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UNIT I INTRODUCTION
(2 Mark Questions and Answers)
1. Define DSP.
DSP - Digital Signal Processing.
It is defined as changing or analyzing information which is measured as discrete time
sequences.
2. List out the basic elements of DSP.
* signal in
* Analog to Digital converter
* Digital Signal processor
* Digital to Analog converter
* signal out
3. Mention the advantages of DSP.
* Veracity
* Simplicity
* Repeatability
4. Give the applications of DSP.
* Telecommunication spread spectrum, data communication
* Biomedical ECG analysis, Scanners.
* Speech/audio speech recognition
* Military SONAR, RADAR
5. 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.
6. Define system.
A set of components that are connected together to perform the particular task.
E.g.; Filters
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two conditions are not satisfied then the signal is said to be Neither energy nor power
signal
13. What is analog signal?
The analog signal is a continuous function of independent variables. The analog
Signal is defined for every instant of independent variable and so magnitude of
Independent variable is continuous in the specified range. here both the independent
Variable and magnitude are continuous.
14. What is digital signal?
The digital signal is same as discrete signal except that the magnitude of signal is
Quantized
15. What are the different types of signal representations?
a. Graphical representation
b. Functional representations
c. Tabular representation
d. Sequence representation.
16. 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.
17. State the classification of discrete time signals.
The types of discrete time signals are
* Energy and power signals
* Periodic and A periodic signals
* Symmetric (Even) and Ant symmetric (Odd) signals
18. Define energy and power signal.
If E is finite i.e. 0<E< , then x (n) is called energy signal.
If P is finite i.e. 0<P< , then the signal x(n) is called a power signal.
19. What are all the blocks are used to represent the CT signals by its samples?
* Sampler
* Quantizer
20. Define sampling process.
Sampling is a process of converting Ct signal into Dt signal.
21. Mention the types of sampling.
* Up sampling
* Down sampling
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h. Circular symmetry
4. State periodicity property with respect to DFT.
If x(k) is N-point DFT of a finite duration sequence x(n), then
x(n+N) = x(n) for all n.
X(k+N) = X(k) for all k.
5. State periodicity property with respect to DFT.
If X1(k) and X2(k) are N-point DFTs of finite duration sequences x1(n) and
x2(n), then DFT [a X1(n) + b X2(n)] = a X1(k) + b X2(k), a, b are constants.
6. State time reversal property with respect to DFT.
If DFT[x(n)] =X(k), then
DFT[x((-n))N] = DFT[x(N-n)] = X((-k))N = X(N-k)
7. Define circular convolution.
Let x1(n) and x2(n) are finite duration sequences both of length n with DFTs
x1(k) and x2(k). If X3(k) = X1(k) X2(k), then the sequence X3(k) can be obtained by
circular convolution.
8. What is the need for DFT?
DFT is used for analysis the both periodic and a periodic signals.
9. What is zero padding? What are its uses?
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
1) We can get better display of the frequency spectrum.
2) With zero padding the DFT can be used in linear filtering.
10 .Why FFT is needed?
The direct evaluation of DFT requires N complex multiplications and
N -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
11. What is FFT?
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
mutation of DFT of a sequence of length N into successively smaller DFTs.
12. How many multiplications and additions are required to compute N point DFT using
Radix-2 FFT?
The number of multiplications and additions required to compute N point DFT
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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.
2.
3.
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.
18. What are the differences and similarities between DIF and DIT algorithms?
Differences:
1) The input is bit reversed while the output is in natural order for DIT,
whereas for DIF the output is bit reversed while the input is in natural order.
2) The DIF butterfly is slightly different from the DIT butterfly, the
difference being that the complex multiplication takes place after the addsubtract operation in DIF. Similarities:
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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.
19. What is meant by radix-2 FFT?
M
If the number of output points N can be expressed as a power of 2, i.e., N = 2
Where M is an integer then this algorithm is known as radix-2 algorithm.
20. What is DIT radix-2 algorithm?
The radix 2 DIT FFT is an efficient algorithm for computing DFT. The
idea is to break N point sequence in to two sequences, the DFT of which can
be combined to give DFT of the original N-point sequence. Initially the N
point sequence is divided in to two N/2 point sequences, on the basis of odd
and even and the DFTs of them are evaluated and combined to give N-point
sequence. Similarly the N/2 DFT s are divided and expressed in to the
combination of N/4 point DFTs. This process is continued until we left with
2-point DFTs.
