Documente Academic
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Documente Cultură
PART-A
UNIT I
1. Distinguish between DFT and DTFT. (NOV 2011)
2. What is zero padding? What are its uses? (NOV 2011)
3. State the advantage of FFT over DFTs. (MAY 2011)
4. What is meant by bit reversal? (MAY 2011)
5. What is radix 2 FFT algorithm? (nov/dec 2009)
UNIT II
1. List the properties of chebyshev filter. (NOV 2011)
2. Draw the direct form structure of IIR filter. (NOV 2011)
3. Why do we go for analog approximation to design a digital filter? (MAY 2011)
4. Give any two properties of chebyshev filters. (MAY 2011)
5. Find the digital transfer function H(z) by using impulse invariant method for the analog
transfer function H(s) = 1/ (s+2). Assume T=0.1 sec. (Nov 2007)
6. State the condition for a digital filter to be causal and stable.
(May 2007)
7. Find the digital transfer function H (z) by using impulse invariant method for the analog
transfer function H(s) = 1/(S+2). Assume T=0.5sec.
8. Give any two properties of Butterworth filter and chebyshev filter.
(Nov/Dec 2006, May/June 2006, Apr 2005 & Nov 2004)
9. Mention any two procedures for digitizing the transfer function of an analog filter.
(Nov 2006)
10. what are the parameters that can be obtained from the chebyshev filter
specification? (Nov 2006/May 2007)
11. Give the equation for the order N, major, minor and axis of an ellipse in case of
chebyshev filter. (Nov 2005)
12. What are the advantages and disadvantages of bilinear transformation?
(May 2006)
13. Define Hanning and Blackman window functions. (May 2006)
14. Write the magnitude function of Butterworth filter. What is the effect of varying order of
N on magnitude and phase response? (Nov 2005)
15. What is impulse invariant mapping? What is its limitation? (Apr/May 2005)
16. What is linear phase? What is the condition to be satisfied by the impulse response in
order to have a linear phase? (Apr 2005 & Nov 2003)
17. What is frequency warping? (Nov2004 & May 2007)
18. What are the limitations of impulse invariant mapping technique? (Apr2004)
19. Give the transform relation for converting low pass to band pass in digital domain.
(Apr 2004)
20. Give the bilinear transformation. (Nov2003)
UNIT III
11) What is wraping effect? What is its effect on magnitude and phase response?
(May 2009)
12) What condition on the FIR sequence h(n) is to be imposed in order that this filter
can be called a linear phase filter? (May
2009)
13) Show that the filter with h (n) = [-1, 0, 1] is a linear phase filter.
ii. (Nov 2008,May 2007)
14) What is prewarping? (Nov 2003,2008)
15) What are the merits and demerits of FIR filters? (Nov 2005 & April 2008)
16) What is the relation between analog and digital frequency in impulse invariant
transformation? (Apr
2008)
17) In the design of FIR digital filters, how is Kaiser Window different from other
windows? (Nov 2007)
18) Compare FIR and IIR filters. (May 2007)
19) What is the condition satisfied by linear phase FIR filter? (Nov/Dec 2003 &
May 2007)
20) Obtain the block diagram representation of a FIR System. (Nov 2006)
21) What are the desirable and undesirable features of FIR Filters? (May2006)
22) Mention the necessary and sufficient condition for linear phase characteristics in
FIR filter. (Nov 2005)
23) List the characteristics of FIR filters designed using window functions.(Nov
2004)
24) Give the Kaiser Window function. (Apr 2004)
25) What are the steps involved in Bartlett method?
26) What are the steps involved in Welch method?
27) Define Blackman and turkey method?
28) Determine the frequency resolution of the bartlett method of power spectrum
estimates for a quality factor Q=15. Assume that the length of the sample
sequence is 1500. (Apr 2008)
29) Define the terms autocorrelation sequence and power spectral density
iii. (Apr 2007)
30) Define power spectral density and cross spectral density.
iv. (May2007)
UNIT IV
1. What is overflow oscillations? (NOV 2011)
2. What are the advantages of floating point arithematic? (NOV 2011)
3. What is meant by fixed point arithmetic? Give example (MAY 2011)
4. Explain the meaning of limit cycle oscillation (MAY 2011)
UNIT V
1. What is multirate signal processing? (NOV 2011)
2. What is meant by down sampling and up sampling? (NOV 2011)
3. State the various applications of DSP (MAY 2011)
4. What is echo cancellation? (MAY 2011)
5. What is the need for multirate signal processing ?
PART B
UNIT I
4. (i) Suppose you have a number of eight-point FFT chips. Suggest a scheme to
interconnect four chips to compute a 32-point DFT.
5. compute the eight point DFT of the sequence x(n) = {0.5, 0.5, 0.5, 0.5, 0, 0, 0, 0}
using the inplace radix-2 DIT algorithm. (May 2009)
(i) discribe the following properties of DFT.
