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Integer Cat Swarm Optimization Algorithm for Multiobjective

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

1

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?

title(’magnitude plot’). discussion boards including MSDN and Wikipedia. if(rem(N.y). 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.phase). Interpolation is the process of increasing the sample rate in dsp whereas decimation is the opposite of this that is. one can make two second order LP filters and cascade them.y). Ans10. subplot(2.’bandpass’. case 2 y=blackman(N1). case 6 y=boxcar(N1). it is used to study the behavior of the circuit.’bandpass’. For non-periodic signals if we need frequency analysis as a whole then fourier transform is applied for the entire duration.mathematically precise 4.wp.beta).6*(wswp)/wsample. multi-layer.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. Suppose we are sending address of thesalve and then data then after i want to read the data which i was sent recently. Provided its energy is finite and follows other conditions as laid out by Dirchilet. p=20*log10(sqrt(rp*rs))-13. N1=N. 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.’stop’.924 and 0.’highpass’.1.512). end [h. end disp(’select the type of filter from the list’). blogs.’bandstop’).A plug-in.A Low cost. 2-connector. ModelDescrip ).[wp ws]. end switch ch case 1 y=bartlett(N1). ’rectangular’).magn). Using this technique described well in many texts. we can use this system behavior to find the output for any input.y).’blackman’. ws=2*ws/wsample. otherwise disp(’enter proper window number’). phase=(180/pi)*unwrap(angle(h)). wsample=input(’enter sampling frequency in hertz’).Difference between DFT and DTFT. wp=2*wp/wsample. the max frequency of the input 2 . If a signal is sampled at 8 kS/S. y=kaiser(N1. we can get the system respnose. switch type case 1 b=fir1(N. otherwise disp(’enter type number properly’). 2. plot(w.1. case 4 b=fir1(N.w]=freqz(b. These listed questions can surely help in preparing for Digital Signal Processing (DSP) interview or job. N=1+floor(p/q).’lowpass’. type=menu(’types of filters’.383. ch=menu(’types of windows’. rp=input(’enter the passband ripple in db’).frequency becomes discrete 4. What is an anti aliasing filter and why is it required? Anti aliasing filter reduces errors due to aliasing. This will give the fundamental and harmonic signal components for periodic signals.wp.physically realizable 3. What is the difference between ProtoPlus and ProtoPlus Lite? ProtoPlus prototyping daughter card . These questions are collected from various resources like informative websites.Ans9.’high’. q=14.2).so after sending the data you will give the stop bit.’bartlett’. and stackable prototyping board that plugs into the Texas Instruments DSK and EVM DSP development systems. rs=input(’enter the stopband ripple in db’). case 3 b=fir1(N. case 2 b=fir1(N. wp=input(’enter the passband frequency in hertz’).grid on. the fourier transform of dirac delta is 1. forums.2)==0) N1=N+1.’kaiser’.[wp ws]. DFT DTFT 1-Limited number of samples of periodic signal 1-unlimited number of samples. low noise.’low’. ws=input(’enter the stopband frequency in hertz’). ProtoPlus Lite prototyping daughter card .1).’hanning’.y). using dirac delta as an input to the system. subplot(2. case 5 beta=input(’enter beta for kaiser window’).input is always periodic 2-input may not always be periodic 3.’hamming’. 10. 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. w=(w*wsample)/(2*pi). plug-in prototyping board that plugs into the Texas Instruments DSK and EVM DSP development systems.grid on. else N=N-1. plot(w. ModelID SMALLINT UNSIGNED NOT NULL. case 3 y=hamming(N1). Do you know How is the non-periodic nature of the input signal handled? Fourier series is applied for periodic signals since they violate Dirchilet’s conditions. magn=20*log10(abs(h)). case 4 y=hanning(N1). 2-connector. it is the process of decreasing the sample rate in dsp.title(’phase plot’). One can use a bilinear transformation approach for realising second order LP filters. Can we create a table with out primary key? yes we can create CREATE TABLE Orders ( OrderID SMALLINT UNSIGNED NOT NULL PRIMARY KEY. Please write a code in C / Verilog to implement a basic FIR filter? disp(’choose the window from the list’).1..

