EC2361 DIGITAL SIGNAL PROCESSING UNIT I INTRODUCTION 9

Classification of systems: Continuous, discrete, linear, causal, stable, dynamic, recursive, time variance; classification of signals: continuous and discrete, energy and power; mathematical representation of signals; spectral density; sampling techniques, quantization, quantization error, Nyquist rate, aliasing effect. Digital signal representation. UNIT II - DISCRETE TIME SYSTEM ANALYSIS 9

Z-transform and its properties, inverse z -transforms; difference equation Solution by z-transform, application to discrete systems - Stability analysis, frequency response Convolution Fourier transform of discrete sequence Discrete Fourier series. UNIT III - DISCRETE FOURIER TRANSFORM & COMPUTATION 9 DFT properties, magnitude and phase representation - Computation of DFT using FFT algorithm DIT & DIF - FFT using radix 2 Butterfly structure. UNIT IV - DESIGN OF DIGITAL FILTERS 9

FIR & IIR filter realization Parallel & cascade forms. FIR design: Windowing Techniques Need and choice of windows Linear phase characteristics. IIR design: Analog filter design - Butterworth and Chebyshev approximations; digital design using impulse invariant and bilinear transformation - Warping, prewarping -Frequency transformation. UNIT V - DIGITAL SIGNAL PROCESSORS 9

Introduction Architecture Features Addressing Formats Functional modes -Introduction to Commercial Processors L = 45, T = 15, TOTAL: 60 PERIODS

UNIT 1 INTRODUCTION (SIGNALS AND SYSTEMS) 1. Define Signal. A signal is defined as any physical quantity that varies with time, space or any other independent variable or variables. 2. What is multidimensional signal? Give examples. A signal which is a function of two or more independent variables is called multidimensional signal. The intensity or brightness of black & white photograph at each point is a function of two independent spacial coordinates x and y. Hence it is a two dimensional signal and can be denoted as I (x,y).The intensity of black & white motion picture is a function of x & y coordinates and time. Hence it is a three dimensional signal and can be denoted as I (x,y,t). 3. What is analog signal? The analog signal is a continuous function of an independent variable such as time, space, etc. The analog signal is defined for every instant of the independent variable and so the magnitude (or the value) of analog signal is continuous in the specified range. Here both the magnitude of the signal and the independent variable are continuous 4. Distinguish between energy and power signals. Energy signals: For discrete time signal, energy (E) of x(n) is given by,

E=
Power signal: Average power of x(n) is P=

( )

( )

5.Define random and deterministic signal. Random signal: A signal is called a random signal if it cannot be described with certainty before it actually occurs. Probabilistic models describe random signals. E.g.: Thermal noise in resistors, transistors etc.

Deterministic signal: A signal is called deterministic signal if it can be described without any uncertainty. 6. Give the expressions for even and odd signals. Any signal x(t) can be broken into a sum of two signals, Even and Odd signals. Even signal is equal to, E{x(n)}=1/2 [x(n) + x(-n)] Odd signal is equal to, O{x(n)}=1/2 [x(n) x(-n)]. 7. What are the properties of linear time invariant systems? Linearity: this principle requires that the response of the system to a weighted sum of i/p signals is equal to the corresponding weighted sum of the responses of the system to each of the individual i/p signals (i.e.) H[ 1x1(n)+ 2x2(n)] = 1 H[x1(n)]+ 2 H[x2(n)]= 1y1(n) + 2 y2(n) Time invariant system (or) shift invariant or fixed if its i/p-o/p relationship does not change with time (i.e.) if y(t) is the response when x(t) is the i/p and if x(t) is delayed by time ,then the o/p of the system is also delayed by the same amount of time . H[x(n-N)] = Y(n-N) 8. State whether the following systems are linear or not. 1. y(n) = x(n)+x(n-100) 2. g(t) = dx(t)/dt Soln: 1. y(n) = x(n)+x(n-100) H[ 1x1(n)+ 2x2(n)] = 1 H[x1(n)]+ 2H[x2(n)]= 1y1(n) + 2 y2(n) RHS y1(n) = x1(n) + x1(n-100) y2(n) =x2(n) + x2(n-100) LHS H[ 1x1(n) + 2x2(n)] = 1x1(n) + 2x2(n) + 1x1(n-100) + 2x2(n-100) RHS 1 H[x1(n) + x1(n-100)] + 2 H[x2(n) + x2(n-100)] = 1x1(n) + 1x1(n-100) + 2x2(n) + 2x2(n-100)