21. What is DIF radix-2 algorithm?
1. The radix 2 DIFFFT is an efficient algorithm for computing DFT in this the out
put sequence x(k) is divided in to smaller and smaller.
2. The idea is to break N point sequence in to two sequences ,x1(n) and x2(n)
consisting of the first N/2 points of x(n)and last N/2 points of x(n) respectively. Then we
find N/2 point sequences f(n) and g(nSimilarly).
3. The N/2 DFT s are divided and expressed in to the combination of N/4 point
DFT s. This process is continued until we left with 2-point DFTs.
22. What are the differences between DIT and DIF algorithms?
* For DIT the input is bit reversed and the output is in natural order, and in DIF
the input is in natural order and output is bit reversed.
* In butterfly the phase factor is multiplied before the add and subtract operation
but in DIF it is multiplied after add-subtract operation.
23. What is meant by in place in DIT and DFT algorithm?
An algorithm that uses the same location to store both the input and output
sequence is called in-place algorithm.
24. Differentiate DTFT and DFT
DTFT output is continuous in time where as DFT output is Discrete in time.
25. Differentiate between DIT and DIF algorithm
DIT Time is decimated and input is bi reversed format output in natural order
DIF Frequency is decimated and input is natural order output is bit reversed
Format.
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26. State the equation for finding the poles in chebyshev filter.
27. State the steps to design digital IIR filter using bilinear method.
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31. Why impulse invariant method is not preferred in the design of IIR filters other
Than low pass filter?
In this method the mapping from s plane to z plane is many to one. Thus there is
an infinite number of poles that map to the same location in the z plane, producing an
aliasing effect. It is inappropriate in designing high pass filters. Therefore this method is
not much preferred.
32. What is meant by impulse invariant method?
In this method of digitizing an analog filter, the impulse response of the resulting
digital filter is a sampled version of the impulse response of the analog filter. For
e.g. if the transfer function is of the form, 1/s-p, then
H (z) =1/1-e-pTz-1
33. What do you understand by backward difference?
One of the simplest methods of converting analog to digital filter is to
approximate the differential equation by an equivalent difference equation.
d/dt(y(t)/t=nT=(y(nT)-y(nT-T))/T
34. What are the properties of chebyshev filter?
1. The magnitude response of the chebyshev filter exhibits ripple either in the stop
band or the pass band.
2. The poles of this filter lies on the ellipse.
35. Give the Butterworth filter transfer function and its magnitude characteristics for
Different orders of filter.
36. Give the equation for the order N, major, minor axis of an ellipse in case of
chebyshev filter?
37. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method (IIM)
3. Bilinear transformation (BLT)
38. write down bilinear transformation.
s=2/T (z-1/z+1)
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The parallel logic unit is a second logic unit that executes logic operations on data
without affecting the contents of accumulator.
21. List the on chip peripherals in C5x.
The on-chip peripherals interfaces connected to the C5x CPU include
w. Clock generator
x. Hardware timer
y. Software programmable wait state generators
z. General purpose I/O pins
aa. Parallel I/O ports
bb. Serial port interface
cc. Buffered serial port
dd. Time-division multiplexed (TDM) serial port
ee. Host port interface
ff. User unmask able interrupts
22. What are the arithmetic instructions of C5x?
ADD, ADDB, ADDC, SUB, SUBB, MPY, MPYU
23. What are the shift instructions?
ROR, ROL, ROLB, RORB, BSAR.
24. What are the general purpose I/O pins?
Branch control input (BIO)
External flag (XF)
25. What are the logical instructions of C5x?
AND, ANDB, OR, ORB, XOR, XORB
26. What are load/store instructions?
LACB, LACC, LACL, LAMM, LAR, SACB, SACH, SACL, SAR, SAMM.
27. Mention the addressing modes available in TMS320C5X processor?
1. Direct addressing mode
2. Indirect addressing mode
3. Circular addressing mode
4. Immediate addressing
5. Register addressing
6. Memory mapped register addressing
28. Give the features of DSPs?
* Architectural features
* Execution speed
* Type of arithmetic
* Word length
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UNIT I INTRODUCTION
PART A
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
17.
18.
19.
20.
21.
22.
tabulation method.
Check whether the system defined by h (n) = [5 (1/2)n + 4 (1/3)n ] u(n) is stable.
Differentiate between analog, discrete, quantized and digital signals.
Differentiate between analog
and digital signals.
Differentiate between one dimensional and two dimensional signal with an example for
each.
28. Name any four elementary time domain operations for discrete time signals.
29. For the signal f (t) = 5 cos (5000t) + sin2 (3000t), determine the minimum sampling rate
for recovery without aliasing.