Q. How does polyphase filtering save computations in a decimation filter? Q. Thus very close lying frenecy tones gets their magnitudes smeared up in the process. Otherwise. 3 .How do we implement a fourth order Butterworth LP filter at 1kHz if sampling frequency is 8 kHz? 17.should be 4 kHz.Please write a code in C / Verilog to implement a basic FIR filter? 11. What is the simplest high pass filter ? write the equation? Q. in that case before im reading is there any need to send a stop bit before read? 7. and one uses a sharp cut off filter with gain of about 1 at 3. What is the use of windowing in digital filters Q. Q. Q. 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.Do you know How is the non-periodic nature of the input signal handled? 3.6 khz.4kHz and gain of about 0. What is the difference between ProtoPlus and ProtoPlus Lite? Q.Please write a code in C / Verilog to implement a basic FIR filter? 2. In signal processing.What is the difference between ProtoPlus and ProtoPlus Lite? 4. What are basis vectors in a transofrm? Q.Explain what is dirac delta function and its fourier transform and its importance? 15.01 at 4. Typically a 3.Explain Is the Gibbs phenomenon ever a factor? 10. Q.4kHz will have an image of 4. Q.How do we implement a fourth order Butterworth LP filter at 1kHz if sampling frequency is 8 kHz? 8.Suppose we are sending address of thesalve and then data then after i want to read the data which i was sent recently.Can we create a table with out primary key? 5.What is an anti aliasing filter and why is it required? 18. in that case before im reading is there any need to send a stop bit before read? 16. Q. What is Interpolation and decimation filters and why we need it? Q. Q. aliasing errors will result. One has to have a guard band of about 10chooses max signal frequency as 0. What are the pros and cons of Discrete Cosine Transform? Q. Q.6 kHz to effectively guard against aliasing.Suppose we are sending address of thesalve and then data then after i want to read the data which i was sent recently. If the side lobes of the windowing function are significant then it leads to energy leakages between the frequency bins/sub-bands. why we are much more interested in orthogonal transform? Q. Whats basic difference b/w winer filter and kalman filter and lms filter Q. 1. Why IIR filters doesnt have Linear phase? Q.What is the difference between ProtoPlus and ProtoPlus Lite? 13.Explain Is the Gibbs phenomenon ever a factor? Q. Thus one does not quite choose max frequency as simply fs/2 where fs is sampling frequency. Q.Do you know How is the non-periodic nature of the input signal handled? 12.9*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.Explain what is dirac delta function and its fourier transform and its importance? 6.What is an anti aliasing filter and why is it required? 9.Can we create a table with out primary key? 14.

Perform the linear convolution of finite duration sequences h(n) and x (n) = by overlap add method 5. Design a chebyshev low pass filter with the specifications = 1db ripple in the pass band 0 0. What two PSK modulation orders differ exactly by a factor of two in spectral efficiency? Q. Is the system stable or unstable? Q. Can you write assembly language programs for DSP? Q. Design a chebyshev low pass filter with the specifications 4 . 2. Compute the N-point DFT of x(n) = 6. What is the concept of stability of an LTI system? How to check if a given system is stable? Q. UNIT II 1.1)*(z 0. Why is FFT faster than DFT? what is the actual concept behind this? Q. = 15 db ripple in the stop band 0. What is Gibbs phenomenon? Q. Suppose we have a system with transfer function H(z) = 1 / ((z 1.3 using Bilinear transformation 2. What is the basic difference between FIR and IIR filters? Q. What is the need of Digital Signal Processing? Q. 0 n 5. 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. How does polyphase filtering save computations in an interpolation filter? Q. Differences b/w butterworth chebyshev? Q. Can IIR filters be Linear phase? how to make it linear Phase? Q. What is your proficiency level of C-language for DSP applications? 1. FFT is in complex domain how to use it in real life signals optimally? Q. What do you mean by spectral resolution? Q. How can you compute fourier transform form Z-transform ? Q. What is the difference between DFT and DTFT? Q. What is Auto Regressive Model? How is the order of auto regressive model is decided? Q. How is the non-periodic nature of the input signal handled? Q.2 . What is the special about minimum phase filter? Q. Determine the IDFT of X(K) = 4. Under what conditions is the available bandwidth of a digital system Fs Hz instead of Fs/2 Hz? Q. b) X(K) is imaginary and odd when x(n) is real and odd.9)). What is aliasing and how do we prevent it? Q. Prove the following properties of DFT when X (K) is the DFT of an N-point sequence h(n) . Why do we need I&Q signals? Q.Q. Is the Gibbs phenomenon ever a factor? Q. How do you reduce spectral leakage? Q. Compute the 8-point DFT of the sequence x(n) = using radix-2 DIT algorithm. Why we use DCT extensilvely in compression? 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. 7. Compute the DFT of x(n) = . 3. Can we create a table with out primary key? Q. How can you determine the stability of an LTI system? Q. If a have two vectors how will i check the orthogonality of those vectors? Q. a) X (K) is real and even when x(n) is real and even. Compute an IDFT of the following sequence X(K) = using DIF algorithm.