LHS = RHS It is linear. 2.g(t) = dx(t)/dt LHS [ 1x1(t) + 2x2(t)] = 1dx1(t)/dt + 2dx2(t)/dt RHS = 1y1(t) + 2y2(t) = 1dx1(t)/dt + 2dy2(t)/dt LHS = RHS It is linear. 9. Define the impulse response of a DISCRETE system. In discrete time systems the unit impulse signal is represented by (n) = 1, n=0 = 0, otherwise when the unit impulse (also called delta function ) (n) is applied to a linear time invariant at n=0,the impulse response of the system is denoted by h(n) = h[ (n)] 10.What is meant by aliasing effect? The superimposition of high frequency behavior on to the low frequency behaviour is referred as aliasing. This effect is also referred as folding. 11.Is the system described by the equation y(n) = x(2n) time invariant or not? why? T[x(n-N)] = y(n-N) LHS =T[x(n-N)] = x[2(n-N)] RHS = y(n-N) = x[2n-N] LHS = RHS. The system is time variant. 12. Is the system y(n) = nx(n) is shift invariant or not. shift invariant condition:T[x(n-N)] = y(n-N)

LHS = T[x(n-N)] = nx(n-N) RHS = y(n-N) = (n-N)x(n-N)= nx(n-N)-Nx(n-N) LHS RHS It is shift variant. 13. What is an LTI system? An LTI system is one which possess two of the basic properties ,linearity and time invariance. Linearity: An LTI system obeys superposition principle which states that the output of the system to a weighted sum of inputs is equal to the corresponding weighted sum of the outputs to each of the individual inputs. Time invariance:If the input-output relation of a system does not vary with time the system is said to be time invariant. 14. Define impulse response of a system? The response obtained from a system when the input signal is a unit sample is known as unit sample or impulse response h(n) = H[ (n)] 15. Define system transfer function Let x(n) and y(n) be the input and output of an LTI system with impulse response h(n).Then the system transfer function of the LTI system is defined as the ratio of Y(Z) and X(Z) as H(Z) = Y(Z)/X(Z),Where Y(Z) is the z transform of the output signal y(n) and X(Z) is the z transform of the input signal x(n). 16. What is the necessary and sufficient condition on the impulse response for stability? The necessary and sufficient condition for LTI system is that it s impulse response is absolutely summable. |h(k)| < 17. Distinguish between recursive and non recursive realization? For recursive realization the present output y (n) is a function of the past outputs and the present inputs. This form corresponds to an infinite impulse response (IIR) digital filter.

For non- recursive realization the present output y(n) is a function of only past and present inputs. This form corresponds to a finite impulse response (FIR) digital filter. 18. What is a causal system? Give an example A system is said to be causal if the output of the system at any time n depends only on present and past input, but does not depend on future inputs. This can be represented mathematically as, y (n) = F[x (n), x (n-1), x (n-2)] e.g. y(n) = x(n) + x(n-1). 19. What is quantization error? The difference between the quantized unquantized signal is called quantization error 20. Define Nyquist rate The sampling frequency must be greater than or equal to twice the maximum frequency 2Fm fs 21. What is an anti-aliasing filter? A filter that is used to reject high frequency signals before it is sampled to remove the aliasing of unwanted high frequency signals is called an ant aliasing filter. 22. What is meant by quantizer? It is a process of converting discrete time continuous amplitude into discrete time discrete amplitude. signal and the original

UNIT II - DISCRETE TIME SYSTEM ANALYSIS 1. Define Z-Transform Z-transform can be defined as Z[x(n)]=X(Z)= ( )
-n

This equation is referred to as Bilateral or two sided Z-transform of x(n). The unilateral z-Transform can be defined as Z[x(n)]=X(Z)= 2. Define Region of convergence The range of values of Z for which X(Z) expressed by the equation Z[x(n)]=X(Z)= ( )
-n