1 = 5000 = 2F1
2 = 3000 = 2F2
23.
24.
25.
26.
27.
F1 = 2.5 kHz
F2=1.5 kHz
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30. For the signal f (t) = cos2 (4000t) + 2 sin (6000t), determine the minimum sampling rate
for recovery without aliasing.
1 = 4000 = 2F1
2 = 6000 = 2F2
F1 = 2 kHz
F2 =3 kHz
Fmax = 3 kHz
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b = 5.18 = 6 bits
42. Determine the number of bits required in computing the DFT of a 1024 point sequence
with an SQR of 30 dB
The size of the sequence is N = 1024 = 210
SQR is X2 / q2 = 22b / N2
N2 = 220
10 log [X2 / q2] = 10 log [22b / 220]
PART - B
1. For each of the following systems, determine whether the system is static stable, causal,
linear and time invariant
a. y(n) = e x(n)
b. y(n) = ax(n) +b
c. y(n) = nk=n0 x(k)
d. y(n) = n+1k= - x(k
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e. y(n) = n x2(n)
f. y(n) = x(-n+2)
g. y(n) = nx(n)
h. y(n) = x(n) +C
i. y(n) = x(n) x(n-1)
j. y(n) = x(-n)
k. y(n) = x(n) where x(n) = [x(n+1) x(n)]
l. y(n) = g(n) x(n)
m. y(n) = x(n2)
n. y(n) = x2(n)
o. y(n) = cos x(n)
p. y(n) = x(n) cos 0n
2. Compute the linear convolution of h(n) = {1,2,1} and x(n) ={1,-3,0,2,2}
3. Explain the concept of Energy and Power signals and determine whether the following are
energy or power signals
a. x(n) = (1/3)n u(n)
b. x(n) = sin ( / 4)n
4. The unit sample response h(n) of a system is represented by
h (n) = n2u(n+1) 3 u(n) +2n u(n-1) for -5 n 5. Plot the unit sample response.
5. State and prove sampling theorem. How do you recover continuous signals from
its samples? Discuss the various parameters involved in sampling and
reconstruction.
6. What is the input x(n) that will generate an output sequence
y(n) = {1,5,10,11,8,4,1} for a system with impulse response h(n) = {1,2,1}
7. Check whether the system defined by h(n) = [5 (1/2)n +4(1/3)n] u(n) is stable?
8. Explain the analog to digital conversion process and reconstruction of analog signal from
digital signal.
9. What are the advantages and disadvantages of digital signal processing compared with
analog signal processing?
10. Classify and explain different types of signals.
11. Explain the various elementary discrete time signals.
12. Explain the different types of mathematical operations that can be performed on a discrete
time signal.
13. Explain the different types of representation of discrete time signals.
14. Determine whether the systems having the following impulse responses are causal and
stable
a. h(n) = 2n u(-n)
b. h(n) = sin n / 2
c. h(n) = sin n + (n)
d. h(n) = e2n u(n-1)
15. For the given discrete time signal
x (n) = { -0.5,0.5, for n = -2, -1
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1,
n=0
3, 2, 0.4
n > 0}
Find the convolution of x (n) = an u (n), a < 1 with h(n) = 1for 0 n N-1
Draw the analog, discrete, quantized and digital signal with an example.
Explain the properties of linearity and stability of discrete time systems with examples.
The impulse response of a linear time invariant system is h (n) = {1, 2, 1,-1}.
Determine the response of the system to the input signal x (n) = {1, 2, 3, 1}.
Determine whether or not each of the following signals are periodic. If a signal is
Periodic specify its fundamental time period.
i.
ii.
iii.
iv.
x(t) = 2 cos 3 t
x(t) = sin 15 t + sin 20 t
x(n) = 5 sin 2n
x(n) = cos (n/8) cos (n / 8)
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7.
8.
9.
10.
11.
12.
13.
14.
