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Question Bank
Anna University, Chennai.
Prepared by,
Prof. U. Vinothkumar, AP/ECE/Dr.N.G.P.IT
UNIT - 1
SIGNALS AND SYSTEMS
Syllabus:
Basic elements of DSP concepts of frequency in Analog and Digital Signals
sampling theorem Discrete time signals, systems Analysis of discrete time LTI
systems Z transform Convolution (linear and circular) Correlation.
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.
(APR-96)
The signal that are defined at discrete instants of time are known as discretetime signals. The discrete-time signals are continuous in amplitude and discrete in
time. They are denoted by x(n).
7. Give some applications of DSP?
(APR-98)
* Speech processing Speech compression & decompression for voice
storage system
* Communication Elimination of noise by filtering and echo cancellation.
* Bio-Medical Spectrum analysis of ECG, EEG etc.
A continuous time signal can be represented in its samples and recovered back
if the sampling frequency Fs 2B. Here Fs is the sampling frequency and B is
the maximum frequency present in the signal.
9. What are the properties of convolution?
(MAR-99)
signal in
Analog to Digital converter
Digital Signal processor
Digital to Analog converter
signal out
Veracity
Simplicity
Repeatability
13. What are the major classifications of the signal?
(APR-98)
Discrete time signals are defined only at discrete times, and for these signals,
the independent variable takes on only a discrete set of values.
Classification of discrete time signal:
1. Periodic and A periodic signal
2. Even and Odd signal
15. Define continuous time signals and classify them.
(MAY-2000)
A signal is said to be energy signal if it have finite energy and zero power. A
signal is said to be power signal if it have infinite energy and finite power. If the
Above two conditions are not satisfied then the signal is said to be neither energy
nor power signal.
18. What is analog signal?
The digital signal is same as discrete signal except that the magnitude of signal
is quantized.
20. Define periodic and non-periodic discrete time signals?
If the discrete time signal repeated after equal samples of time then it is called
periodic signal. When the discrete time signal x[n] satisfies the condition
x[n+N]=x(n), then it is called periodic signal with fundamental period N samples.
If x(n) * x(n+N) then it is called non periodic signals.
21. What are all the blocks are used to represent the CT signals by its samples?
Sampler
Quantizer
22. Define sampling process.
(APR-98,2000)
The sampling frequency must be at least twice the maximum frequency
present in the signal.
That is Fs = > 2fm
Where,
Fs = sampling frequency
fm = maximum frequency
24. Define aliasing or folding.
To avoid the aliasing effect the sampling frequency must be twice the
maximum frequency present in the signal.
26. Define z transform?
(APR-2002)
( )z-n
The region of convergence (ROC) is defined as the set of all values of z for
Which X(z) converges.
28. What are the properties of ROC?
(MAY-2004)
Z-transform is used for analysis the both periodic and a periodic signals.
32. State the convolution properties of Z transform?
(MAY-98,2002)
The convolution property states that the convolution of two sequences in
time domain is equivalent to multiplication of their Z transforms.
33. What are the conditions of stability of a causal system?
All the poles of the system are within the unit circle. The sum of impulse
response for all values of n is bounded.
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
(MAY-2006)
UNIT - 2
FREQUENCY TRANSFORMATIONS
Syllabus:
Introduction to DFT Properties of DFT Filtering methods based on DFT
FFT Algorithms Decimation in time Algorithms, Decimation in frequency
Algorithms Use of FFT in Linear Filtering DCT.
Two mark questions:
1. Define DFT.
(APR-2006)
It is a finite duration discrete frequency sequence, which is obtained by
sampling one period of Fourier transform. Sampling is done at N equally spaced
points over the period extending from w=0 to 2.
DFT is defined as X(w)= x(n)e-jwn. Here x(n) is the discrete time sequence
X(w) is the fourier transform of x(n).
2. Define Twiddle factor.
The Twiddle factor is defined as WN=e-j2 /N
3. Define Zero padding.
The method of appending zero in the given sequence is called as Zero padding.
4. State circular convolution.
(MAY-2004)
This property states that multiplication of two DFT is equal to circular
convolution of their sequence in time domain.
5. State parsevals theorem.
Consider the complex valued sequences x(n) and y(n).If x(n)y*(n)=1/N
X(k)Y*(k)
6. List the properties of DFT.
Linearity, Periodicity, Circular symmetry, symmetry, Time shift, Frequency
shift, complex conjugate, convolution, correlation and Parsevals theorem.
