DSP IMPORTANT QUESTIONS UNIT WISE

UNIT 1

[Link] the impulse response ℎ[𝑛] of the system described by the difference equation

8𝑦 [𝑛]+ 6𝑦[ 𝑛 – 1] = 𝑥[𝑛]

2. Discuss the sampling rate conversion by a factor I with the help of a neat block
diagram.

3. Define time invariant system. Show that the interpolator is a time-variant system.

4. Check the following filter for time invariant, causal and linear.

(i) 𝑦(𝑛) = (𝑛 − 1)𝑥2 (𝑛 + 1) (ii) 𝑦(𝑛) = 𝑛2 𝑥(𝑛 − 2)

[Link] the frequency domain representation of discrete time signals. c) Explain the
terms: i) Up – sampling ii) Down- sampling

6. Calculate the total response of the system described by

Y(n)-4y(n-1)-12y(n-1)=x(n) ; y(- 1) =1, y( -2) =2.

[Link] the transfer function of the system defined by

Y(n)-2y(n-1)=x(n)

[Link] with mathematical equations, how sampling rate can be decreased by a

factor of D.

9. Briefly introduce the concepts of Multirate Digital Signal Processing

10Check for the stability and Causality of the following systems.

i) y(n) = x(-n-3) ii) y(n) = nx(n)

[Link] the linear convolution for the two sequences x(n)={3,2,1,2},

h(n)={1,2,1,2}.

12 Find impulse response of the system described by the difference equation

y(n)+ y(n-1)-2y(n-2)= x(n-1)+2x(n-2).

13. Discuss about the power signal and Energy signal.

[Link] are the conditions for stability and causality of an LTI system? Explain.

15 Explain in detail the classification of discrete-time systems.

16 What is the need for multi-stage implementation of sampling rate converters?
Explain with an example

17. Check whether the following systems are Stable, Causal, Linear, Time Invariant and
Memory less i) 𝑦(𝑛) = sin 𝑥(𝑛) ii)

18

19. Find the output response of the discrete time system described by the following
difference equation 𝑦(𝑛) − 0.86𝑦(𝑛 − 1) + 0.186𝑦(𝑛 − 2) = 𝑥(𝑛) where 𝑥(𝑛) = ( 8/ 9 )n 𝑢(𝑛)
subjected to the initial conditions 𝑦(−1) = 1 and 𝑦(−2) = 2. Also find out the step
response

20. Discuss about the applications of Digital Signal Processing

21, Calculate the total response of the system described
Y(n)=3/2 y(n-1)-1/2y(n-2)+x(n)
With initial conditions y(-1)=0,y(-2)=-2 and input x(n)=(1/4)n u(n)
22. Explain the frequency domain representation of discrete time signals.

UNIT 2

1Find the linear convolution of the sequences x[n] = {1,4,0,9, -1} and h[n]= {-3, -4,0,7}.
[Link] the DFT of the sequence x(n) = sin[nπ/4], where N=8 using DIT FFT
algorithm.

3. Write five properties of DFS.

[Link] x(n) = {1,2,3,4,4,3,2,1}, find X(k) using DIF FFT algorithm.

[Link] the following properties of DFS.

i) Time shifting ii) Time reversal iii) Convolution.

[Link] the butterfly diagram for DITFFT algorithm.

7. Calculate the 8 point DFT of the sequence x n( ) {1,- 2,3,1,- 1,2} using DIF-FFT and
DIT-FFT

[Link] the output sequence y(n) of a filter whose impulse response is h(n)=[1,1,1]; and
input signal x[n]=[3,-1,0,1,3,2,0,1,3,1 ] using overlap save method and overlap add
method.

9. Explain use of FFT in linear filtering and correlation.

10. Compute the DFT of the following sequence using Radix -2, DIT-FFT algorithm

x[n] = [1, 1, 1, 1, 0, 0, 0, 0].

[Link] Quantization errors in the direct computation of DFT

12. Find 8-point DFT X(K) of the real sequence.

𝑥(𝑛) = {0.707,1,0.707,0, −0.707, −1, −0.707,0} by using DIF radix-2 FFT

13 Find the N-point DFT of 𝑥(𝑛) = 𝑏𝑛 cos 𝑎𝑛 using the linearity property.

14 State and prove any two properties of Discrete Fourier series.

15. Given x(n) = 2n and N=8, find X(k) using DIT-FFT algorithm

16. Find the 4-point DFT of the sequence:

x(n) = {1, −1, 1, −1}
Also, using time shift property, find the DFT of:
y(n) = x(n−2)

17. Find the IDFT of the sequence 𝑋(𝐾) = {1,1 + 𝑗, 2,1 − 2𝑗, 0,1 + 2𝑗, 0,1 + 𝑗}

[Link] the DFT for the sequence 𝑥(𝑛) = {0.5,0.5,0.5,0.5,0,0,0,0}. Also plot the
magnitude and phase response.

