Wednesday, October 28, 2009
DIGITAL SIGNAL PROCESSING |
PART-A (10 x 2 = 20 Marks) |
1. Differentiate between analog and digital signal. Why Digital signal processing is widely |
used than analog signal processing. |
2. State Shannon’s Sampling theorem.3. Determine the Z-transform of (1/2)n[ u[n]-u[n- |
8]] and indicate its ROC. |
4. Compare FIR and IIR filter. |
5. What are the advantages of linear phase characteristics? Which systems exhibit linear |
phase? |
6. Show that the system described by the difference equation is an all pass system3 y(n) – |
y(n-1) = -x(n) + 3x(n-1) |
7. Mention few application areas where speech coding is required. |
8. Explain the circular addressing mode of DSP processor |
9. Find the DFT of the signal x(n)= {1,3,5,7}. |
10. Distinguish between recursive and non-recursive realizations of filters. |
PART-B (5x16 = 80 Marks) |
11.i) Show that Z-Transform of x*(n) is X*(z*). (4) |
ii) Consider a linear shift-invariant discrete system with input x(n) and output y(n) for |
whichy(n-2) – 2.5 y(n-1) +y(n) =x(n)By considering the pole-zero pattern associated with |
the difference equation, determine the three possible choices for the unit-sample |
response of the system. Comment on the stability of the system in each case. (12) |
12.a)i) Find the DFT of the sequence {1,1,1,1,2,2,2,2} using radix-2 Decimation-in-Time |
FFT. Sketch the magnitude and phase plot. (12) |
ii) What is the need for FFT? (4) |
(OR) |
12.b)i) Find the DFT of the sequence {1,1,1,1,2,2,2,2} using radix-2 Decimation-in- |
Frequency FFT. (12) |
ii) Write about over lap save method. (4) |
13.a) The specification of the desired low pass filter are:Amin = 22 dB and Amax = 3 dB ?p |
= 0.2? and ?s = 0.4?Design a Butterworth digital filter using Bilinear Transformation. (Amin |
and Amax are attenuation) (16) |
(OR) |
13.b) Design and also realize a high pass FIR filter with a cutoff frequency of 1.3 rad/sec |
and N=9. (16) |
14.a)i) Perform the linear convolution of (1/4)n u(n) and (1/2)n u(n). (6) |
ii) Is it possible to perform linear convolution through circular convolution. If so |
how?(2)iii) Find the Discrete Fourier Series of the following periodic sequence. (8)(OR) |
14.b)i) Explain about the Frequency Transformation that will be adopted in IIR filter |
design. (4) |
ii) The specification of the desired low pass digital filter areAmin = 12.4 dB and Amax = |
0.915 dB ?p = 0.25? and ?s = -0.5?Design a Chebyshev digital filter using impulse invariant |
transformation. (Amin and Amax are attenuation). (12) |
15.ai) Highlight the special blocks of the Digital Signal Processor Architecture over the |
regular Micro-Controller based Architectures. (16) |
(OR) |
15.b)i) Explain how bit-reversal is achieved in the Texas based DSP Processor. (8) |
ii) Show that FFT can be evaluated with lesser machine cycles using DSP processor |
compared to any of Micro-controller. (8) |
http://www.ziddu.com/download/7201561/dsp3.pdf.html
DIGITAL SIGNAL PROCESSING |
PART A – (10 X 2 = 20 marks) |
1. State Sampling Theorem? |
2. Find the Poles of the system. |
3. Find the DFT of the sequence x(n) ={1, 1, 0, 0 } DFT is obtained by FFT. |
4. Calculate the number of multiplications needed in the calculation of 512 point |
radix2FFT when compared to Direct DFT? |
5. What are the properties that are maintained same in the transfer of analog filter into |
adigital filter? |
6. What is warping effect? |
7.Draw the direct from realization of FIR system? |
8.What are the describe features of a window function? Name the different types |
ofwindowing function? |
9. What is truncation? |
10. Draw a sample/ hold circuit and explain its operations? |
PART B – (5 X 16 = 80 marks) |
11. (a) (i) For each of the following discrete time system, determine whether or not |
