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DSP Question Bank for Summer 2022

This document is a question bank for the Digital Signal Processing subject at Shivaji University, Kolhapur, for the March 2022 examination. It includes questions on various topics such as Discrete Fourier Transform, FIR and IIR filter design, DSP processors, and modulation techniques. Each chapter contains multiple questions that cover theoretical concepts, practical applications, and design methods.

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0% found this document useful (0 votes)
101 views4 pages

DSP Question Bank for Summer 2022

This document is a question bank for the Digital Signal Processing subject at Shivaji University, Kolhapur, for the March 2022 examination. It includes questions on various topics such as Discrete Fourier Transform, FIR and IIR filter design, DSP processors, and modulation techniques. Each chapter contains multiple questions that cover theoretical concepts, practical applications, and design methods.

Uploaded by

mahiyam0671
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

Shivaji University , Kolhapur

Question Bank For Mar 2022 ( Summer ) Examination

Subject Code : 81532 Subject Name : Digital Signal Processing

Question Bank
Chapter 1 Discrete Fourier Transform

1. Write any 6 properties of Discrete Fourier Transform.


2. Find the 4 point DFT of a sequence 𝑥(𝑛) = {1,1,0,0}.
3. Find circular convolution of two finite duration sequences 𝑥1 (𝑛) = {1, −1, −2,3, −1} and
𝑥2 (𝑛) = {1,2,3}
4. Find circular convolution of two sequences using concentric circle method.
𝑥1 (𝑛) = {1,2,2,1} and 𝑥2 (𝑛) = {1,2,3,1}
5. Find circular convolution of two sequences using matrix method.
𝑥1 (𝑛) = {1,2,2,1} and 𝑥2 (𝑛) = {1,2,3,1}
6. Obtain linear convolution of 𝑥1 (𝑛) = {1,2,2,1} and 𝑥2 (𝑛) = {1, −1,1, −1} using circular
convolution.
7. Find the output y(n) of a filter whose impulse response is ℎ(𝑛) = {1,1,1} and input signal
𝑥(𝑛) = {3, −1,0,1,3,2,0,1,2,1} using overlap save method.
8. Find the output y(n) of a filter whose impulse response is ℎ(𝑛) = {1,1,1} and input signal
𝑥(𝑛) = {3, −1,0,1,3,2,0,1,2,1} using overlap add method.
9. Find the output y(n) of a filter whose impulse response is ℎ(𝑛) = {1,2} and input signal
𝑥(𝑛) = {1,2, −1,2,3, −2, −3, −1,1,1,2, −1} using overlap save method.
10. Find the output y(n) of a filter whose impulse response is ℎ(𝑛) = {1,2} and input signal
𝑥(𝑛) = {1,2, −1,2,3, −2, −3, −1,1,1,2, −1} using overlap add method.
11. Find DFT of a sequence 𝑥(𝑛) = {1,2,3,4,4,3,2,1} using DIT FFT algorithm.
12. Find DFT of a sequence 𝑥(𝑛) = {1,1,1,1,1,0,0,0} using DIT FFT algorithm.
13. Find DFT of a sequence 𝑥(𝑛) = {1,2,3,4,4,3,2,1} using DIF FFT algorithm.
14. Write the steps for 8-point DIT FFT algorithm.
15. Write the steps for 8-point DIF FFT algorithm.

Chapter 2 FIR Filter design and realization

1. What are the different types of filters based on frequency response? Explain in brief.
2. What are different characteristics of FIR filter?
3. Write advantages and disadvantages of FIR filter over IIR filter.
4. Compare FIR and IIR filters.
5. What are the techniques used to design FIR filters?
6. Explain any one method to design FIR filters.
7. Explain hamming window to design FIR filter in detail.
8. Explain hanning window to design FIR filter in detail.
Shivaji University , Kolhapur
Question Bank For Mar 2022 ( Summer ) Examination

Subject Code : 81532 Subject Name : Digital Signal Processing

9. Explain Kaiser window to design FIR filter in detail.


10. Explain Rectangular window to design FIR filter in detail.
11. Determine the direct form realization of system function 𝐻 (𝑧) = 1 + 2𝑧 −1 − 3𝑧 −2 −
4𝑧 −3 + 5𝑧 −4 .
12. Obtain cascade form realization of system function 𝐻 (𝑧) = (1 + 2𝑧 −1 − 𝑧 −2 )(1 +
𝑧 −1 − 𝑧 −2 ).
5
13. Obtain cascade form realization of system function 𝐻 (𝑧) = 1 + (2) 𝑧 −1 + 2𝑧 −2 +
3𝑧 −3 ).
14. Determine the direct form realization of system function 𝐻 (𝑧) = 1 + 3𝑧 −1 + 4𝑧 −2 +
4𝑧 −3 + 3𝑧 −4 + 𝑧 −5 .
15. Determine the cascade form realization of system function 𝐻 (𝑧) = 1 + 3𝑧 −1 + 4𝑧 −2 +
4𝑧 −3 + 3𝑧 −4 + 𝑧 −5 .