( )

-n

approaches to a finite value is called Region of convergence(ROC). 3. State Initial value theorem If x(n) is causal then x(0) = It can be used to find the initial value of x(n) 4. State Final value theorem If Z[x(n)]=X(Z) then X()=Lt x(n)=Lt (1-Z-1) X(Z) 5.What are the properties of ROC? 1. 2. 3. ROC doesnot contain any poles. If x(n) is a finite duration then the ROC is the entire Z plane except possibly Z=0 and/Or Z= If x(n) is a r ight sided sequence and if the circle with |Z|=r o is in ROC then all the finite values of Z for which Z> ro will also be in ROC. If x(n) is a left sided sequence and if the circle with |Z|=r o is in ROC then all the finite values of Z for which 0<Z <ro will also be in ROC. ( ) = ( )

4.

5. If x(n) is a two sided sequence and if the circle with |Z|=r o is in ROC then ROC will consist of a ring in the Z plane which includes Z=ro

6. List out any 4 properties of Z transform. 1. Linearity: X(Z)=a x1(z)+bx2(z) 2. Time shifting : Z( x(nno))=Zno X(Z) 3. Frequency Shift: Z(ejwn x(n))=X(e-jwZ) 4. Time reversal: Z(x(-n))=x(1/Z)=x(z-1)

7. What are the three methods to obtain the inverse Z transform ? 1.Long division method 2.Partial fraction expansion method 3.Residue method 8. Find the Z.T of x(n)= (n) From the definition of Z.T X(Z)= Where (n) =1 for n=0 =0 otherwise 9. Find the Z-transform of x(n)=(1/2)n u(n) We know that Z transform of an u(n) = z/(z-a). Similarly here a = . Therefore X(Z) = z/(z-1/2) = 2z/(2z-1). 10. Define discrete linear convolution. The discrete convolution of the two discrete variable function x(n) andh(n) is the discrete variable function y(n) given by the summation y(n) = ( ) ( ) ( )
-n

( )

-n

= 1.z0 = 1

11. Define discrete circular convolution. Given two real N periodic sequences ,x(n)&h(n) the circular or periodic convolution sequence y(n) is also n periodic sequence given by , y(n) = ( )( ( ))N

12. What are the advantages of sectioned convolution? If one of the sequences is very much larger than the other, then it is very difficult to compute convolution using other types of convolution. Long delay in getting output. Therefore entire sequence is required before convolution operation. Memory required also very large to store the sequence. These problems can be eliminated in sectioned convolution. 13. What is sectioned convolution? In sectioned convolution the larger sequence is sectioned in to smaller sequences and then linear or circular convolution is done. Output sequence obtained from convolution of all sections are combined to get overall output sequence. 14. What are the different methods of sectioned convolution? 1. overlap add method 2. overlap save method 15. What is the draw back in Fourier transform and how it is overcome? The drawback in Fourier transform is that it is a continuous function o f and so it cannot be processed by digital system. This drawback is overcome by using Discrete Fourier Transform. The DFT converts the continuous function of to a discrete function of , 16. Calculate the DFT of the sequence x (n) = {1,1,-2,-2}. The N-point DFT of x(n) is given by DFT{x(n)} = X(k) = ( )
-j2kn/N

; for k = 0,1,2,(N-1)

Since x(n) a 4-point sequence, we can take 4-point DFT.

X(k) =
=

( )

-j2kn/4

( )

-jkn/2

= x(0) e0 + x(1) e-jk/2 + x(2) e-j2k + x(3) e-j3k/2 1+ e-jk/2 - 2 e-j2k 2 e-j3k/2 ; for k = 0,1,2,3

17. Give two applications of DFT? i) The DFT is used for spectral analysis of signals using a digital computer.

ii) The DFT is used to perform filtering operations on signals using digital computer.

18. What is the relation between Z-transform & DFT?

(MAY 2011)

Let N-point DFT of x(n) be X(k) and the Z-transform of x(n) be X(z).The N-point sequence X(k) can be obtained form X(z) by evaluating X(z) at N equally spaced points around the unit circle.

