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DSP IMP Questions

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Siddu Charan Gurram
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52 views

DSP IMP Questions

Important links

Uploaded by

Siddu Charan Gurram
Copyright
© © All Rights Reserved
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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SRINIVASA INSTITUTE OF TECHNOLOGY &SCIENCE :: KADAPA

Digital Signal Processing


Question Bank
UNIT-1

2MARKS:

1.If x(n) = {1, 2, 3, 4, 6, 7, 8, 9} and DTFT [x(n)] = X(ejω), then find X(ejω) at ω = 0.

2.Match the following signals and operations involved.

(i) 2x(2n) (ii) x(-n/2) (iii) 2x(-n) (iv) x(2n+3)

(a) Time shifting. (b) Time scaling. (c) Folding. (d) Amplitude scaling. (e) Time shifting.

3.Draw the block diagram of digital signal processing system.

4.What is the condition for z-transform to exist?

5. State the classification of discrete time signal.

6.Define causal system and also time invariant system.

10 MARKS:

1.How to analyze the discrete system T[x(n)] = y(n) for linearity and shift invariance? Explain with a
suitable example.

2.Draw the graphical representation and sequence form of a discrete time signal
∞
x(n) = r(n+2) -r(n-3) - 5u(n-4) and also Evaluate the summation ,∑ x (n) where r(n) is unit ramp
n=0

sequence.
3. How to examine the discrete system having the rational system function H(z) for the causality and
stability? Explain with a suitable example.

4. (i) Frequency response.


(ii) Magnitude response and
(iii) Phase response of a discrete system having LCCDE.
(𝑛−1)/2(𝑛−1)=𝑥(𝑛)+1/2𝑥(𝑛−1).
5. Explain the classification of discrete-time signals
6. Determine the impulse response h(n) for the system described by the second order difference equation
y(n) – 2y(n-1) = x(n) + x(n-1).
7. Explain in detail about the classification of Discrete Time Systems.
8. a) Determine the convolution sum of following two sequences:
x(n) = {3,2,1,2}; h(n) = {1,2,1,2}
b) State and prove the conditions for causality & stability of an LTI system
UNIT-2

2 MARKS:

1.What is the magnitude of a phase factor164W?

2.How many number of complex adders and complex multipliers required to compute 64-Point DFT of a
sequence in direct DFT?

3.What is decimation in frequency FFT?

4.State any two properties of DFS.

5.Find the IDFT of X (K) = {1, 1, 1, 1}.

6.What are the differences and similarities between DIF and DIT algorithms?

7.What is zero padding? What are its uses?

10 MARKS:

1.Discuss the computational process of N-point DIT radix-2 FFT algorithm, hence draw the 8-point
butterfly structure by indicating the samples of input and output sequences

2.Compute the 4-point IDFT of a sequence X(k) = {10, -2+2j, -2, -2-2j}.
3. Define IDFT of a N-point sequence X(k). Show that the sequence x(n) is periodic with a periodic of N
samples, if X(k) is a finite duration sequence with a duration of N samples over the range 0 ≤ k ≤ N-1.
Given that the N-point IDFT[X(k)] = x(n).
4. Apply DIF radix-2 FFT algorithm to compute the 8-Point DFT of a sequence
x(n) = {1, 0, -1, 0, 1, 0, -1, 0}.
5.Given X(k) = {36,−4+𝑗𝑗9.656,−4+𝑗𝑗4,−4+𝑗𝑗1.656,−4,−4−𝑗𝑗1.656,−4−𝑗𝑗4,−4−𝑗𝑗9.656},
find x(n).
6.State and prove any two properties of DFT.
7.Given x(n) = {1,2,2,3,3,2,2,1}. Find X(k) using DIT FFT algorithm.
8.Compute 8-poin DFT of the sequence x(n)= {1,1,1,1,1,1,0,0}
9.Compute DFT of the sequence x(n)={1,2,3,4,4,3,2,1} using DITFFT algorithm
10.Compute IDFT of the sequence x(n)={ 7,-0.707-j0.707,-j, 0.707-j0.707,1, 0.707+j0.707,j, -
0.707+j0.707}
11. Develop an 8-point DIF-FFT algorithm. Draw the signal flow graph. Determine the DFT of the
following sequence, x(n)= {1,1,1,0,0,1,1,1}
12. Explain about decimation in time FFT algorithm.
13. Explain about decimation in frequency FFT algorithm
14. Explain the radix 2 FFT – DIT algorithm for the computation of DFT of the given 8 points sequence.
15.Find the DFT of the sequence x(n) defined by:
x(n) = 1 for 2 ≤ n ≤ 6
= 0 for n = 0, 1 and 7.
16. Find the circular convolution using DFT and IDFT of the sequence 𝑥1 (𝑛) = {4,3,1,2} and 𝑥2 (𝑛) =
Use DIF algorithm. Give all intermediate results

{1,3,5,3}.

UNIT-3

2 MARKS:

1.Match the following:

(a) Impulse Invariant Transformation (b) Bilinear Transformation.

(i) It is many-to-one mapping; (ii) It is one-to-one mapping.

(iii) Relation between analog and digital frequency is linear.

(iv) Relation between analog and digital frequency is nonlinear.

(v) Aliasing problem, (vi) Frequency warping problem.

2.Which of the following statement is true for impulse invariant transformation?

(i) It is a one to one mapping.

(ii) Relation between analog and digital frequency is ω = ΩT.

(iii) Transform ⇒1aTze11as1−−→−.

(iv) Suffering from frequency warping.

3.What are the different design techniques available for IIR filters?

