Integer Cat Swarm Optimization Algorithm for Multiobjective

Integer Problems
Shahid Ali Murtza
ICT R&D Fund
Taxila,Pakistan

Ali Murtza Shahid

ICT R&D Fund
Taxila,Pakistan

Murtza Shahid Ali

ICT R&D Fund
Taxila,Pakistan

Sensitivity analysis

1.1

DSP

Q1.- Classify signals.

Ans1. Continuous-time, continuous amplitude (Analog Signals)
Discrete time, continuous amplitude
Continuous time, discrete amplitude
Discrete-time, discrete-amplitude Q2.-What is the use of Random Signals?

Ans2. Random signals are used to test dynamic response statistically for very small amplitudes and timeduration.
Q3.- Classify Systems.
Ans3. Linear, stable and time-invariant.
Q4.-What do you mean by aliasing in digital signal processing?
How it can be avoided?
Ans4. Aliasing refers to an effect due to which different signals become indistinguishable. It also refers to
distortion in the reconstructed signal when it is reconstructed from the original continuous signal.
To avoid aliasing we can simply filter out the high frequency components of the signal by using anti-aliasing
filter like optical anti-aliasing filter.
Q5. What are the differences between a microprocessor and a DSP processor?
Ans5. DSP processors are featured to support high performance and repeatitive and intensive tasks whereas
microprocessors are not application specific and they are designed to process control-oriented tasks.
Q6. What is the convolution?
Ans6. Convolution is the technique of adding two signals in time domain. We can also do this quite easily by
changing the domain of signals from time domain to frequency domain using Fast Fourier Transform (FFT).
Q7.- What is FFT?
Ans7. FFT is a fast way to calculate Discrete Fourier Transform (DFT). It is much more efficient then DFT
and require less number of coding lines. Due to FFT several kind of techniques are feasible.
Q8.- What is the advantage of a Direct form II FIR over fom I?
Ans8. Direct Form II FIR filters requires half the number of delay units as much as used by Form I.
Q9.- What is interpolation and decimation?