X(k) = X(z)|z=ej2nk/N ; for k = 0,1,2,(N-1)

19.State the conditions for the existence of fourier series. (i). The function x(t) should be single valued in any finite time interval T (ii). The function x(t) should have atmost finite number of discontinuities in any finite time interval T. (iii). The function x(t) should have finite number of maxima and minima in any time interval T. (iv) The function x(t) should be absolutely integrable. 20. Define fourier transform & inverse fourier transform of a discrete time signal. The fourier transform of a discrete time signal x(n) is defined as F{x(n)}=X() = The fourier transform exists only if ( )
-jn

( )<

The inverse fourier transform of X() is defined as

F-1{X()}=
21.State Rayleighs energy theorem.

( ) ejn d

Rayleighs energy theorem states that the energy of the sign al may be written in frequency domain as superposition of energies due to individual spectral frequencies of the signal.

22.State Parsevals power theorem. Parsevals power theorem states that the total average power o f a periodic signal x(t) is equal to the sum of the average powers of its phasor components. UNIT III - FAST FOURIER TRANSFORM 1. What is zero padding? Adding zeros to the sequence to make the sequence of equal length to perform linear convolution via circular is known as zero padding. 2. What are the advantages of FFT algorithm? Fast fourier transform reduces the computation time. In DFT computation, number of multiplication is N2 and the number of addition is N(N-1). In FFT algorithm, number of multiplication is only N/2(log2N) . Hence FFT reduces the number of elements (adder, multiplierZ&delay elements). This is achieved by effectively utilizing the symmetric and periodicity properties of Fourier transform. 3. What are the properties of DIT FFT? 1. Computation are done in place. Once a butterfly structure operation is performed on a pair of complex numbers (a,b) to produce (A,B) there is no need to save the input pair (a,b). Hence we can store the results (A,B) in 4. What is decimation in time? The decimation in time algorithm is based on the decomposition of DFT computation by forming smaller and smaller sequences of the input sequence x(n). 5. What is decimation in frequency? Dividing the output sequence X(k) in to smaller and smaller subsequences is called decimation in frequency. 6. What is composite FFT algorithm? FFT is mixed or composite when N is a composite number which has more than one prime factor.(ie) N = m1,m2,.mr if N=m1N1 where N1=m2,m3,.,mr. the input sequence can be separated in to m1 sub sequences of N1 elements each.

7.Compare DIT & DIF FFT algorithms DIT FFT 1. input is bit reversed order 2. output is normal order 3.time domain sequence is decimated DIF FFT 1. input is normal order 2. output is bit reversed order 3. freq.domain sequence is decimated 4. total no. of multipliers required is 4. multipliers N/2(log2N) N/2(log2N) required is

adders required is N(log2N) 5. In butterfly diagram, each stage of computation, the phase factor is multiplied before add &subtract operation. 5. In this case, in each stage of computation, the phase factors are multiplied after add & subtract operation.

8. What is twiddle factor? It is a complex valued phase factor or weight vector represented as

WNKn = e-(j2kn/N)
9.What are the steps involved in computing IDFT through FFT? 1. take conjugate of x(k) 2. compute N point DFT of complex conjugate x*(k) using FFT. 3. take again the conjugate of the output sequence. 4. then the resultant sequence is divided by N. 10. Compute the DFT of x(n) =(n).

X(k) =

( ) WNKn = =

( ) WNKn

11.Define Duality property for DFT. DFT: F (k) F (n) g (n)

1/N g (-k)