4.What is the main advantage of direct form II realization when compared to direct form I realization?

5. What are the advantages and disadvantages of Bilinear transformation

6. Using Bilinear transformation, find H(z) from H(s) = 2/ [(S+1) (S-1)] with T = 1 sec.

7. Explain cascade form structure for IIR systems

10 MARKS:

1.Discuss the various properties of Butterworth and Chebyshev IIR approximation methods.

2.Design a digital filter by converting the transfer function of analog filter H(s) into digital filter H(z) by
using impulse invariant transformation method with a sampling period of T = 0.1sec. Given.
2
H(s) = 2
s + 3 s+ 2

3.Design a digital low pass filter by using IIR Butterworth approximation and Bilinear transformation
method by taking the sampling period of T = 0.1sec to satisfy the following specifications.
0.6 ≤|(𝑗𝜔)|≤ 1.0; 0≤𝜔≤0.35𝜋
|(𝑗𝜔)| ≤ 0.1; 0.7𝜋≤𝜔≤𝜋

4.Realize the system given by difference equation;


y(n) = -0.1y(n-1) + 0.71 y(n-2) + 0.7 x(n) - 0.252 x(n-2) in Cascade form and Parallel form.
5.Compare Butterworth and Chebyshev Filters
6.Discuss in brief about the basic structures of IIR filters.
7.Obtain the direct form I , direct form-II ,cascade and parallel form realization for the system
y(n)= -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
7. Realize system with following difference equation y(n) = (3/4) y(n-1) – (1/8) y(n-2) + x(n) + (1/3)x(n-
1)
(a)direct form-I
(b)direct form-II
8.Realise the discrete system y(n) = -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
Using,
(a) Cascade forms
(b) Parallel forms
9.Explain the features of Chebyshev approximation.
10.Design a digital low pass IIR Chebysher filter for pass band cut off frequency of 1500 Hz, stop band
cut off frequency of 7500 Hz, attenuation in pass band 3dB and attenuation in stop band 15dB. Assume
suitable sampling frequency? Use Bilinear transformation.
11. Determine the order of low pass digital FIR filter using an appropriate window function for the
following specifications: Pass band cut off frequency fp = 150 Hz, stop band frequency fs = 250 Hz.
Pass band ripple Ap = 0.1dB stop band attenuation As = 40dB sampling frequency F = 100 Hz. Also
give the design procedure for the above problem.
12.Design an analog Butterworth filter that has a 2dB pass band attenuation at a frequency of 20
radians/sec and at least 10db stop band attenuation at 30 radians/sec?

UNIT-4

2 MARKS:

1.What is linear phase design of FIR filters?

2.Define Hann window function over the range 0 to N – 1.

3.What are the effects of windowing?

4.What are the differences between IIR and FIR filters?

5.Explain direct form structure for FIR systems.

10 MARKS:

1. What are the various window functions used in the design of FIR filters? Explain.
2. Draw the ideal and practical frequency response characteristics of band stop filter and obtain the
expression for the impulse response hd(n) from the frequency response of desired filter Hd(ejw).
3. Design a digital high pass filter through FIR method by considering 7 samples of impulse response
with a cutoff frequency of 0.8π rad/sample by using hamming window.
4.Explain in brief about the different window functions used in FIR filter design.

Hd (e𝑗𝜔) = −𝑗3𝜔, −𝜋/4 ≤ 𝜔 ≤ 𝜋/4


5.Design a filter with

0 ,/4 ≤ |𝜔| ≤ 𝜋
Using Hamming window with N = 7.
6. Discuss the realization of FIR filter structures.
7.Realize FIR filter with system function in cascade form
H (z) = 1 + (5/2) z-1+2z-2+2z-3
8.Explain the FIR filter design using windowing technique.
9.Compare FIR and IIR filters.

𝐻𝑑(𝑒𝑗𝜔)= 𝑒−𝑗3𝜔; − 1/4 𝜋 ≤ 𝜔 ≤ 1/4 𝜋


10.Design a filter with desired frequency response:

𝐻𝑑(𝑒𝑗𝜔) = 0; 1/4 𝜋< |𝜔| ≤ 𝜋


Using Hamming window with N = 7.
UNIT-5

2-MARKS:

1.What are quantization errors?


2.What is Interpolation?
3.What is multirate DSP?
4.What is quantization error?
5.What is Multirate signal processing?
6.What is the effect of up-sampling & down-sampling ?
7.What are the advantages of Multirate signal processing?
8.Define Decimation.
9.What are the applications of Multirate signal processing?
10 MARKS:

1.What is multirate signal processing? What are applications? Explain in detail.


2. Evaluate the sequences
(i) y1(n) = x(2n) + x(n/2). (ii) y2(n) = x(2n) - x(n/2).
Given x(n)={1, 2, 3, 4, 5, 5, 6, 7, 8, 9}.
3. What is down sampling? Explain with a suitable example.
4. Evaluate the z-domains, X(z) and Y(z). Given y(n) = x(2n) and
x(n) = {1, 1, 2, 2, 3, 3, 4, 4, 5, 6, 6, 7, 7, 8, 8, 9, 9}.
5.Explain about Round-off effects in digital filters.
6.Explain the following:
Decimation by a factor D.
(b)Interpolation by a factor I.
7.Define down sampling and up sampling with suitable example.
8.With the help of block diagram explain the sampling rate conversion by arational factor `I/D'. Obtain
necessary expressions
9.(a) What is decimation and interpolation? Explain briefly with suitable sketches.

(b) What is aliasing? What is the need for anti- aliasing filter prior to down sampling

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