(1) convolution
(2) time reversal
(3) time shift
(4) periodicity
(ii) compare the computational complexity of direct of direct DFT
computation of a sequence with n=64 (nov/dec 2009)
6. (i) explain decimation in time FFT algorithm for N=8 .
8. (a) Obtain an 8-point DIT FFT flow graph from first principles.
(May 2007,Nov 2008)
(b) Using the above flow graph compute DFT of x(n) = cosn /4 for n=0,1,,7(Nov
2008)
9. (a) Discuss in detail the important properties of the Discrete Fourier Transform
b) find the 4 point DFT of the sequence x(n) = Cos n/4 (Apr 2008)
10. Compute an 8 point DFT using DIF FFT radix 2 algorithm X(n) = {1,2,3,4,4,3,2,1}
(May 2006 & Apr 2008)
11. (a) Obtain an 8-point DIF FFT flow graph from first principles.
(v) (b) Using the above flow graph compute DFT of x(n) = cosn /4 for
n=0,1,,7 (Nov 2007,April 2008)
12. Two finite duration sequences are given by X(n) = sin(n /2 for n=0.1.2.3 and h(n)
=2n for n=0,1,2,3 find circular convolution using DFT method.
(Nov 2007)
18. Draw the butterfly diagram using 8 point DIT FFT for the following sequences.
(i) X(n) = { 1,0,0,0,0,0,0,0}
(May2007)
(ii) Compute the DFT of each of the following
1. x(n) = (n-n0)
2. y(n) = x1(n) x2(n) (May 2007)
19. A DFT program is available, how will you this to compute inverse DFT.
(May 2007)
20. Two real signals of x(n) and y(n) are of length M. find the FT of x(n) and y(n) with
minimum computation. (May2007)
20.Compute the DFT of the sequence,x(n)={1,0,1,0,1,0,1,0} and hence find X(2).
(APR 2005 CS)
21.Draw the FFT flowchart for radix 2,DIT algorithm. Assume N=8.(APR 2005 CS)
22. Find the 8 pt DFT of the sequence (APR 2005 IT)
x(n)={ 1 0<n<7
0 otherwise (using DIT FFT )
23. Compute the 8 pt DFT of the sequence (NOV 04 IT)
x(n)={0.5,0.5,0.5,0.5,0,0,0,0} using DIT FFT
26.Determine 8pt DFT of x(n)=1 for 3<n<3 using DIT-FFT algorithm. (APR 04 IT)
UNIT II
1) (a) design a butterworth filter using impulse invariance method satisfying the
constraints.
Assume T=1sec.
0.8 |H(ejw)| 1 ; 0 w 0.2
|H(ejw)| 0.2 ; 0.6 w (16)
(b)obtain the direct form I ,direct form II and cascade form realization of the
following system functions.
y(n)=0.1 y(n-1)+0.2 y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2) (16)
UNIT III
1) (a)Design an ideal differentiator with frequency response,
H(ejw)=jw ; -w
Using hamming window with N=8. (16)
(b) deisgn an ideal high pass filter using hanning window with a frequency response
Hd(ejw) = 1; /4 |w|
= 0; |w| /4 .
Assume N=11. (16)
2 (a) Realise the system function H(z)=[ ]z+1+[ ] by linear phase FIR structure.
(16)
(ii) derive thee condition for linear phase FIR filters. (nov/dec 2009)
UNIT IV
1.(a)With respect to finite word length effects in digital filters, with examples discuss
about
(i) Over flow limit cycle oscillation (10)
Signal scaling (6)
(b) (i) distinguish between fixed point and floating point arithematic . (6)
(ii) consider a second order iir filter with
H(z)=( 1.0)/(1-0.5z-1)(1-0.45z-1)
2. Find the effect on quantization on pole locations of the given system function in direct
form and in cascade form .Assume b=3 bits. (10)
(a) Explain the quantization process and errors introduced due to quantization.
(16)
(b). i) Explain how reduction of product round-off error is achieved in digital filters.
(8)
ii) Explain the effects of coefficient quantization in FIR filters.
(8)
3. (i) Realize the first order transfer function H(z)=1/(1-az-1) and draw its quantization
model. Find the steady state noise power due to product round off.
(ii) Explain in detail about the zero-input limit cycle oscillations due to finite word
length of registers. (May 2009)
4.(i) What is the need of signal scaling? How the scaling is performed?
(ii) for a second order digital filter H(z)=1/(1-2rcosz-1+r2z-2); |r| <1.0. Draw the
direct form II realization and find the scale factor S0 to avoid overflow. (May 2009)
5.(i)Consider (b+1)-bit (including sign bit) biplar ADC. Obtain an expression for signal to
quantization noise ratio.State the assumptions made. (Nov 2008)
(ii)Consider the truncation of negative fraction numbers represented in (+1) bit
fixed point binary form including sign bit. Let (-b) bits be truncated. Obtain the
range of truncation errors for signed magnitude. 2s complement and 1s complement
representations of the negative numbers. (Nov 2007,Nov 2008)
(ii)An IIIR causal filter is defined by the difference equation y(n) = x(n)-
0.96y(n).The unit sample response h(n) is computed such that the computed values
are rounded to one decimal place. Show that under these stated conditions, the filter
output exhibits dead band effect. What is the dead band range? (Nov 2008)
6. Discuss in detail the truncation error and Round-off error for sign magnitude and twos
complement representation. (Apr 2008)
7. Explain the quantization effects in converting analog signal into digital signal.
(Apr 2008)
8. (a) A digital system is characterized by the difference equation Y(n) = 0.9y(n-
1)+x(n) with x(n) = 0 and initial condition Y(-1) = 12. Determine the dead band of
the system.