12.What is the relation between Z Transform and DFT. Let N point DFT of x(n) be X(K) and the Z transform of x(n) be X(Z).The N point sequence of X(K) can be obtained from X(Z) at N equally spaced points around the unit circle. X(K)=X(Z)/Z=e j2k/N for K=0,1,2,..(N-1) 13.What is a decimation in time algorithm? DIT algorithm is used to calculate the DFT of a N point sequence. Initially the N point sequence is divided into two N/2 point sequences Xeven (n) and Xodd (n). The N/2 point DFTs of these two sequences are evaluated and combined to give the N point DFT. Similarly the N/2 point DFTs can be expressed as a combination of N/4 point DFTs. This process is continued until left with 2 point DFT. This algorithm is called decimation in time because the sequence X (n) is often splitted into smaller sequences. 14. What is meant by radix-2 FFT? The FFT algorithm is most efficient in calculating N point DFT. If the number of point N can be expressed as a power of 2 i.e. N= 2M where M is an integer, then this algorithm is known as radix-2 FFT algorithm. 15. What is decimation in frequency algorithm? It is one of the FFT algorithms. In this the output sequence X(k) is divided into smaller subsequence, that is why the name decimation in frequency. Initially the input sequence is divided into two consisting of the first N/2 samples of X(n) and the last N/2 samples of X(n).The above procedure can now be iterated to express each N/2 point DFT as a combination of two N/4 point DFTs.This process is continued until we are left with 2 point and 1 DFT. 16. What are the differences and similarities between DIF and DIT algorithms? Differences : For DIT the input is bit reversed while the output is in natural order , whereas for DIF the input is in natural order while the output is bit reversed. 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. Both algorithms can be done in place and both need to perform bit reversal at some place during the computation. 17. Explain In-place computation. To compute the elements p and q of the mth array , it is required to have elements in the p and q of the (m-1) array. If Xm(p) and Xm(q) are stored in the same register as Xm-1(p) and Xm-1(q) respectively ,it is possible to implement the above computation with only N array of complex storage registers. This kind of computation is commonly referred to as In-place computation. 18. Calculate the number of multiplications needed in the calculation of DFT and FFT with 64 point sequence. Number of complex multiplications required using direct computation is N2 = 642 = 4096 Number of complex multiplications required using FFT is (N/2) log N = ((64/2) log 64 = 192 speed improvement factor (4096/192) = 21.33. 19. Define discrete linear convolution. The discrete convolution of the two discrete variable function x(n) and h(n) is the discrete variable function y(n) given by the summation y(n) = ( ) ( )

20. What are the properties of DIT FFT? 1.Computation are done in place. Once a butterfly structure operation is performed on a pair of complex numbers(a,b) to produce (A,B) there is no need to save the input pair (a,b). Hence we can store the results(A,B) in the same location as(a,b). 2. Data x(n) after decimation is stored in reverse order. 21. What are the advantages of FFT algorithm? Fast fourier transform reduces the computation time. In DFT computation, number of multiplication is N2 and the number of addition is

N(N-1). In FFT algorithm, number of multiplication is only N/2(log 2N) . Hence FFT reduces the number of elements (adder,multiplierZ&delay elements). This is achieved by effectively utilizing the symmetric and periodicity properties of Fourier transform. 22. What are the computational savings in evaluation of DFT using radix-2 FFT? Multiplications: N/2 logN2 Additions: N/ logN2 23.State the meaning of bit reversal in FFT algorithm. DIT FFT: I/P bit reversal order & O/P normal order DIF FFT: O/P bit reversal order & I/P normal order. UNIT IV - DESIGN OF DIGITAL FILTERS 1. What are the disadvantages of Impulse invariant method? Although this method is useful for implementing LPF and HPF the method is unsuccessful for implementing digital filters for which |H(j)| does not approach zero for large value of such as the high pass filter . 2. What are the advantages of Bilinear transformation method? The Bilinear transform method provides non linear one to one mapping of the frequency points on the j axis in the S plane to those on the unit circle in the Z plane .i.e. Entire j axis for - < < maps uniquely on to a unit circle -/T < /T < -/T .This procedure allows us to implement digital high pass filters from their analog counter parts. 3. Define the pole mapping rule in Bilinear transformation method. A pole at s = sp in the s plane is transferred into a zero at z= -1 and a pole at Z = (2+ sp Ts)(2- sp Ts) in the z plane . 4. Define prewarping or prescaling. For large frequency values the non linear compression that occurs in the mapping of to is more apparent .This compression causes the transfer function

