DSP - Ece - 5th Sem (2mark Q&A)
DSP - Ece - 5th Sem (2mark Q&A)
DSP - Ece - 5th Sem (2mark Q&A)
Question Bank
Anna University, Chennai.
Prepared by,
Prof. U. Vinothkumar, AP/ECE/Dr.N.G.P.IT
UNIT - 1
DISCRETE FOURIER TRANSFORM
Syllabus:
Discrete Signals and Systems- A Review Introduction to DFT Properties
of DFT Circular Convolution Filtering methods based on DFT FFT
Algorithms Decimation in time Algorithms, Decimation in frequency Algorithms
Use of FFT in Linear Filtering.
Two mark questions:
1. Define Signal.
Signal is a physical quantity that varies with respect to time, space or any
other independent variable.
(Or)
It is a mathematical representation of the system
Eg y(t) = t. and x(t)= sin t.
2. Define system.
A set of components that are connected together to perform the particular
task. E.g. Filters
(Or)
A System is defined as a physical device that generates a response or an output
signal, for a given input signal.
3. State the classification of discrete time signals.
The types of discrete time signals are
* Energy and power signals
* Periodic and A periodic signals
* Symmetric (Even) and Ant symmetric (Odd) signals
4. State the classification of discrete time system.
They types of discrete time systems are
* Static and Dynamic systems
* Causal and non-causal systems
* Linear and non-linear systems
* Time variant and time in-variant systems
2.
DFT
DTFT
Continuous function of
UNIT - 2
IIR Filter Design
Syllabus:
Structures of IIR Analog filter design Discrete time IIR filter from analog
filter IIR filter design by Impulse Invariance, Bilinear transformation,
Approximation of derivatives (LPF, HPF, BPF, BRF) filter design using
frequency translation.
Two mark questions:
1. Define IIR filter?
IIR filter has Infinite Impulse Response.
2. What are the various methods to design IIR filters?
* Approximation of derivatives
* Impulse invariance
* Bilinear transformation.
3. Which of the methods do you prefer for designing IIR filters? Why?
Bilinear transformation is best method to design IIR filter, since there is no
aliasing in it.
4. What is the main problem of bilinear transformation?
Frequency warping or nonlinear relationship is the main problem of bilinear
transformation.
5. What is pre-warping?
Pre-warping is the method of introducing nonlinearly in frequency
relationship to compensate warping effect.
6. Why an impulse invariant transformation is not considered to be one-toone?
In impulse invariant transformation any strip of width 2/T in the s-plane for
values of s-plane in the range (2k-1)/T (2k-1) /T is mapped into the entire
z-plane. The left half of each strip in s-plane is mapped into the interior of unit
circle in z-plane, right half of each strip in s-plane is mapped into the exterior of
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
unit circle in z-plane and the imaginary axis of each strip in s-plane is mapped on
the unit circle in z-plane. Hence the impulse invariant transformation is many-toone.
7. What is Bi-linear transformation?
The bilinear transformation is conformal mapping that transforms the splane to z-plane. In this mapping the imaginary axis of s-plane is mapped into the
unit circle in z-plane, the left half of s-plane is mapped into interior of unit circle in
z-plane and the right half of s-plane is mapped into exterior of unit circle in zplane. The Bilinear mapping is a one-to-one mapping and it is accomplished.
8. How the order of the filter affects the frequency response of Butterworth
filter.
The magnitude response of butterworth filter is shown in figure, from which
it can be observed that the magnitude response approaches the ideal response as the
order of the filter is increased.
9. What is the importance of poles in filter design?
The stability of a filter is related to the location of the poles. For a stable
analog filter the poles should lie on the left half of s-plane. For a stable digital filter
the poles should lie inside the unit circle in the z-plane.
10. How analog poles are mapped to digital poles in impulse invariant
transformation?
In impulse invariant transformation the mapping of analog to digital poles
are as follows,
* The analog poles on the left half of s-plane are mapped into the interior of
unit circle in z-plane.
* The analog poles on the imaginary axis of s-plane are mapped into the unit
circle in the z-plane.
* The analog poles on the right half of s-plane are mapped into the exterior
of unit circle in z-plane.
11.What is impulse invariant transformation?
The transformation of analog filter to digital filter without modifying the
impulse response of the filter is called impulse invariant transformation.
12.Where the j axis of s-plane is mapped in z-plane in bilinear
transformation?