thesystem is Linear Time, Variant, Causal and Stable? |
11.(a) (ii) Determine the transfer function, magnitude & phase response, impulseresponse |
for the system. |
11. (b) (i) Find the Ztransformof1) x(n) = 2 n u (n2)2) x(n) = n 2 u (n) |
11.(b)(ii) Use convolution to find x(n), given |
11.(b) (iii) Determine the cross correlation values of the sequence x1(n) ={ 1, 2, 3, 4 |
}x2(n) = {4, 3, 2, 1}12. (a) (i) Compute linear and circular convolution of the two |
sequencex1(n) ={ 1, 2, 2, 2 } and x2(n) = {1, 2, 3, 4} |
12.(a) (ii) Compute the FFT using DIT algorithm for the sequence x(n) = {1, 2, 3, 4, 4, 3,2, |
1 } and draw the corresponding flow diagram. |
12.(b) (i)Prove that multiplication DFT’s of 2 sequence is equivalent to the DFT of |
thecircular convolution of the 2 sequence in time domain? |
12.(b)(ii) Discuss in detail the use of FFT algorithm , in linear filtering? |
13.(a) Find H(z) using impulse invariant technique for the analog system function. |
13 (b) (i) Obtain the direct form II, Cascade form parallel form structures for the system? |
13.(b) (ii) Design a butterworth filter using linear transformation that satisfies |
thefollowing constraint? |
14(a) The desired response of a low pass filter is? |
14.(b) Explain the Type I & Type 2 design of FIR filter using frequency samplingTechnique? |
15.(a) The output of A/D converter is applied to a digital filter withsystem function find |
the o/p noise power for the digital filter when the inputsignal is quantized to 8Hz. |
15.(b)(i) A digital system is characterized by the difference equation y(n)=0.95 |
y(n1)+x(n). Determine the dead band the system when x(n) =0 and y(n) = 1315. |
(b) |
(ii) With neat diagram explain the analysis and synthesis part of a recorder in detail? |
DIGITAL SIGNAL PROCESSING, |
PART – A (10 X 2 = 20 MARKS) |
1. State and prove the convolution property of Z transform. |
2. Check the system is linear or not y(n) = x(n)+ay(n-1) |
3. Write equations for finding DFT and IDFT using Z transform. |
4. Draw the radix 2 butterfly structure for DIF |
5. Draw the implementation for the generalized for IIR filter using direct form II. |
6. Explain how the addition and multiplication of (H1, H2) impulse responses |
implemented in filter design |
7. Write equations for Hanning and Blackman window. |
8. Why frequency prewarping procedure is adopted in the design of IIR filter? |
9. Write two advantages of musical sound processing and briefly explain. |
10. Explain the effects due to upsampling. |
PART - B (5 x 16 = 80 Marks) |
11.i) The impulse response of a linear TI system is h(n) = {1, 0, 1, -1}. Find the response |
of the system to the input signal x(n) = {1, 0, 2, 1}. |
ii) Check whether the system y(n) = x(n) – x(n-1) is LTI and stable. |
12.a) Develop and draw the 8 point radix-2 DIT FFT algorithm for DFT computation. |
(OR) |
12.b) Compute the DFT of the following sequence x(n) = 0 0£ n £ 2= 1 3£ n £ 6= 0 n=7 Plot |
magnitude and phase spectra |
13.a) Design a LPF with following specifications. Use Hamming window and at least 8 |
points. |
(OR) |
13.b)i) Obtain H(z) from H(s) when T = 1 sec. |
ii) Design a digital BPF using w1 & w2 as cutoff frequencies |
14.a)i) Perform the following using Floating Point arithmetic.1.5 x 1.75 and 1.5 x 1.75 |
ii) Realize the following H(z) given byusing cascade and Parallel form with Direct form-I. |
(OR) |
14.b)i) What is meant by quantization error? Explain briefly. |
ii) Realize the following filter using cascade technique with DF-I and DF-II. |
15.a) Briefly explaina. Interpolatorb. Decimatorc. Effects due to sampling rate conversion |
(OR) |
15.b)i) Write a note on Musical sound processing |
ii) Explain how the data compression is achieved in speech signal and discuss a technique |
to check the quality. |
http://www.ziddu.com/download/7201562/dsp2.pdf.html