Are you interested in Digital Communications? If yes then you can take up a Digital Signal Processing job to improve the accuracy of communication in this digital world. On our wisdomjobs page, we share with you information of the skills required, training courses available and various job opportunities related to the Digital Signal Processing job. The knowledge of Digital Signal Processing can be applied in the field of telecommunication, biomedical imaging, seismology and control systems in the aerospace industry among others. The various applications of Digital Signal Processing has increased the demand for its users and has created new job opportunities for them. You can browse though this bank of job requirements available on the wisdomjobs page and read the Digital Signal Processing job interview questions and answers, that will land you with a specialized job in your hands.
Question 1. Define Discrete Time Signal?
Answer :
A discrete time signal x (n) is a function of an independent variable that is an integer. a discrete time signal is not defined at instant between two successive samples.
Question 2. Define Discrete Time System?
Answer :
A discrete or an algorithm that performs some prescribed operation on a discrete time signal is called discrete time system.
Question 3. What Are The Elementary Discrete Time Signals?
Answer :
Unit sample sequence (unit impulse)
δ (n)= {1 n=0
0 Otherwise
Unit step signal
U (n) ={ 1 n>=0
0 Otherwise
Unit ramp signal
Ur(n)={n for n>=0
0 Otherwise
Exponential signal
x (n)=an where a is real
x(n)-Real signal
Question 4. State The Classification Of Discrete Time Signals?
Answer :
The types of discrete time signals are:
Question 5. Define Periodic And Aperiodic Signal?
Answer :
A signal x (n) is periodic in period N, if x (n+N) =x (n) for all n. If a signal does not satisfy this equation, the signal is called aperiodic signal.
Question 6. Define Symmetric And Antisymmetric Signal?
Answer :
A real value signal x (n) is called symmetric (even) if x (-n) =x (n). On the other hand the signal is called antisymmetric (odd) if x (-n) =x (n).
Question 7. State The Classification Of Systems?
Answer :
Question 8. Define Dynamic And Static System?
Answer :
A discrete time system is called static or memory less if its output at any instant n depends almost on the input sample at the same time but not on past and future samples of the input.
e.g. y(n) =a x (n)
In anyother case the system is said to be dynamic and to have memory.
e.g. (n) =x (n)+3 x(n-1)
Question 9. Define Time Variant And Time Invariant System?
Answer :
A system is called time invariant if its output , input characteristics dos not change with time.
e.g.y(n)=x(n)+x(n-1)
A system is called time variant if its input, output characteristics changes with time.
e.g.y(n)=x(-n).
Question 10. Define Stable And Unable System?
Answer :
A system is said to be stable if we get bounded output for bounded input.
Question 11. Define Region Of Convergence?
Answer :
The region of convergence (ROC) of X(Z) the set of all values of Z for which X(Z) attain final value.
Question 12. State Properties Of Roc.?
Answer :
Question 13. State The Methods For Evaluating Inverse Z-transform.?
Answer :
Question 14. State The Properties Of Dft?
Answer :
Question 15. How To Obtain The Output Sequence Of Linear Convolution Through Circular Convolution?
Answer :
Consider two finite duration sequences x(n) and h(n) of duration L samples and M samples. The linear convolution of these two sequences produces an output sequence of duration L+M-1 samples, whereas , the circular convolution of x(n) and h(n) give N samples where N=max(L,M).In order to obtain the number of samples in circular convolution equal to L+M-1, both x(n) and h(n) must be appended with appropriate number of zero valued samples. In other words by increasing the length of the sequences x(n) and h(n) to L+M-1 points and then circularly convolving the resulting sequences we obtain the same result as that of linear convolution.
Question 16. What Is Zero Padding?what Are Its Uses?
Answer :
Let the sequence x(n) has a length L. If we want to find the N-point DFT(N>L) of the sequence x(n), we have to add (N-L) zeros to the sequence x(n). This is known as zero padding.
The uses of zero padding are:
Question 17. Define Sectional Convolution?
Answer :
If the data sequence x(n) is of long duration it is very difficult to obtain the output sequence y(n) due to limited memory of a digital computer. Therefore, the data sequence is divided up into smaller sections. These sections are processed separately one at a time and controlled later to get the output.
Question 18. What Are The Two Methods Used For The Sectional Convolution?
Answer :
The two methods used for the sectional convolution are:
Question 19. What Is Overlap-add Method?