7. What is the disadvantage of direct computation of DFT?
For the computation of N-point DFT, N2 complex multiplications and
N[N-1] Complex additions are required. If the value of N is large than the number
of computations will go into lakhs. This proves inefficiency of direct DFT
computation.
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
10
13. What are the differences and similarities between DIF and DIT algorithms?
Differences: 1. For DIT, the input is bit reversal while the output is in natural
order, whereas for DIF, the input is in natural order while the output is bit reversed.
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 in DIF.
Similarities: Both algorithms require same number of operations to compute
the DFT. Bot algorithms can be done in place and both need to perform bit reversal
at some place during the computation.
14. What is a decimation-in-frequency algorithm?
(MAR-2006)
In this the output sequence X (K) is divided into two N/2 point sequences and
each N/2 point sequences are in turn divided into two N/4 point sequences.
15. Distinguish between DFT and DTFT.
S.No.
DFT
DTFT
1.
Obtained by performing sampling Sampling is performed only in time
operation in both the time and domain.
frequency domains.
2.
Continuous function of
11
( )
12
W kN
Xm (q)q
Xm(p)+Wmk Xm(q)
Xm(q)q
W
Xm(q)q
Xm(q)]WNk
Xm(p)+Xm(q)
Xm(p)-WnkXm(q)
[Xm(p)-
Xm+1(q)=|Xm(p)-Xm(q)|WNk
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UNIT - 3
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 (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?
(MAY-2009)
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 zplane. 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 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-to-one.
14
15
(NOV-2010)
Digital filter
i)
ii)
iii)
iv)
Analog filter
i)
ii)
iii)
iv)
16
18. What are the requirements for an analog filters to be stable and causal?
i.
The analog filter transfer function H(s) should be a rational function of s
and the coefficients of s should be real.
ii.
The poles should lie on the left half of s-plane.
iii.
The number of zeros should be less than or equal to number of poles.
19. Distinguish between IIR and FIR filters.
(NOV-2010)
The filter design starts from ideal frequency response. By taking inverse
fourier transform of ideal frequency response, the desired impulse response is
obtained, which consists of infinite number of samples.
The digital filter design by selecting only N samples of the impulse response
are called FIR filters. The digital filters designed by considering all the infinite
samples of impulse response are called IIR filters.
20. Compare IIR and FIR filters.
IIR Filter
i.
All the infinite samples of
impulse
response
are
considered.
ii.
The impulse response cannot
be directly converted to digital
filter transfer function.
iii.
The design involves design of
analog filter and then
transforming analog filter to
digital filter.
iv.
The specifications include the
desired characteristics for
magnitude response only.
v.
Linear phase characteristics
cannot be achieved.
i.
ii.
iii.
iv.
v.
FIR Filter
Only N samples of impulse
response are considered.
The impulse response can be
directly converted to digital
filter transfer function.
The digital filter can be
directly designed to achieve
the desired specification.
The specifications include the
desired characteristics for
both magnitude and phase
response.
Linear phase filter can be
easily designed.
17
22. Mention any two techniques for digitizing the transfer function of an analog
filter.
The bilinear transformation and the impulse invariant transformation are the
two techniques available for digitizing the analog filter transfer function.
23. What are the properties that are maintained same in the transformation of
analog to digital filer?
The analog filter should be stable and causal for effective transformation to
digital filters. While transforming the analog filer to digital filters these two
properties (i.e. stability and causality) are maintained same, which means that the
transformed digital filer should also be stable and causal.
24. What is aliasing?
(MAY-2012)
The phenomena of high frequency sinusoidal components acquiring the
identity of low frequency sinusoidal components after sampling is called aliasing.
The aliasing problem will arise if the sampling rate does not satisfy the Nyquist
sampling criteria.
25. What is frequency warping?
In bilinear transformation the relation between analog and digital frequencies
is non-linear. When the s-plane is mapped in to z-plane using bilinear
transformation, this non-linear relationship introduce distortion in frequency axis,
which called frequency warping.
26. What is butterworth approximation?
In butterworth approximation, the error function is selected such that the
magnitude is maximally flat in the origin (i.e., at = 0) and monotonically
decreasing with increasing .