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
Analog filter
i)
ii)
iii)
iv)
UNIT - 3
FIR Filter Design
Syllabus:
Structures of FIR Linear phase FIR filter Fourier series - Filter design
using windowing techniques (Rectangular Window, Hamming Window, Hanning
Window), Frequency sampling techniques Finite word length effects in digital
Filters: Errors, Limit Cycle, Noise Power Spectrum.
Two mark questions:
1. What is FIR filters?
The specifications of the desired filter will be given in terms of ideal
frequency response Hd(w). The impulse response hd(n) of the desired filter can be
obtained by inverse fourier transform of Hd(w), which consists of infinite samples.
The filters designed by selecting finite number of samples of impulse response are
called FIR filters.
2. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types 1. IIR filter 2. FIR filter
The IIR filters are of recursive type, whereby the present output sample
depends on the present input, past input samples and output samples.
The FIR filters are of non- recursive type, whereby the present output
sample depends on the present input, and previous output samples.
3. What are the different types of filter based on frequency response?
The filters can be classified based on frequency response. They are,
i)
Low pass filter
ii)
High pass filter
iii) Band pass filter
iv) Band reject filter.
4. What are the techniques of designing FIR filters?
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There are three well-known methods for designing FIR filters with linear
phase. These are 1) windows method 2) Frequency sampling method 3) Optimal or
mini-max design.
5. What is the reason that FIR filter is always stable?
FIR filter is always stable because all its poles are at origin.
6. What are the properties of FIR filter?
1. FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.
7. Write the steps involved in FIR filter design.
Choose the desired (ideal) frequency response Hd(w).
Take inverse fourier transform of Hd(w) to get hd(n).
Convert the infinite duration hd(n) to finite duration h(n).
Take Z-transform of h(n) to get the transfer function H(z) of the FIR
filter.
8. What are the advantages of FIR filters?
Linear phase FIR filter can be easily designed.
Efficient realization of FIR filter exist as both recursive and nonrecursive structures.
FIR filters realized non-recursively are always stable.
The round-off noise can be made small in non-recursive realization of
FIR filters.
9. What are the disadvantages of FIR filters?
The duration of impulse response should be large to realize sharp cutoff
filters.
The non-integral delay can lead to problems in some signal processing
applications.
10. What is the necessary and sufficient condition for the linear phase
characteristic of an FIR filter?
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The necessary and sufficient condition for the linear phase characteristic of
an FIR filter is that the phase function should be a linear function of w, which in
turn requires constant phase and group delay.
11. When cascade form realization is preferred in FIR filters?
The cascade form realization is preferred when complex zeros with absolute
magnitude less than one.
12.What are the conditions to be satisfied for constant phase delay in linear
phase FIR filters?
The conditions for constant phase delay are
Phase delay, = (N-1)/2 (i.e., phase delay is constant)
Impulse response, h(n) = -h(N-1-n) (i.e., impulse response is
antisymmetric)
13. How constant group delay & phase delay is achieved in linear phase FIR
filters?
The following conditions have to be satisfied to achieve constant group
delay & phase delay. Phase delay, = (N-1)/2 (i.e., phase delay is constant) Group
delay, = /2 (i.e., group delay is constant) Impulse response, h(n) = -h(N-1-n)
(i.e., impulse response is antisymmetric)
14.What are the possible types of impulse response for linear phase FIR
filters?
There are four types of impulse response for linear phase FIR filters
Symmetric impulse response when N is odd.
Symmetric impulse response when N is even.
Antisymmetric impulse response when N is odd.
Antisymmetric impulse response when N is even.
15. List the well-known design techniques of linear phase FIR filters.
There are three well-known design techniques of linear phase FIR filters. They
are
Fourier series method and window method
Frequency sampling method.
Optimal filter design methods.
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
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21. List the features of FIR filter designed using rectangular window.
The width of the transition region is related to the width of the main lobe
of window spectrum.
Gibbs oscillations are noticed in the passband and stopband.
The attenuation in the stopband is constant and cannot be varied.
22. Write the characteristic features of hanning window spectrum.
The main lobe width is equal to 8/N.
The maximum side lobe magnitude is 41dB.
The side lobe magnitude remains constant for increasing w.
UNIT - 4
FINITE WORDLENGTH EFFECTS
Syllabus:
Fixed point and floating point number representations ADC
Quantization- Truncation and Rounding errors -Quantization noise coefficient
quantization error Product quantization error - Overflow error Round-off noise
power - limit cycle oscillations due to product round off and overflow errors
Principle of scaling
Two mark questions:
1. What do finite word length effects mean?
The effects due to finite precision representation of numbers in a digital
system are called finite word length effects.