Answer :
In this method the size of the input data block xi(n) is L. To each data block we append M-1 zeros and perform N point circular convolution of xi(n) and h(n). Since each data block is terminated with M-1 zeros the last M-1 points from each output block must be overlapped and added to first M-1 points of the succeeding blocks.This method is called overlap-add method.
Question 20. What Is Overlap-save Method?
Answer :
In this method the data sequence is divided into N point sections xi(n).Each section contains the last M-1 data points of the previous section followed by L new data points to form a data sequence of length N=L+M-1.In circular convolution of xi(n) with h(n) the first M-1 points will not agree with the linear convolution of xi(n) and h(n) because of aliasing, the remaining points will agree with linear convolution. Hence we discard the first (M-1) points of filtered section xi(n) N h(n). This process is repeated for all sections and the filtered sections are abutted together.
Question 21. Why Fft Is Needed?
Answer :
The direct evaluation DFT requires N2 complex multiplications and N2 –N complex additions. Thus for large values of N direct evaluation of the DFT is difficult. By using FFT algorithm the number of complex computations can be reduced. So we use FFT.
Answer :
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of the symmetry and periodicity properties of twiddle factor to effectively reduce the DFT computation time.It is based on the fundamental principle of decomposing the computation of DFT of a sequence of length N into successively smaller DFTs.
Answer :
The number of multiplications and additions required to compute N point DFT using radix-2 FFT are N log2 N and N/2 log2 N respectively,.
Question 24. What Is Meant By Radix-2 Fft?
Answer :
The FFT algorithm is most efficient in calculating N point DFT. If the number of output points N can be expressed as a power of 2 that is N=2M, where M is an integer, then this algorithm is known as radix-2 algorithm.
Question 25. What Is Dit Algorithm?
Answer :
Decimation-In-Time algorithm is used to calculate the DFT of a N point sequence. The idea is to break the N point sequence into two sequences, the DFTs of which can be combined to give the DFt of the
original N point sequence.This algorithm is called DIT because the sequence x(n) is often splitted into smaller sub- sequences.
Question 26. What Dif Algorithm?
Answer :
It is a popular form of the FFT algorithm. In this the output sequence X(k) is divided into smaller and smaller sub-sequences , that is why the name Decimation In Frequency.
Question 27. What Are The Applications Of Fft Algorithm?
Answer :
The applications of FFT algorithm includes:
Question 28. Why The Computations In Fft Algorithm Is Said To Be In Place?
Answer :
Once the butterfly operation is performed on a pair of complex numbers (a,b) to produce (A,B), there is no need to save the input pair. We can store the result (A,B) in the same locations as (a,b). Since the same storage locations are used troughout the computation we say that the computations are done in place.
Question 29. Distinguish Between Linear Convolution And Circular Convolution Of Two Sequences?
Answer :
Linear convolution:
If x(n) is a sequence of L number of samples and h(n) with M number of samples, after convolution y(n) will have N=L+M-1 samples.
It can be used to find the response of a linear filter.
Zero padding is not necessary to find the response of a linear filter.
Circular convolution:
If x(n) is a sequence of L number of samples and h(n) with M samples, after convolution y(n) will have N=max(L,M) samples.
It cannot be used to find the response of a filter.
Zero padding is necessary to find the response of a filter.
Question 30. What Are Differences Between Overlap-save And Overlap-add Methods?
Answer :
Overlap-save method:
In this method the size of the input data block is N=L+M-1
Each data block consists of the last M-1 data points of the previous data block followed by L new data points
In each output block M-1 points are corrupted due to aliasing as circular convolution is employed
To form the output sequence the first
M-1 data points are discarded in each output block and the remaining data are fitted together
Overlap-add method:
In this method the size of the input data block is L
Each data block is L points and we append M-1 zeros to compute N point DFT
In this no corruption due to aliasing as linear convolution is performed using circular convolution
To form the output sequence the last
M-1 points from each output block is added to the first M-1 points of the succeeding block
Question 31. What Are The Differences And Similarities Between Dif And Dit Algorithms?
Answer :
Differences:
Similarities:
Both algorithms require same number of operations to compute the DFT.Both algorithms can be done in place and both need to perform bit reversal at some place during the computation.
Question 32. What Are The Different Types Of Filters Based On Impulse Response?
Answer :
Based on impulse response the filters are of two types:
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 sample and previous input samples.