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PART B
1. Perform circular convolution of the sequence using DFT and IDFT technique
(8)
x1(n) = {2, 1,2,1} x2 (n) = {0,1,2,3}
2. Compute the DFT of the sequence x(n) = {1,1,1,1,1,1,0,0}
(8)
3. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIT FFT algorithm. Using the above compute the DFT of sequence x(n) =
{0.5,0.5,0.5,0.5,0,0,0,0}
(16)
4. State and prove the circular convolution property of DFT.Compute the circular
convolution of x(n) = {0,1,2,3,4} and h(n) = {0,1,0,0,0}
(8)
5. Perform circular convolution of the sequence using DFT and IDFT technique
(8)
x1(n) = {1,1,2,1} x2 (n) = {1,2,3,4}
6. Compute the DFT of the sequence x(n) = {1,1,1,1,1,1,0,0}
(8)
7. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute
its 8 point DFT of x(n) by radix-2 DIF-FFT (16)
8. Compute 5 point circular convolution of x1(n) = (n) + (n-1)- (n-2) - (n-3) and x2(n) =
(n) (n-2)+ (n-4)
(8)
9. Explain any five properties of DFT.
(10)
10. Derive DIF FFT algorithm. Draw its basic butterfly structure and compute the DFT x(n) = (1)n using radix 2 DIF FFT algorithm.
(16)
11. Perform circular convolution of the sequence using DFT and IDFT technique
(8)
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1}
12. Compute the DFT of the sequence x (n) = 1/3 (n) 1/3 (n-1) +1/3 (n -2) (6)
13. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) = 2 sin n
(16)
/ 4 for 0 n 7
14. What is circular convolution? Explain the circular convolution property of DFT and
compute the circular convolution of the sequence x(n)=(2,1,0,1,0) with
itself
(8)
15. Perform circular convolution of the sequence using DFT and IDFT technique
(8)
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1}
(4)
16. i) Compute the DFT of the sequence x (n) = (-1)n
ii) What are the differences and similarities between DIT FFT and DIF FFT
algorithms?
(4)
17. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) = cos n / 4
(16)
for 0 n 7
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(6)
(6)
20. Explain the procedure for finding IDFT using FFT algorithm
(6)
21. Compute the output using 8 point DIT FFT algorithm for the sequence
x(n) = {1,2,3,4,5,6,7,8}
(16)
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1. An analog filter has a transfer function H(s) = 1 / s+2. Using impulse invariance
method, obtain pole location for the corresponding digital filter with T = 0.1s.
2. What is frequency warping in bilinear transformation?
3. If the impulse response of the symmetric linear phase FIR filter of length 5 is
h (n) = {2,3,0,x,y}, find the values of x and y.
4. What is prewarping? Why is it needed?
5. Find the digital transfer function H (z) by using impulse invariance method for the analog
transfer function H(s) = 1 / s+2.
6. What are the different structures of realization of FIR and IIR filters?
7. What are the methods used to transform analog to digital filters?
8. State the condition for linear phase in FIR filters for symmetric and anti symmetric
response.
9. Draw a causal FIR filter structure for length M= 5.
10. What is bilinear transformation? What are its advantages?
11. Write the equation of Barlett (or) triangular and Hamming window.
12. Write the equation of Rectangular and Hanning window.
13. Write the equation of Blackman and Kaiser window.
14. Write the expression for location of poles of normalized Butter worth filter.
Pk = c ej (N+2k +1) / 2N
Where k = 0, 1, . (N-1) and for a normalized filter c = 1 rad / sec
15.
16.
17.
18.
19.
20.
21.
22.
23.
24.
25.
26.
27. State the condition for a digital filter to be causal and stable.
28. What are the desirable characteristics of windows?
29. Give the magnitude function Butterworth filter. What is the effect of varying the order of N
on magnitude and phase response?
30. List out the properties of Butterworth filter.
31. List out the properties of Chebyshev filter.
32. Give the Chebyshev filters transfer function and draw its magnitude response.
33. Give the equation for the order N and cut off frequency c of Butterworth filter
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34. Why impulse invariance method is not preferred in the design of IIR filter other than low
pass filters?
35. What are the advantages and disadvantages FIR filters?
36. What are the advantages and disadvantages IIR filters?
37. What is canonic structure?
If the number of delays in the structure is equal to the order of the difference
equation or order of transfer function, then it is called canonic form of realization.
38. Compare Butterworth and Chebyshev filters.
39. What are the desirable and undesirable features of FIR filters?
40. What are the design techniques of designing FIR filters?
Fourier series method
Windowing technique
Frequency sampling method
PART B
1. With suitable examples, describe the realization of linear phase FIR filters
(8)
2. Convert the following analog transfer function H(s) = (s+0.2) / [(s+0.2)2 + 4] into
equivalent digital transfer function H (z) by using impulse invariance method assuming
T= 1 sec.
(8)
3. Convert the following analog transfer function H(s) = 1 / (s+2) (s+4) into
equivalent digital transfer function H (z) by using bilinear transformation with T = 0.5
sec.