Ans9. Interpolation is the process of increasing the sample rate in dsp whereas decimation is the opposite of
this that is, it is the process of decreasing the sample rate in dsp.
10.- Difference between DFT and DTFT.
Ans10.
DFT DTFT 1-Limited number of samples of periodic signal 1-unlimited number of samples. 2- input is always
periodic 2-input may not always be periodic 3- physically realizable 3- mathematically precise 4- frequency
becomes discrete 4- frequency is continuous
Digital Signal Processing (DSP) Interview Questions and Answers This page contains the collection of Digital
Signal Processing (DSP) Interview Questions and Answers / Frequently Asked Questions (FAQs) under category
Computer Programming. These questions are collected from various resources like informative websites, forums,
blogs, discussion boards including MSDN and Wikipedia. These listed questions can surely help in preparing for
Digital Signal Processing (DSP) interview or job. Please write a code in C / Verilog to implement a basic FIR
filter?
disp(choose the window from the list); ch=menu(types of windows,bartlett,blackman,hamming,hanning,kaiser,
rectangular); rp=input(enter the passband ripple in db); rs=input(enter the stopband ripple in db); wsample=input(enter sampling frequency in hertz); wp=input(enter the passband frequency in hertz); ws=input(enter
the stopband frequency in hertz); wp=2*wp/wsample; ws=2*ws/wsample; p=20*log10(sqrt(rp*rs))-13; q=14.6*(wswp)/wsample; N=1+floor(p/q); N1=N; if(rem(N,2)==0) N1=N+1; else N=N-1; end switch ch case 1 y=bartlett(N1);
case 2 y=blackman(N1); case 3 y=hamming(N1); case 4 y=hanning(N1); case 5 beta=input(enter beta for
kaiser window); y=kaiser(N1,beta); case 6 y=boxcar(N1); otherwise disp(enter proper window number); end
disp(select the type of filter from the list); type=menu(types of filters,lowpass,highpass,bandpass,bandstop);
switch type case 1 b=fir1(N,wp,low,y); case 2 b=fir1(N,wp,high,y); case 3 b=fir1(N,[wp ws],bandpass,y); case
4 b=fir1(N,[wp ws],stop,y); otherwise disp(enter type number properly); end [h,w]=freqz(b,1,512); magn=20*log10(abs(h));
phase=(180/pi)*unwrap(angle(h)); w=(w*wsample)/(2*pi); subplot(2,1,1); plot(w,magn),grid on;title(magnitude
plot); subplot(2,1,2); plot(w,phase),grid on;title(phase plot); Do you know How is the non-periodic nature of
the input signal handled?
Fourier series is applied for periodic signals since they violate Dirchilets conditions. This will give the fundamental
and harmonic signal components for periodic signals.
For non-periodic signals if we need frequency analysis as a whole then fourier transform is applied for the entire
duration. Provided its energy is finite and follows other conditions as laid out by Dirchilet. What is the difference
between ProtoPlus and ProtoPlus Lite?
ProtoPlus prototyping daughter card - A plug-in, 2-connector, multi-layer, low noise, and stackable prototyping
board that plugs into the Texas Instruments DSK and EVM DSP development systems.
ProtoPlus Lite prototyping daughter card - A Low cost, 2-connector, plug-in prototyping board that plugs into
the Texas Instruments DSK and EVM DSP development systems. Can we create a table with out primary key?
yes we can create CREATE TABLE Orders ( OrderID SMALLINT UNSIGNED NOT NULL PRIMARY KEY,
ModelID SMALLINT UNSIGNED NOT NULL, ModelDescrip ); Explain what is dirac delta function and its
fourier transform and its importance?
Dirac delta is a continuous time function with unit area and infinite amplitude at t=0. the fourier transform of
dirac delta is 1. using dirac delta as an input to the system, we can get the system respnose. it is used to study
the behavior of the circuit. we can use this system behavior to find the output for any input. Suppose we are
sending address of thesalve and then data then after i want to read the data which i was sent recently, in that
case before im reading is there any need to send a stop bit before read?
Before reading the data if you are giving the stop bit then the communication is stopped.so after sending the
data you will give the stop bit. How do we implement a fourth order Butterworth LP filter at 1kHz if sampling
frequency is 8 kHz?
A fourth order Butterworth filter can be made as cascade of two seond order LP filters with zeta of 0.924 and
0.383. One can use a bilinear transformation approach for realising second order LP filters. Using this technique
described well in many texts, one can make two second order LP filters and cascade them. What is an anti
aliasing filter and why is it required?
Anti aliasing filter reduces errors due to aliasing. If a signal is sampled at 8 kS/S, the max frequency of the input

should be 4 kHz. Otherwise, aliasing errors will result. Typically a 3.4kHz will have an image of 4.6 khz, and one
uses a sharp cut off filter with gain of about 1 at 3.4kHz and gain of about 0.01 at 4.6 kHz to effectively guard
against aliasing. Thus one does not quite choose max frequency as simply fs/2 where fs is sampling frequency. One
has to have a guard band of about 10chooses max signal frequency as 0.9*fs/2 Explain Is the Gibbs phenomenon
ever a factor?
Yes Gibbs phenomenon becomes constraining when we are analysing signals containing frequency tones quite
close to each other. If the side lobes of the windowing function are significant then it leads to energy leakages
between the frequency bins/sub-bands. Thus very close lying frenecy tones gets their magnitudes smeared up in
the process.
1.Please write a code in C / Verilog to implement a basic FIR filter?
2.Do you know How is the non-periodic nature of the input signal handled?
3.What is the difference between ProtoPlus and ProtoPlus Lite?
4.Can we create a table with out primary key?
5.Explain what is dirac delta function and its fourier transform and its importance?
6.Suppose we are sending address of thesalve and then data then after i want to read the data which i was sent
recently, in that case before im reading is there any need to send a stop bit before read?
7.How do we implement a fourth order Butterworth LP filter at 1kHz if sampling frequency is 8 kHz?
8.What is an anti aliasing filter and why is it required?
9.Explain Is the Gibbs phenomenon ever a factor?
10.Please write a code in C / Verilog to implement a basic FIR filter?
11.Do you know How is the non-periodic nature of the input signal handled?
12.What is the difference between ProtoPlus and ProtoPlus Lite?
13.Can we create a table with out primary key?
14.Explain what is dirac delta function and its fourier transform and its importance?
15.Suppose we are sending address of thesalve and then data then after i want to read the data which i was sent
recently, in that case before im reading is there any need to send a stop bit before read?
16.How do we implement a fourth order Butterworth LP filter at 1kHz if sampling frequency is 8 kHz?
17.What is an anti aliasing filter and why is it required?
18.Explain Is the Gibbs phenomenon ever a factor?
Q. Why should we go for digital signal processing where as the most of the real world data is in analog mode?
What are the differences between a microprocessor and a DSP processor?
What is the convolution?
Why do we need Forrier transform in DSP?
What is use of windowing in digital filters?
Tell some thing about Interpolation and decimation?
What is the need of FFT ?
Whats the difference between FFT and DFT?
What is the advantage of a Direct form II FIR over fom I?
What is the difference between equiripple filter and FIR filter?
What is the application fo Cross correlation and Auto Correlation?
Explain using convolution the effects of taking an FFT of a sampe with no windowing (rectangular window).
Q. What are basis vectors in a transofrm?
Q. In signal processing, why we are much more interested in orthogonal transform?
Q. How does polyphase filtering save computations in a decimation filter?
Q. Why IIR filters doesnt have Linear phase?
Q. Whats basic difference b/w winer filter and kalman filter and lms filter Q. What is the use of windowing in
digital filters Q. What are the pros and cons of Discrete Cosine Transform?
Q. What is Interpolation and decimation filters and why we need it?
Q. What is the simplest high pass filter ?
write the equation?
Q. What is the difference between ProtoPlus and ProtoPlus Lite?
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Q. What is Auto Regressive Model?