Question 33. What Are The Different Types Of Filters Based On Frequency Response?
Answer :
Based on frequency response the filters can be classified as:
Question 34. Distinguish Between Fir Filters And Iir Filters?
Answer :
FIR filter:
IIR filter:
Question 35. What Are The Design Techniques Of Designing Fir Filters?
Answer :
There are three well known methods for designing FIR filters with linear phase .They are (1.)Window method (2.)Frequency sampling method (3.)Optimal or minimax design.
Question 36. What Is Gibb’s Phenomenon?
Answer :
One possible way of finding an FIR filter that approximates H(ejw) would be to truncate the infinite Fourier series at n=±(N-1/2).Direct truncation of the series will lead to fixed percentage overshoots and undershoots before and after an approximated discontinuity in the frequency response.
Question 37. What Are The Desirable Characteristics Of The Window Function?
Answer :
The desirable characteristics of the window are:
Question 38. Give The Equations Specifying The Following Windows?
Answer :
1. Rectangular window:
The equation for Rectangular window is given by
W(n)= 1 0 ≤ n ≤ M-1
0 otherwise
2. Hamming window:
The equation for Hamming window is given by
WH(n)= 0.54-0.46 cos 2пn/M-1 0 ≤ n ≤ M-1
0 otherwise
3. Hanning window:
The equation for Hanning window is given by
WHn(n)= 0.5[1- cos 2пn/M-1 ] 0 ≤ n ≤ M-1
0 otherwise
4. Bartlett window:
The equation for Bartlett window is given by
WT(n)= 1-2|n-(M-1)/2| 0 ≤ n ≤ M-1
M-1
0 otherwise
5. Kaiser window:
The equation for Kaiser window is given by
Wk(n)= Io[α√1-( 2n/N-1)2] for |n| ≤ N-1
Io(α) 2
0 otherwise
where α is an independent parameter.
Answer :
The necessary and sufficient condition for linear phase characteristic in FIR filter is, the impulse response h(n) of the system should have the symmetry property i.e.,
H(n) = h(N-1-n)
where N is the duration of the sequence.
Question 40. What Are The Advantages Of Kaiser Window?
Answer :
Question 41. What Is The Principle Of Designing Fir Filter Using Frequency Sampling Method?
Answer :
In frequency sampling method the desired magnitude response is sampled and a linear phase response is specified .The samples of desired frequency response are identified as DFT coefficients. The filter coefficients are then determined as the IDFT of this set of samples.
Question 42. For What Type Of Filters Frequency Sampling Method Is Suitable?
Answer :
Frequency sampling method is attractive for narrow band frequency selective filters where only a few of the samples of the frequency response are non zero.
Question 43. When Cascade Form Realization Is Preferred In Fir Filters?
Answer :
The cascade form realization is preferred when complex zeros with absolute magnitude is less than one.
Answer :
For an M_stage filter , αm-1(0) =1 and km = αm(m)
αm-1(k) = αm(k) - αm(m) • αm(m-k) , 1≤k≤m-1
1-αm2 (m)
Question 45. State The Structure Of Iir Filter?
Answer :
IIR filters are of recursive type whereby the present o/p sample depends on present i/p, past i/p samples and o/p samples. The design of IIR filter is realizable and stable.
The impulse response h(n) for a realizable filter is
h(n)=0 for n≤0
Question 46. State The Advantage Of Direct Form ΙΙ Structure Over Direct Form Ι Structure.?
Answer :
In direct form ΙΙ structure, the number of memory locations required is less than that of direct form Ι structure.
Question 47. How One Can Design Digital Filters From Analog Filters?
Answer :
Question 48. Mention The Procedures For Digitizing The Transfer Function Of An Analog Filter.?
Answer :
The two important procedures for digitizing the transfer function of an analog filter are:
Question 49. What Do You Understand By Backward Difference?
Answer :
One of the simplest method for converting an analog filter into a digital filter is to approximate the differential equation by an equivalent difference equation.
d/dt y(t)=y(nT)-y(nT-T)/T
The above equation is called backward difference equation.
Answer :
The mapping procedure between S-plane & Z-plane in the method of mapping of differentials is given by
H(Z) =H(S)|S=(1-Z-1)/T
The above mapping has the following characteristics:
Question 51. What Is Meant By Impulse Invariant Method Of Designing Iir Filter?
