(n)=u(n) and x(n) =2 –n u(n).Test its stability (16)
23. Determine the frequency response for the system given by
y(n)-y3/4y(n-1)+1/8 y(n-2) = x(n)- x(n-1) (16)
24. Determine the pole and zero plot for the system described difference equations
y(n)=x(n)+2x(n-1)-4x(n-2)+x(n-3) (16)
25.A system has unit sample response h(n) =-1/4 _(n+1)+1/2 _(n)-1/4 _(n-1).Is the system
BIBO stable? Is the filter is Causal? Find the frequency response? (16)
26. Find the output of the system whose input- output is related by the difference equation
y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the step input. (16)
27. Find the output of the system whose input- output is related by the difference equation
y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the x(n) =4 n u(n). (16)
UNIT – III
FAST FOURIER TRANSFORM
PART A ( 2marks)
1. How many multiplication and additions are required to compute N point DFT using
radix 2 FFT? (2)
2. Define DTFT pair. (2)
3. What are Twiddle factors of the DFT? (2)
4. State Periodicity Property of DFT. (2)
5. What is the difference between DFT and DTFT? (2)
6. Why need of FFT? (2)
7. Find the IDFT of Y (k) = (1, 0, 1, 0) (2)
8. Compute the Fourier transform of the signal x(n) = u(n) – u(n-1). (2)
9. Compare DIT and DIF? (2)
10. What is meant by in place in DIT and DIF algorithm? (2)
11. Is the DFT of a finite length sequence is periodic? If so, state the reason. (2)
12. Draw the butterfly operation in DIT and DIF algorithm? (2)
13. What is meant by radix 2 FFT? (2)
14. State the properties of W N
k ? (2)
15. What is bit reversal in FFT? (2)
16. Determine the no of bits required in computing the DFT of a 1024 point sequence with
SNR of 30dB. (2)
17. What is the use of Fourier transform? (2)
18. What are the advantages FFT over DFT? (2)
19. What is meant by section convolution? (2)
20. Differentiate overlap adds and save method? (2)
21.Distinguish between Fourier series and Fourier transform. (2)
22.What is the relation between fourier transform and z transform. (2)
23. Distinguish between DFT and DTFT. (2)
PART B
1.(a) Determine the Fourier transform of x (n) =a |n|; -1<1 (8)
(b) Determine the Inverse Fourier transform H (w) = (1-ae-jw) -1 (8)
2. State and proof the properties of Fourier transform (16)
FFT:
PART B
1. Prove that an FIR filter has linear phase if the unit sample response satisfies the condition
h(n) = ± h(M-1-n), n =0,1,….. M-1.Also discuss symmetric and anti symmetric cases of FIR
filter. (16)
2. Explain the need for the use of window sequence in the design of FIR filter. Describe the
window sequence generally used and compare the properties. (16)
3. Explain the type 1 design of FIR filter using Frequency sampling technique. (16)
4. A LPF has the desired response given below (16)
H (e j )= e -3 j , 0 _ _ _ _/2
0. _/2 _ _ _ _ .Determine the filter coefficients h(n) for M=7
using frequency sampling technique.
5. Design a HPF of length 7 with cut off frequency of 2 rad/sec using Hamming window. Plot
the magnitude and phase response. (16)
6. Explain the principle and procedure for designing FIR filter using rectangular window (16)
7. Design a filter with H d (e- j) = e - 3 j , _/4 _ _ _ _/4
0. _/4 _ _ _ _ using a Hanning window with N=7. (16)
8.Design a FIR filter whose frequency response (16)
H (e j )= 1 _/4 _ _ _ 3_/4
0. | _ | _3 _/4.Calculate the value of h(n) for N=11 and hence find H(z).
9.Design an ideal differentiator with frequency response H (e j )= jw -_ _ _ _ _ using
hamming window for N=8 and find the frequency response. (16)
10.Design an ideal Hilbert transformer having frequency response
H (e j )= j -_ _ _ _ 0
-j 0 _ _ _ _ for N=11 using Blackman window. (16)
FIR structures:
12.(a) Determine the direct form of following system (8)
H (z) =1+2z-1 - 3z-2 + 4z-3 - 5z-4
(b) Obtain the cascade form realizations of FIR systems (8)
H (z) = 1+5/2 z-1+ 2z-2 +2 z-3
UNIT - VIII FINITE WORDLENGTH EFFECTS
PART A ( 2marks)
1. What are the three quantization errors due to finite world length registers in digital
filters?(2)
2. What do you mean by limit cycle oscillations? (2)
3. Explain briefly quantization noise. (2)
4. Represent 15.75 in fixed point and in floating point representations. (2)
5.What is the need for scaling in digital filters? (2)
6.List the well known techniques for linear phase FIR filter? (2)
7.What is quantization step size? (2)
8.State the advantages of floating point over fixed point representations? (2)
9.Why rounding preferred over truncation in realizing digital filter? (2)
10.What is meant by dead band? (2).
11.What is over flow limit cycle? How overflow can be eliminated? (2)
12.Sketch the noise probability density functions for rounding? (2)
13.Sketch the noise probability density functions for truncation?. (2)
14.What is meant by finite word length effect in digital filter? (2)
15.Explain the fraction 7/8 and -7/8 in sign magnitude ,1’s, 2’s complement. (2)
16.Convert in decimal to binary 20.675 (2)
17. Convert in binary to decimal 1110.01 (2)
18. What is product quantization error? (2)
19. What is input quantization error? (2)
20.What is coefficient quantization error? (2)
PART B
1. Explain in details about quantization in floating point realizations of IIR filter? (16)
2. Describe the effects of quantization in IIR filter. Consider a first order filter with difference
equation y(n) = x(n)+0.5 y(n-1) assume that the data register length is three bits plus a sign
bit.