2. List some of the finite word length effects in digital filters.
Errors due to quantization of input data.
Errors due to quantization of filter co-efficient
Errors due to rounding the product in multiplications
Limit cycles due to product quantization and overflow in addition.
3. What are the different formats of fixed-point representation?
Sign magnitude format
Ones Complement format
Twos Complement format.
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
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In all the three formats, the positive number is same but they differ only in
representing negative numbers.
4. Explain the floating-point representation of binary number.
The floating-point number will have a mantissa part. In a given word size
the bits allotted for mantissa and exponent are fixed. The mantissa is used to
represent a binary fraction number and the exponent is a positive or negative
binary integer. The value of the exponent can be adjusted to move the position of
binary point in mantissa. Hence this representation is called floating point.
5. What are the types of arithmetic used in digital computers?
The floating point arithmetic and twos complement arithmetic are the two
types of arithmetic employed in digital systems.
6. What is truncation?
The truncation is the process of reducing the size of binary number by
discarding all bits less significant than the least significant bit that is retained. In
truncation of a binary number of b bits all the less significant bits beyond b th bit are
discarded.
7. What is rounding?
Rounding is the process of reducing the size of a binary number to finite
word size of b-bits such that, the rounded b-bit number is closest to the original unquantized number.
8. Explain the process of upward rounding?
In upward rounding of a number of b-bits, first the number is truncated to bbits by retaining the most significant b-bits. If the bit next to the least significant
bit that is retained is zero, then zero is added to the least significant bit of the
truncated number. If the bit next to the least significant bit that is retained is one
then one is added to the least significant bit of the truncated number.
9. What are the errors generated by A/D process?
The A/D process generates two types of errors. They are quantization error
and saturation error. The quantization error is due to representation of the sampled
signal by a fixed number of digital levels. The saturation errors occur when the
analog signal exceeds the dynamic range of A/D converter.
10. What is quantization step size?
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
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In digital systems, the numbers are represented in binary. With b-bit binary
we can generate 2b different binary codes. Any range of analog value to be
represented in binary should be divided into 2 b levels with equal increment. The 2b
levels are called quantization levels and the increment in each level is called
quantization step size. If R is the range of analog signal then, Quantization step
size, q = R/2b
11. How the digital filter is affected by quantization of filter coefficients?
The quantization of the filter coefficients will modify the value of poles &
zeros and so the location of poles and zeros will be shifted from the desired
location. This will create deviations in the frequency response of the system. Hence
the resultant filter will have a frequency response different from that of the filter
with un-quantized coefficients.
12. What is meant by product quantization error?
In digital computations, the output of multipliers i.e., the product are
quantized to finite word length in order to store them in registers and to be used in
subsequent calculations. The error due to the quantization of the output of
multiplier is referred to as product quantization error.
13. Why rounding is preferred for quantizing the product?
In digital system rounding due to the following desirable characteristic of
rounding performs the product quantization
The rounding error is independent of the type of arithmetic
The mean value of rounding error signal is zero.
The variance of the rounding error signal is least.
14. What are limit cycles?
In recursive systems when the input is zero or some nonzero constant value,
the nonlinearities die to finite precision arithmetic operations may cause periodic
oscillations in the output. These oscillations are called limit cycles.
15. What is zero input limit cycles?
In recursive system, the product quantization may create periodic
oscillations in the output. These oscillations are called limit cycles. If the system
output enters a limit cycles, it will continue to remain in limit cycles even when the
input is made zero. Hence these limit cycles are also called zero input limit cycles.
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Sign bit
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UNIT - 5
DSP APPLICATIONS
Syllabus:
Multi-rate signal processing: Decimation, Interpolation, Sampling rate
conversion by a rational factor Adaptive Filters: Introduction, Applications of
adaptive filtering to equalization.
Two mark questions:
1. What is multi-rate signal processing?
The theory of processing signals at different sampling rates is called multirate signal processing.
2. Define down sampling.
Down sampling a sequence x(n) by a factor M is the process of picking
every Mth sample and discarding the rest.
3. What is mean by up-sampling?
Up-sampling by a factor L is the process of inserting L-1 zeros between two
consecutive samples.
4. If the spectrum of sequence x(n) is X(ejw), then what is the spectrum of a
signal down-sampled by factor 2?
Y(ejw)=(1/2)[X(ejw/2)+ X(ejw((w/2)-)]
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
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Y(z)= (1/M)
X
k=0
(z(1/M)e(-j2k/M))
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