[Link] the circular convolution using DFT and IDFT of the sequence 𝑥1 (𝑛) = {4,3,1,2}
and 𝑥2 (𝑛) = {1,3,5,3}.

20. For a given discrete-time signal x[n] and its Z-transform X(z), express the relationship
between DTFT, DFS, DFT, and Z-Transform for this signal

21. Explain use of FFT in linear filtering and correlation.

22. Compute the DFT of the following sequence using Radix -2, DIT-FFT algorithm
x[n] = [1, 1, 1, 1, 1, 1, 1, 1]

UNIT 3

1Design a Chebyshev filter with a maximum passband attenuation of 2 dB; at

Ωp=20rad/sec and the stopband attenuation of 35 dB at Ωs=50 rad/sec.

[Link] the impulse response of digital filter to correspond to an analog filter with
impulse response ha(t) = 0.5 𝑒-2t and with a sampling rate of 1.0kHz using impulse
invariant method.

3. Differentiate “maximally flat magnitude response” and “equiripple magnitude

response” filters.
4. Convert the analog filter to a digital filter whose system function is
1
𝐻(𝑆) = (S+2)2 (S+1) , Use bilinear transformation.

[Link] the differences between bilinear transform and impulse invariant method.

[Link] the differences between analog and digital filters.

7. Design butterworth high pass filter for the given specifications:

8. Explain the steps in designing an analog low pass chebyshev filter.

9. Using the bilinear transform, design a high pass filter, monotonic in pass band with
cut off frequency of 1000 Hz and down 10 dB at 350 H. the sampling frequency is 5000
Hz.

[Link] an Chebyshev filter transfer function that satisfies the constraints

0.707 ≤│H(w)│ ≤ 1 0 ≤ w ≤ 0.2π

│H(w)│ ≤ 0.1 0.5π ≤ w ≤ π

11. Give the advantages and disadvantages of the digital filters

12. Design a digital low pass filter using Chebyshev filter that meets the following
specifications: Passband magnitude characteristics that is constant to within 1dB for
recurrences below ω = 0.2𝜋 and stopband attenuation of atleast 15dB for frequencies
between ω = 0.3𝜋 and 𝜋. Use bilinear transformation.

13. Derive the relation between digital and analog frequencies in bilinear
transformation.

[Link] a Butterworth analog high pass filter that will meet the following
specifications

i) Maximum pass band attenuation = 2dB

ii) Passband edge frequency = 200rad/sec

iii)Minimum stopband attenuation=20dB

iv) Stop band edge frequency = 100 rad/sec

15. For the given specification, design an analog Butterworth filter using impulse
invariance method with T = 1 sec:

0.9 ≤ |H(jΩ)| ≤ 1 for 0 ≤ Ω ≤ 0.2π

|H(jΩ)| ≤ 0.2 for 0.4π ≤ Ω ≤ π
16. Design a Butterworth filter using bilinear transformation with T = 1 sec:

0.8 ≤ |H(eʲω)| ≤ 1 for 0 ≤ ω ≤ 0.2π

|H(eʲω)| ≤ 0.2 for 0.6π ≤ ω ≤ π

17. Consider an analog filter with transfer function 𝐻(𝑠) = 1/ (𝑆+1)(𝑆2 +𝑆+1) Is this a
Butterworth or Chebyshev filter? Obtain the transfer function of an IIR digital filter using
impulse invariant transformation. Assume T = 1 Sec.

18. Design a Band pass Butterworth filter with sampling frequency F=7KHz, 𝛼𝑝 = 3𝑑𝐵 in
the passband 800Hz ≤ f ≤ 1000 Hz, 𝛼𝑠 = 40𝑑𝐵 in the stopband 4000Hz ≤ f ≤ ∞

19. Describe the design of Analog Low pass filter using Chebyshev Type-I filter with
necessary equations.

UNIT 4

[Link] is a Kaiser window? In what way is it superior to other window functions?