at high frequency to be highly distorted when it is translate to the domain. This compression is being compensated by introducing a prescaling or prewarpping to frequency scale. For bilinear transform scale is converted into * scale (i.e.) * =2/Ts tan ( Ts/2) (prewarped frequency) 5. What are the disadvantages of FIR filter? Long sequences of h (n) are generally required to adequately approximate sharp cut off filters. A large amount of processing is required to realize the filter if slow convolution is used .By using FFT algorithms these filters can be designed more efficiently. 6. What are the methods used to design FIR filter? 1. Window Method: It involves straight forward analytical procedure however in some cases iteration is required to obtain the desired result 2.Frequency Sampling: A desired frequency response is uniformly sampled and filter coefficient are then determined from these samples using the discrete Fourier transform. 3. Optimal or minimax design: Minimizing the maximum error between the desired and the actual frequency response by spreading the error in PB and SB. 7. Why direct Fourier series method is not used in FIR filter design? The impulse response h(n) is infinite in duration. The filter is unrealizable since the impulse response begins at- i.e. no finite amount of delay can make the impulse response realizable. Therefore the filter which results from a Fourier series representation of h (ejw) is an unrealizable FIR Filter.

8.Compare analog and digital filters. ANALOG FILTER 1. In analog filter both input and output continuous time signal . DIGITAL FILTER 1. In digital filter,both the input and output are discrete time signals.

2. It can be constructed using active 2. It can be constructed using adder, and passive components. multiplier and delay units. 3. these filters operate in infinite frq. Range, theoretically but in practice it is limited by finite max. operating freq. depending upon the devices used. 3. freq. range is restricted to half the sampling range and it is also restricted by max. computational speed available for particular application

4. It is defined by linear differential eqn. 4.It is defined by linear difference eqn

9.What are the advantages of digital filter? 1.Filter coefficient can be changed any time thus it implements the adaptive future. 2. It does not require impedance matching between input and output. 3. Multiple filtering is possible. 4. Improved accuracy, stability and dynamic range. 10.What are disadvantages of Digital Filter? 1. The bandwidth of the filter is limited by sampling frequency. 2.The performance of the digital filter depends on the hardware used to implement the filter. 3.The quantization error arises due to finite word length effect in representation of signal and filter coefficient.

11.What is the difference between Chebyshev Filter type I and typeII? Filter TypeI: It is all pole filter and exhibits equiripples in the passband and monotonic characteristics in the stopband. Filter Type II: It contains both poles and zeros and exhibits a monotonic behaviour in the pass band and equiripple in the stop band. 12. What are the properties of chebyshev filter? 1.For 1 H(j) decreases monotonically towards zero. 2.For 1 H(j) it oscillates between 1 and 1\(1+^2) 13. Compare Butterworth filter and chebyshev filter. Butterworth filter 1.The Magnitude response of Butterworth filter decreases monotonically as the frequency increases. 2. The Transition width is more 3. The order of butterworth filter is more, thus it requires more elements to construct and is expensive. 4. The Poles of the butterworth filter lies along the circle. 5. Magnitude response is flat at =0 thus it is known as maximally flat filter. Chebyshev Filter 1.The Magnitude response of chebyshev filter will not decrease monotonically with frequency because it exhibits ripples in pass band or stop band. 2.The Transition width is very small 3. For the same specifications the order of the filter is small and is less complex and inexpensive.

4.The poles of chebyshev filter lies along the ellipse. 5.Magnitude response produces ripples in the pass band or stop band thus it is known as equripple filter. 14.Compare Bilinear transformation and impulse invariant transformation BILINEAR TRANSFORMATION IMPULSE INVARIANT TRANSFORMATION

1. It is one to one mapping 1. It is many to one mapping 2. The relation between analog and 2.The relation between analog and digital digital frequency is nonlinear, ie frequency is linear,ie =T or =/T =2/T tan( /2) 3. Due to nonlinear relation between 3. The aliasing error occur due to sampling and distortion occurs in thus this method is suitable for design of only frequency domain of digital filter. band limitied filters such Low pass and Band pass. 4. Due to the warping effect both 4. The frequency response of analog can be amplitude and phase response of preserved by selecting low sampling time or analog filter are affected but the high sanpling frequency. magnitude response may be preserved by applying pre- warping procedure.