The input x(n) = 0.875 _(n).Explain the limit cycle oscillations in the above filter, if
quantization is performed by means of rounding and signed magnitude representation is
used.(16)
3. Explain briefly
(1) Effects of coefficient quantization in filter design. (6)
(2) Effects of product round off error in filter design. (6)
(3) Speech recognition (4)
4. Explain briefly
(A)Define limit cycle oscillation. Explain. (8)
(B) Explain the different representation of fixed and floating point representation. (8)
5.Two first order LPF whose system function are given below connected in cascade.
Determine
the over all output noise power (16)
H1(z) = 1/1-0.9z-1 and H2(z) =1/1-0.8z-1
6. (a) Describe the quantization error occur in rounding and truncation in twos
complement.(8)
(b) Draw a sample and hold circuit and explains its operation? (8)
7. (a)Explain dead band in limit cycles? (8)
(b)Draw the stastical model of fixed point product quantization and explain (8)
8. (a)What is dead band of a filter? Derive the dead band of second order linear filter? (12)
(b)Consider all pole second order IIR filter described by equation y(n) = -0.5 y(n-1) – 0.75
y(n-2) + x(n).Assuming 8 bits to represent pole, determine the dead band region governing
the
limit cycle. (4)
9. Determine the variance of the round off noise at the output of two cascaded of the filter
with
system function H(z) =H1(z) .H2(z) where H1(z) =1 / 1-0.5 z-1 H2(z) = 1 / 1-0.25z-1 (16)
10. Explain with suitable examples the truncation and rounding off errors (16)
11 .a) Explain the application of DSP in Speech processing? (8)
b) What is a vocoder? Explain with a block diagram? (8)
12. Determine the dead band of the filter of y(n) = 0.95 y(n-1) +x(n) (16)
21. KNOWN GAPS, IF ANY AND INCLUSION OF THE SAME IN LECTURE SCHEDULE:
Known Gaps:
1. Comparison of received signal with the reference signal (not just by cross
correlation but by using time shift parameter).
References
AKAIKE, H. 1969. “Power Spectrum Estimation Through Autoregression Model Fitting,” Ann.
Inst. Stat. Math., Vol. 21, pp. 407-149
BARTLETT, M. S. 1961. Stochastic Processes, Cambridge University Press,
London, UK
DYM, H., and McKean , H.P.1972. Fourier Series and Integrals, Academic, New York
WEBSITES:
1.www.pearsoned.co.in/johngproakis
2.www.google.com
3.www.wikipedia.com
4.www.analogdevices.com
5.www.dspguru.com
AKKENAPALLY
1 12R11A04C1 KIRAN KUMAR 34 12R11A04F4 MOHAMMAD GOUSEPASHA
ALLA BALA
MURALI
2 12R11A04C2 KRISHNA 35 12R11A04F5 MORA SANDEEP REDDY
ANKIT
3 12R11A04C3 AGARWAL 36 12R11A04F6 MORABOINA ASHWINI
ANNAVARAPU
VENKATA SAI NAMA VEERA VIGNESHWARA
4 12R11A04C4 KIRAN 37 12R11A04F7 SAI AKHIL KUMAR
ANUPURAM
5 12R11A04C5 NARESH 38 12R11A04F8 PABBOJU DIVYA SREE
ARUGONDA
6 12R11A04C6 SAHITHYA 39 12R11A04F9 P ASHOK RAJU
BANDI
7 12R11A04C7 SASIKALA 40 12R11A04G0 P SAIVARUN REDDY
PAPITHOTTI VIDHYA
8 12R11A04C8 B PRAVEEN 41 12R11A04G1 MADHURI
BACHALA
PRAVEEN
9 12R11A04C9 KUMAR 42 12R11A04G2 PASUPULATI JAYASHREE RAO
DEEPAK REDDY
10 12R11A04D0 B 43 12R11A04G3 PINISETTY SRI SAI SRAVANTHI
BHAVANI
11 12R11A04D1 TADAM 44 12R11A04G4 POKKUNURI SAI RAM SUDEEP
CHERUKUPALLE
12 12R11A04D2 TANMAI 45 12R11A04G5 PONUGOTU VISHAL
DARSHI
STEPHEN DEVA
13 12R11A04D3 RAJ 46 12R11A04G6 PRIYATHAM REDDY M
DORNALA VIJAY
14 12R11A04D4 BHARGAV 47 12R11A04G7 R KAVYA
DOSAPATI
15 12R11A04D5 MOULIKA 48 12R11A04G8 RAMANI PRIYA KOPALLEY
AKKENAPALLY KIRAN
1 12R11A04C1 KUMAR 34 12R11A04F4 MOHAMMAD GOUSEPASHA
ALLA BALA MURALI
2 12R11A04C2 KRISHNA 35 12R11A04F5 MORA SANDEEP REDDY
GROUP 2 GROUP 8
GROUP 3 GROUP 9
DARSHI STEPHEN
13 12R11A04D3 DEVA RAJ 46 12R11A04G6 PRIYATHAM REDDY M
DORNALA VIJAY
14 12R11A04D4 BHARGAV 47 12R11A04G7 R KAVYA
GROUP 4 GROUP 10
GROUP5 GROUP11
GROUP6
31 12R11A04F1 MADAS SUDHEERA 64 13R15A0414 NYALAM LAXMAN