27. How the poles of butterworth transfer function are located in s-plane?
The poles of the normalized butterworth transfer function symmetrically lies
on a unit circle in s-plane with angular spacing of /N.
28. What is the properties of butterworth filter?
(APR-2014)
i.
The butterworth filters are pole design.
ii.
At the cutoff frequency c, the magnitude of normalized butterworth filter
is 1/2.
iii.
The filter order N, completely specifies the filter and as the value of N
increases the magnitude response approaches the ideal response.
iv.
The magnitude is maximally flat at the origin and monotonically
decreasing with increasing .
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
18
19
UNIT - 4
FIR FILTER DESIGN
Syllabus:
Structures of FIR Linear phase FIR filter Filter design using windowing
techniques, Frequency sampling techniques Finite word length effects in digital
Filters
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?
(APR-2012)
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.
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21
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.
16. What are the desirable characteristics of the frequency response of
window function?
(NOV-2007)
The desirable characteristics of the frequency response of window function are
The width of the main lobe should be small and it should contain as much
of the total energy as possible.
The side lobes should decrease in energy rapidly as w tends to .
17. What is Gibbs phenomenon (or Gibbs Oscillation)?
(APR-2012)
In FIR filter design by Fourier series method the infinite duration impulse
response is truncated to finite duration impulse response. The abrupt truncation of
impulse response introduces oscillations in the passband and stopband. This effect
is known as Gibbs phenomenon (or Gibbs Oscillation).
Prepared by Prof. U. Vinothkumar AP / ECE/ Dr.NGPIT
22
18. Write the procedure for designing FIR filter using frequency-sampling
method.
Choose the desired (ideal) frequency response Hd(w).
Take N-samples of Hd(w) to generate the sequence
Take inverse DFT of to get the impulse response h(n).
The transfer function H(z) of the filter is obtained by taking z-transform
of impulse response.
19. What are the drawback in FIR filter design using windows and frequency
sampling method? How it is overcome?
(APR-2013)
The FIR filter design using windows and frequency sampling method does
not have Precise control over the critical frequencies such as wp and ws. This
drawback can be overcome by designing FIR filter using Chebyshev approximation
technique. In this technique an error function is used to approximate the ideal
frequency response, in order to satisfy the desired specifications.
20. Write the characteristic features of rectangular window.
The main lobe width is equal to 4/N.
The maximum side lobe magnitude is 13dB.
The side lobe magnitude does not decrease significantly with increasing w.
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.
23. 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.
23
24
25
26
UNIT - 5
APPLICATIONS
Syllabus:
Multi-rate signal processing Speech compression Adaptive filter
Musical sound processing Image enhancement.
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.
(NOV-2012)
Down sampling a sequence x(n) by a factor M is the process of picking every
th
M 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)-)]
5. If the Z-transform of a sequence x(n) is X(z) then what is the Z-transform of
a sequence down-sampled by a factor M?
Y(z)= (1/M)
(z(1/M)e(-j2k/M))
6. If the z-transform of a sequence x(n) is X(z) then what is the z-transform of
a sequence up-sampled by a factor L?
(DEC-2012)
L
Y(z)= X(z )
7. What is the need for anti-imaging filter after up-sampling a signal?
The frequency spectrum of up-sampled signal with a factor L, contains (L-1)
additional images of the input spectrum. Since we are not interested in image spectra,
a low-pass filter with a cutoff frequency wc = (/L) can be used after up-sampler.
This filter is known as anti-imaging filter.
27
28
(NOV-2012)
y(n)
v(n)
20. Draw the frequency domain representation of downsampler. (DEC-2010)
x(n)
y(n) = x(Dn)
D
x(ejw)
Y(ejw)=[1/D] x(ejw/D)
29
23. Draw the Multirate signal processing system with analysis and synthesis
filter banks.
30
yD(n)
yD(n)= x(Mn)
For an input sequence x(n), select only the samples which occur at integer
multiples of M. The other samples are thrown away.
Aliasing will occur in yD(n) unless x(n) is sufficiently bandlimited loss of
information.
28. Draw the structure of L-folder expander.
(DEC-2005)
For an input sequence x(n), insert L 1 zeros between each sample.
x(n) can always be recovered from yE(n) no loss of information, no aliasing.
31
X(n)
yE(n)
YE(n)= x(Mn)
29. Develop an expression for the output y(n) as a function of the input x(n)
for the multirate structure of below fig.
(APR-2014)