15.What are the disadvantages of FIR filter? i.The duration of impulse response should be large to realize sharp cut off filters. ii.The non-integral delay can lead to problems in some signal processing applications.

16.What is the necessary and sufficient conditions for linear phase characteristics of a FIR filter? The necessary and sufficient conditions for linear phase characteristics of a FIR filter is that the phase function should be a linear function of , which in turn requires constant phase delay or constant phase and group delay. 17. Define following terms: i. Transition band/region ii. Bandwidth of the filter. Transition band: The transition of the freq response from the passband to stop band defines the transition band or transition region of the filter. BW of filter: The bandwidth of passband of the filter is called as bandwidth of the filter . 18. What do you mean by Gibbs Phenomenon? In the design of FIR filters using windows,the truncation of the fourier series introduce ripples in the freq response characteristics H() due to the non-uniform convergence of the fourier series at a discontinuity.This oscillatory behaviour near the band edge of the filter is called Gibbs phenomenon. 19. What are thepossible types of impulse response for linear phase FIR filter? There are 4 types. i.Symmetric impulse response when N is odd ii. Symmetric impulse response when N is even iii. Antisymmetric impulse response when N is odd iv. Antisymmetric impulse response when N is even.

20.. List well known design techniques for linear phase FIR filter? i.Fourier series method and window method. ii.Freq sampling method iii.Optimal filter design methods. 21. Write the expression for order of Butterworth filter?

N= log(/ )1/2/ log(1/k)1/2

22. Write the expression for the order of chebyshev filter?

N = cosh-1(/)/ cosh-1(1/k)
23.Why ideal frequency selective filters are not realizable? Ideal frequency selective filters are not realizable because they are noncausal. That is, its impulse response is present for negative values of n also. 24. Why direct form-II structure is preferred most and why? The numbers of delay elements are reduced in direct form-II structure compared to direct form-I structure. That means the memory locations are reduced in direct form-II structure. 25. Why direct form-I and direct form-II are called as direct form structures? The direct form-I and direct form-II structures are obtained directly from the corresponding transfer function without any rearrangements. So these structures are called as direct form structures. 26. What is advantage of direct form structure? Implementation of direct form is very easy. 27. Give the disadvantage of direct form structure? Both direct form structures are sensitive to the effects of quantization errors in the coefficients. So practically not preferred

28. How phase distortion and delay distortion are introduced? 1. The phase distortion is introduced when the phase characteristics of a filter is Nonlinear with in the desired frequency band 2. The delay distortion is introduced when the delay is not constant with in the Desired frequency band.

UNIT V - DIGITAL SIGNAL PROCESSORS 1. What is the advantage of having separate program & database in C5x processors Separate program & databases allow simultaneous access to program instructions & data, providing a high degree of parallelism. 2. What are the elements in CALU of C5X processor? i. 16x16 bit parallel multiplier ii. Arithmetic Logic Unit (ALU) iii. Accumulator (ACC) iv. Accumulator buffer (ACCB) v. Product register (PREG) 32 bits vi. 0-16 bit left barrel shifter & right barrel shifter. 3. Draw the bit assignment diagram pf ST0 (status register 0) and ST1 (Status register 1) 12 15-13
ARP

11 OVM

10 1

9 INTM

0-8 DP

OV

ST0 -Bit Assignment

15 13

12

11

10

8-7 6

3-2

1-0

ARB

CNF

TC

SXM

11

HM

FX

11

PM

ST1 Bit Assignment

4. What is BMAR? BMAR is Block Move Address Register. The 16 but BMAR holds an address value to be used with block moves & multiply / accumulate operations. This register provides the 16-bit address for an indirect addressed second operand. 5. What is DARAM & SARAM? DARAM is Data / Program Dual Access RAM & SARAM is Single Access RAM. 6. What are the on-chip peripherals in C5X processors? 1. Clock Generator 2. Hardware Times 3. Software Programmable Wait state Generators. 4. Parallel I/O Ports 5. Host Post Interface (HPI) 6. Serial Port 7. Buffered Serial Port (BSP) 8. time division multiplexed (TDM) serial port 9. User Maskable Interrupts 7. What are the different addressing modes of C5X processors? 1. Direct addressing 2. Memory mapped register addressing 3. Indirect addressing 4. Immediate addressing 5. Dedicated register addressing 6. Circular addressing 8. Explain any two Move instruction in C5X processor? DMOV -Copies the data from one memory location to the next higher location. BLDP - Block move data from data memory to program memory.