How is the order of auto regressive model is decided?
Q. What is the basic difference between FIR and IIR filters?
Q. What two PSK modulation orders differ exactly by a factor of two in spectral efficiency?
Q. Under what conditions is the available bandwidth of a digital system Fs Hz instead of Fs/2 Hz?
Q. What is the difference between DFT and DTFT?
Q. FFT is in complex domain how to use it in real life signals optimally?
Q. What is Gibbs phenomenon?
Q. Can we create a table with out primary key?
Q. Suppose we have a system with transfer function H(z) = 1 / ((z 1.1)*(z 0.9)). Is the system stable or
unstable?
Q. Differences b/w butterworth chebyshev?
Q. How do you reduce spectral leakage?
Q. Why is FFT faster than DFT?
what is the actual concept behind this?
Q. Is the Gibbs phenomenon ever a factor?
Q. What is the concept of stability of an LTI system?
How to check if a given system is stable?
Q. How does polyphase filtering save computations in an interpolation filter?
Q. If a have two vectors how will i check the orthogonality of those vectors?
Q. Can IIR filters be Linear phase?
how to make it linear Phase?
Q. How can you compute fourier transform form Z-transform ?
Q. How is the non-periodic nature of the input signal handled?
Q. What is aliasing and how do we prevent it?
Q. How can you determine the stability of an LTI system?
Q. What is the need of Digital Signal Processing?
Q. Why do we need I&Q signals?
Q. What do you mean by spectral resolution?
Q. What is the special about minimum phase filter?
Q. How can you compute fourier transform form Z-transform ?
Q. Why after DCT we use a zig zag manner for run length coding?
Q. Why we use DCT extensilvely in compression?
Q. Can you write assembly language programs for DSP?
Q. What is your proficiency level of C-language for DSP applications?
1. Compute the DFT of x(n) = , 0 n 5. 2. Prove the following properties of DFT when X (K) is the DFT of
an N-point sequence h(n) . a) X (K) is real and even when x(n) is real and even. b) X(K) is imaginary and odd
when x(n) is real and odd. 3. Determine the IDFT of X(K) = 4. Perform the linear convolution of finite duration
sequences h(n) and x (n) = by overlap add method 5. Compute the N-point DFT of x(n) = 6. Compute the
8-point DFT of the sequence x(n) = using radix-2 DIT algorithm. 7. Compute an IDFT of the following sequence
X(K) = using DIF algorithm.
UNIT II
1. Design a chebyshev low pass filter with the specifications = 1db ripple in the pass band 0 0.2 , = 15 db
ripple in the stop band 0.3
using Bilinear transformation 2. Using bilinear transformation design a digital
band pass Butterworth filter with the following specifications Sampling frequency of 8 KHz =2 db in the pass
band 800Hz f 1000 Hz = 20 db in the stop band 0 f 400 Hz and 2000Hz f
3. Design a chebyshev low pass filter with the specifications