4. Convert the following analog transfer function H(s) = (s+0.1) / [(s+0.1)2 + 9] into
equivalent digital transfer function H (z) by using impulse invariance method assuming
T= 1 sec.
(8)
5. Convert the following analog transfer function H(s) = 2/ (s+1) (s+3) into
equivalent digital transfer function H (z) by using bilinear transformation with T = 0.1
sec.Draw the diect form II realization of digital filter.
(8)
6. Design a high pass filter of length 7 samples with cut off frequency of 2 rad / sec
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(16)
(16)
8. Describe the effects of quantization in IIR filter. Consider a first order filter with
difference equation y (n) = x (n) + 0.5 y (n-1).Assume that the data register length is 3
bits plus a sign bit. The input x (n) = 0.875 (n). Explain the limit cycle oscillations in
the above filter, if quantization is preferred by means of rounding and signed
magnitude representation is used.
(16)
9. With a neat sketch explain the architecture of TMS 320 C54 processor.
(16)
(16)
(16)
12. Discuss about the window functions used in design of FIR filters
(8)
13. Obtain the cascade and parallel realization of system described by difference equation
y(n) = -0.1 y(n-1) + 0.2 y(n-2) + 3x(n) +3.6 x(n-1) + 0.6 x(n-2)
(10)
14. Design a digital Butterworth filter satisfying the following constraints with T= 1 sec,
using Bilinear transformation.
0.707 H () 1.0, 0 /2
H () 0.2, 3/4
15. Design a digital Chebyshev filter satisfying the following constraints with T= 1 sec,
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(16)
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(16)
18. Draw and explain cascade form structure for a 6th order FIR filter.
(6)
(10)
20. Derive an expression between s- domain and z- domain using bilinear transformation.
Explain frequency warping.
(10)
21. Draw the structure for IIR filter in direct form I and II for the following transfer
Function H (z) = (2 + 3 z-1) (4+ 2 z-1 +3 z-2) / (1+0.6 z-1) (1+ z-1+0.5 z-2) (10)
22. Design a filter with
Hd() = e-j2
=0
- /4 /4
/4
(16)
(8)
- 3/4 3/4
3/4
(16)
25. Using the bilinear transformation and a low pass analog Butterworth prototype,
design a low pass digital filter operating at a rate of 20 KHz and having pass band
extending to a 4 KHz with a maximum pass band attenuation of 0.5 dB and stop band
starting at 5KHzwith a minimum stop band attenuation of 10 dB.
(16)
26. Using the bilinear transformation and a low pass analog Chebyshev type I prototype,
design a low pass digital filter operating at a rate of 20 KHz and having pass band
extending to a 4 KHz with a maximum pass band attenuation of 0.5 dB and stop band
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(16)
27. Obtain the cascade realization of linear phase FIR filter having system function
H(z) = ( 1+1/2 z-1 + z-2) (2 + z-1 +2z-2) using minimum number of multipliers.(8)
28. Design an ideal Hilbert transformer having frequency response
H(ej) = j
= -j
for - 0
for
D () = ej3
= 0 for
for
-3 / 4 3 / 4
3 / 4
33. Design an ideal differentiator for N=9 using Hanning and triangular window
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(12)
2. What are the different buses of TMS 320 C54 processor? Give their functions. (4)
3. Explain the function of auxiliary registers in the indirect addressing mode to point the
data memory location.
(8)
(8)
(8)
(8)
7. Explain the characteristics of a limit cycle oscillation with respect to the system
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described by the equation y (n) = 0.95 y (n-1) + x (n).Determine the dead band of the
filter.
(10)
8. Draw the product quantization noise model of second order IIR filter.
(6)
9. In a cascaded realization of the first order digital filter, the system function of the
individual section are H19z) = 1 / (1-0.9 z-1) and H2(z) = 1 / (1-0.8z-1). Draw the
product quantization noise model of the system and determine the output noise
power.
(16)
(16)
11. Give a detailed note on Direct memory Access controller in TMS 320 C54x
processor.
12. Find the effect of quantization on the pole locations of the second order IIR filter
Given by H(z) = 1 / (1-0.5z-1) (1- 0.45 z-1) when it is realized in direct form I and in
cascade form. Assume a word length of 3 bits.
13. Determine the variance of the round off noise at the output of the two cascade
realizations of the filters with system functions H1 (z) = 1 / 1-0.5 z-1
H2 (z) = 1 / 1- 0.25 z-1
Cascade I, H (z) = H1 (Z) H2 (z)
Cascade II, H (z) = H2 (z) H1 (z)
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