Answer :
In this method of digitizing an analog filter, the impulse response of resulting digital filter is a sampled version of the impulse response of the analog filter.The transfer function of analog filter in partial fraction form.
Question 52. Give The Bilinear Transform Equation Between S-plane And Z-plane?
Answer :
S=2/T(1-Z-1/1+Z-1)
Question 53. What Is Bilinear Transformation?
Answer :
The bilinear transformation is a mapping that transforms the left half of S-plane into the unit circle in the Z-plane only once, thus avoiding aliasing of frequency components.
The mapping from the S-plane to the Z-plane is in bilinear transformation is
S=2/T(1-Z-1/1+Z-1)
Question 54. What Are The Properties Of Bilinear Transformation?
Answer :
Question 55. Write A Short Note On Pre-warping.
Answer :
The effect of the non-linear compression at high frequencies can be compensated. When the desired magnitude response is piece-wise constant over frequency, this compression can be compensated by introducing a suitable pre-scaling, or pre-warping the critical frequencies by using the formula.
Question 56. What Are The Advantages & Disadvantages Of Bilinear Transformation?
Answer :
Advantages:
Disadvantage:
Question 57. What Is The Advantage Of Cascade Realization?
Answer :
Quantization errors can be minimized if we realize an LTI system in cascade form.
Question 58. Define Signal Flow Graph?
Answer :
A signal flow graph is a graphical representation of the relationships between the variables of a set of linear difference equations.
Question 59. What Is Transposition Theorem & Transposed Structure?
Answer :
The transpose of a structure is defined by the following operations:
According to transposition theorem if we reverse the directions of all branch transmittance and interchange the input and output in the flowgraph, the system function remains unchanged.
Question 60. What Are The Different Types Of Arithmetic In Digital Systems.?
Answer :
There are three types of arithmetic used in digital systems. They are fixed point arithmetic, floating point ,block floating point arithmetic.
Question 61. What Is Meant By Fixed Point Number?
Answer :
In fixed point number the position of a binary point is fixed. The bit to the right represent the fractional part and those to the left is integer part.
Question 62. What Are The Different Types Of Fixed Point Arithmetic?
Answer :
Depending on the negative numbers are represented there are three forms of fixed point arithmetic. They are sign magnitude,1’s complement,2’s complement
Question 63. What Is Meant By Sign Magnitude Representation?
Answer :
For sign magnitude representation the leading binary digit is used to represent the sign. If it is equal to 1 the number is negative, otherwise it is positive.
Question 64. What Is Meant By 1’s Complement Form?
Answer :
In 1,s complement form the positive number is represented as in the sign magnitude form. To obtain the negative of the positive number ,complement all the bits of the positive number.
Question 65. What Is Meant By 2’s Complement Form?
Answer :
In 2’s complement form the positive number is represented as in the sign magnitude form. To obtain the negative of the positive number ,complement all the bits of the positive number and add 1 to the LSB.
Question 66. What Is Meant By Floating Pint Representation?
Answer :
In floating point form the positive number is represented as F =2CM,where is mantissa, is a fraction such that1/2<M<1and C the exponent can be either positive or negative.
Question 67. What Are The Advantages Of Floating Point Representation?
Answer :
Answer :
Question 69. What Is Input Quantization Error?
Answer :
The filter coefficients are computed to infinite precision in theory. But in digital computation the filter coefficients are represented in binary and are stored in registers. If a b bit register is used the filter coefficients must be rounded or truncated to b bits ,which produces an error.
Question 70. What Is Product Quantization Error?
Answer :
The product quantization errors arise at the out put of the multiplier. Multiplication of a b bit data with a b bit coefficient results a product having 2b bits. Since a b bit register is used the multiplier output will be rounded or truncated to b bits which produces the error.
Question 71. What Are The Different Quantization Methods?
Answer :
Truncation and Rounding
Question 72. What Is Truncation?
Answer :
Truncation is a process of discarding all bits less significant than LSB that is retained
Question 73. What Is Rounding?
Answer :
Rounding a number to b bits is accomplished by choosing a rounded result as the b bit number closest number being unrounded.
Question 74. What Are The Two Types Of Limit Cycle Behavior Of Dsp?
Answer :
Question 75. What Are The Methods To Prevent Overflow?
Answer :
Question 76. State Some Applications Of Dsp?
Answer :
Speech processing ,Image processing, Radar signal processing.
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