Refer book: Digital signal processing by Ramesh Babu.(pgno:MQ.16)
(b) what is meant by the co-eefficient quantization? Explain. (Apr 2008)
9. An 8-bit ADC feeds a DSP system characterized by the following trnafer function
H(z) = 1/(z+0.5) estimate the steady state quantization noise power at the output of
the system. (Nov 2007)
11. An IIIR causal filter has the system function H(z) = z / (z-0.97) assume that the input
signal is zero-valued and the computed oputput signal values are rounded to one decimal
place. Show that under these stated conditions, the filter output exhibits dead band
effect. What is the dead band range? (Nov 2007)
12. (i)The input to the system y(n) = 0.999y(n-1)+x(n) is applied to an ADC. What is the
power produced by the quantization noise at the output of the filter if the input is
quantized to (i) 8 bits (ii) 16 bits (May 2007)
(ii)convert the following decimal number into binary: (May 2007)
a. (20.675)10
b. (120.75)10
13. consider the transfer function H(z)=H1(z)H2(z) where H1(z) = 1/(1-a1z-1) and H2(z) = 1/
(1-a2z-1) . find the output round off noise power. Assume a 1 = 0.5 and a2 = 0.6 and find
output round off noise power. (May 2007,Nov 2006)
14. explain the characteristics of a limit cycle oscillation with respect to the system
described by the difference equation y(n) =0.95y(n-1)+x(n). determine the dead band of
the filter. (Nov2006)
15. Draw the product quantization noise model of second order IIR system
.(Nov 2006)
16. Expain the effect of finite word length effects. (APR 05 EC)
17. Derive the steady state noise power at the output if an LTI
system due to quantization at the input. (NOV 04 EC)
18. Explain about fixed point and floating point representation. (NOV 04 EC)
19. Discuss limit cycles in digital filters. (NOV 03 EC)
20.Draw the quantization noise model for a second order
system with system function. (APR 05 EC)
H(z) = 1
------------------------------
1 - 2rcos0 z-1 + r2 z-2
Determine the steady state noise.
21.Explain coefficint quantization effects in direct form realization of IIR filter.
(APR 04 EC)
24.Explain the characteristics of a limit cycle oscillation with respect to the system
described by the difference equation y (n) =0.95y (n-1) +x (n)
Determine the dead band of the filter. (NOV05EC)
H(z) = 1
------------------------ (NOV05EC)
1 - 0.9 z-1+ 0.2 z-1
UNIT V
1. (a) for the multirate system shown in figure , find the relation between x(n) and y(n) (16)
x(n)
z-1 z z
z-1
y(n)
z z
+
(OR)
(b) explain the efficient transversal structure for decimeter and interpolator. (16)
2.(a) (i) Explain how various sound effects can be generated with the help of DSP.
(10)
(ii) State the applications of multirate signal processing
(6)
(b) (i) Explain how DSP can be used for speech processing. (8)
c)Show that the spectrum in part (b) is simply the Fourier transform of x(2n).
4. The sequence x(n) is obtained by sampling an analog signal with period T. From this
signal a new signal is derived having the sampling period T/2 by use of a linear interpolation
method described by the equation.
a)Show that this linear interpolation scheme can be realized by basic digital signal
processing elements.
0, otherwise
0, otherwise
5. An analog signal xa (t) is bandlimited to the range 900 F 1100Hz. It is used as an input
to the system shown in Fig. In this system, H() is an ideal low pass filter with cutoff
frequency Fc = 125Hz.
a)Determine and sketch the spectra for the signals x(n), w(n), v(n), and y(n).
b)Show that it is possible to obtain y(n) sampling xu (t) with period T=4 millisec.Your
browser may not support display of this image.
6.Design a decimator that downsamples an input signal x(n) by a factor D=5. Use the
Remez algorithm to determine the coefficients of the FIR filter that has 0.1 dB ripple in the
passband (0 /5) and is down by at least 30dB in the stopband. Also determine the
corresponding polyphase filter structure for implementing the decimator.
7.Consider the two different ways of cascading a decimator with an interpolator shown in
Fig.
a)If D=1, show that the outputs of the two configurations are different. Hence, in general,
the two systems are not identical.
b)Show that the two systems are identical if and only if D and I are relatively prime.
8.Design a two stage decimator for the following specifications
D=100
Passband: 0F50
Transition band: 50F55
Input sampling rate: 10,000Hz
Ripple: 1 = 10-1 , 2 = 10-3