9. Explain any two multiplication instruction in C5X processor? MPY - Multiply numbers in 2s complement form MPYU - Multiply unsigned numbers. 10. Explain any two shift / logical instruction in C5X processor? ANDB - ANDing Acc with ACCB. ROL - Rotate ACC left once. 11. What is PUSH instruction and POP instruction? The PUSH instruction pushes the values down one level in the seven lower locations of the stack.The POP instruction pops the top of the stack to ACC. 12. What is absolute addressing? In the absolute addressing mode, the complete address of the operand is explicitly specified in the instruction itself. The content of neither the data pointer / stack pointer nor any of the registers including the accumulators and ARS are used for finding the address. 13. What is Stack addressing? They system stack is used to automatically store the program counter during interrupts & subroutines. It can also be used at your discretion to stone additional items of context on to pass data values. The stack is filled from the highest to the lowest memory address. 14. What is Immediate addressing? The immediate addressing mode can be used to load either a 16-bit constant or a constant of length 13, 9 or 7 This mode is indicated by the symbol #. For eg. ADD # 56h adds 56h to ACC. 15. Write any two instruction for addition and subtraction in C5X processor? ADD 55h, 2 ADD # 23h SUB 55h, 2 SUB # 23h

16. Write any two Load / Store instruction in C5X processor? LACB SACB 17. What are control instructions? i. Branch & Call instruction ii. PUSH & POP instruction iii. RET instruction iv. Repeat Instruction 18. How many buses does C54X processor have & what are they? C54X has eight major 16-bit bases (four program / data buses & four address bus) 19. What are the elements present in CPU of 54X processor? i. 40-Bit Arithmetic Logic Unit (ALU) ii. Two 40- Bit Accumulator Registers iii. Barrel Shifter iv. Multiply / Accumulate Block v. 16-Bit Temporary Register (T) vi. 16-Bit Transition Register vii. Compare, select and store unit viii. Exponent Encoder 20. What is IMR and IFR? IMR (Interrupt mask register) individually masks off specific interrupts at required times. IFR (Interrupt flag register) indicates the current status of the interrupts. 21. What is ST0 & ST1? ST0 & ST1 (Status Registers 0 & 1) contain the status of the various conditions and modes of 54X devices.ST0 contains the flags produced by arithmetic operations and bit manipulations, in addition to the DP and ARP fields. ST1 reflects the status of modes and instructions executed by the processor.

22. What is PMST? PMST is Processor Mode Status Register. The PMST register us loaded with memory mapped instructions such as STM. 23. What is Barrel Shifter? The Barrel Shifter is used for scaling operations such as prescaling an input data-memory operand or the accumulator value before an ALU operation; performing a logical or arithmetic shift of the accumulator value; normalising the accumulator; postscaling the accumulator before storing the accumulator value into data memory. 24. What is DAGEN and PAGEN? DAGEN is Data Address Generation Logic & PAGEN is Program Address Generation logic. 25. List out the typical features of Digital Signal Processor (MAY 2011) Fixed-point processor (TMS320C5000, 56000...) or floating point processor (TMS320C67, 96000...) Architecture optimized for intensive computation. For instance the TMS320C67 can do 1000 Million floating point operations a second (1 GIGA Flop). Narrow address bus supporting a only limited amounts of memory. Specialized addressing modes to efficiently support signal processing operations (circular addressing for filters, bit-reverse addressing for Fast Fourier Transformsetc.) Narrow data formats (16 bits or 32 bits typical). Many specialized peripherals integrated on the chip (serial ports, memory, Low power consumption & Low cost. 26. State the intended applications of DaVinci Digital Media processors Image compression, Image coding ,speech compression, multirate signal filters