[Link] a rectangular window technique, design a low pass filter with pass band gain of
unity, cut-off frequency of 1000Hz and working at a sampling frequency of 5 KHz. The
length of the impulse response should be 7.

3. Given the filter specifications as

using rectangular window, calculate causal impulse response coefficients

4. Design a band pass filter which approximates the ideal filter with cutoff frequencies
at 0.2 rad/sec and 0.3 rad/sec. The filter order is M=7. Use the Hanning window function

5. Design a FIR low pass filter satisfying the following specifications αp≤0.1 dB; αs≥44.0
dB; p= 20 rad/sec; s=600 rad/sec and sf=100 rad/sec.

6. Explain different windows techniques in FIR digital filters.

7. Prove that for a linear phase FIR filter the impulse response is symmetric

8. Explain the type II frequency sampling method of designing an FIR digital filter.
[Link] a band pass filter which approximates the ideal filter with cutoff-frequencies
at 0.2rad/sec and 0.3rad/sec. The filter order is M=7. Use the Hanning window function

10. For the symmetric FIR filter with N odd, find the expression for the frequency
response H(eʲω).
11. The desired frequency response of a low-pass filter is given by:

Hₑ(eʲω) = e⁻ʲ³ω for |ω| ≤ 3π/4

Hₑ(eʲω) = 0 for 3π/4 < |ω| < π

Determine the expression h(n) of FIR filter if Hanning window is used with N = 7.

12. Compare IIR and FIR filters.

13. Explain the frequency-sampling method of FIR filter design with an example.

14. Design an ideal High Pass Filter the frequency of

Hd (e jω )= 1 for, π/4 ≤│ω│≤π

= 0 for │ω│≤π/4

Using Hanning window with N=11

15. Design an FIR low-pass filter with a cutoff frequency of 1000 Hz using the Hanning
window function. The filter should have a passband ripple of 0.1 dB and a stopband
attenuation of at least 60 dB. Use a sampling frequency of 8000 Hz.

UNIT 5

[Link] the z-transform, find the total solution to the following difference equation with
initial conditions, for discrete time 𝑛 ≥ 0.

5𝑦 [𝑛 + 2] − 3𝑦 [𝑛 + 1] + 𝑦 [n] = (0.8)n 𝑢 [𝑛], 𝑦 [0] = −1, 𝑦 [1] = 10

2. Determine direct form I and cascade realization of the following system:

3. Realize the following system equation in direct form-I and direct -form II

4. Write the differences between direct form-I and canonical form

[Link] the Direct form-I structure of the system given by

y[n] + 2 y[n-1] + y[n-2] = x[n] + 0.75 x[n-1]

6. Find the impulse response of the system described by difference equation

y(n) – 3y(n-1) – 4y(n-2) = x(n) + 2x(n-1) using z transform.

7. Realize following digital filter by using direct form – II realization.

y(n) = 0.5y(n −1) − 0.25y(n − 2) + x(n) + x(n −1)

[Link] the impulse response of LTI system described by

y(n) = x(n) + 0.9 y(n-1) - 0.81y(n-2)

9. Explain coefficient quantization of IIR filters.

[Link] is Round-off Noise in IIR Digital Filters? Discuss its effects in IIR filters.

11. Describe various Structures of IIR filters with suitable diagrams.

12. Explain the limit cycle oscillations due to product round-off and overflow errors.

13. Realize the IIR filter using Direct Form I structure:

H(z) = (1 + 0.5z⁻¹) / (1 − 0.8z⁻¹ + 0.15z⁻²)

14. A system is given by the difference equation 𝑦(𝑛) – 3/ 7 𝑦(𝑛 − 1) = 𝑥(𝑛). Determine the
solution when the input is 𝑥(𝑛) = ( 1 /7 )n 𝑢(𝑛) and the initial condition is given by 𝑦(−1) = 1
using Z-transform.

15. For the system function , draw a signal flow graph that
implements this system as a cascade form & parallel form realization.

16. Determine 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, when x(n) = +0.875 for n=0 and y(-1)=0. Assume 4 bit sign magnitude
representation.

17. Realize the system y(n) = 3/4 y(n-1) -1/8 y(n-2) + x(n) + 1/3 x(n-1), using cascade form
and parallel form.

18. Investigate the dead band effects in the implementation of an IIR digital filter with
the transfer function 𝐻(𝑧) = 1/ 1−0.8𝑧−1 . Determine the magnitude of dead band errors
introduced by quantizing the filter coefficients to 12 bits and inputs ranging from -2048
to 2047.