Chapter 3 IIR Filter design and realization

1. Given the specifications ∝𝑝 = 1 𝑑𝐵, ∝𝑠 = 30 𝑑𝐵, Ω𝑝 = 200 rad/sec, Ω𝑠 = 600 rad/sec.


determine the order of the low pass Butterworth filter.
2. Determine the order of the low pass Butterworth filter that has a 3 dB attenuation at 500
Hz and an attenuation of 40 dB at 1000 Hz.
3. Design an analog Butterworth filter for given specifications.
0.9 ≤ |𝐻(𝑗𝑤)| ≤ 1 for 0 ≤ 𝑤 ≤ 0.2Π
|𝐻(𝑗𝑤)| ≤ 0.2 for 0.4Π ≤ 𝑤 ≤ Π
2
4. For the analog transfer function 𝐻(𝑠) = (𝑠+1)(𝑠+2)
, determine H(z) using Impulse
Invariance method. Assume T=1 sec.
10
5. For the analog transfer function 𝐻(𝑠) = , determine H(z) using Impulse
𝑠 2+7𝑠+10
Invariance method. Assume T=0.2 sec.
2
6. For the analog transfer function 𝐻(𝑠) = (𝑠+1)(𝑠+2)
, determine H(z) using Bilinear
transformation. Assume T=1 sec.
𝑠 2+4.525
7. For the analog transfer function 𝐻(𝑠) = , determine H(z) using Bilinear
𝑠 2+0.692𝑠+0.504
transformation. Assume T=1 sec.
8. Obtain Direct form – I realization for the system described by difference equation 𝑦(𝑛) =
0.5𝑦(𝑛 − 1) − 0.25𝑦(𝑛 − 2) + 𝑥(𝑛) + 0.4𝑥(𝑛 − 1)
9. Obtain Direct form – II realization for the system described by difference equation
𝑦(𝑛) = 0.5𝑦(𝑛 − 1) − 0.25𝑦(𝑛 − 2) + 𝑥 (𝑛) + 0.4𝑥(𝑛 − 1)
Shivaji University , Kolhapur
Question Bank For Mar 2022 ( Summer ) Examination

Subject Code : 81532 Subject Name : Digital Signal Processing

10. Obtain Direct form – I realization for the system described by difference equation 𝑦(𝑛) =
−0.1𝑦(𝑛 − 1) + 0.2𝑦(𝑛 − 2) + 3𝑥(𝑛) + 3.6𝑥(𝑛 − 1) + 0.6𝑥(𝑛 − 2)
11. Obtain Direct form – II realization for the system described by difference equation
𝑦(𝑛) = −0.1𝑦(𝑛 − 1) + 0.2𝑦(𝑛 − 2) + 3𝑥(𝑛) + 3.6𝑥(𝑛 − 1) + 0.6𝑥(𝑛 − 2)
12. Obtain cascade realization for the system described by difference equation 𝑦(𝑛) =
−0.1𝑦(𝑛 − 1) + 0.2𝑦(𝑛 − 2) + 3𝑥(𝑛) + 3.6𝑥(𝑛 − 1) + 0.6𝑥(𝑛 − 2)
13. Obtain parallel realization for the system described by difference equation 𝑦(𝑛) =
−0.1𝑦(𝑛 − 1) + 0.2𝑦(𝑛 − 2) + 3𝑥(𝑛) + 3.6𝑥(𝑛 − 1) + 0.6𝑥(𝑛 − 2)
14. Obtain parallel realization for the system described by difference equation 𝑦(𝑛) =
−0.1𝑦(𝑛 − 1) + 0.72𝑦(𝑛 − 2) + 0.7𝑥(𝑛) + 0.252𝑥(𝑛 − 2)
15. Obtain cascade realization for the system described by difference equation 𝑦(𝑛) =
3 1
(4) 𝑦(𝑛 − 1) − (8) 𝑦(𝑛 − 2) + 𝑥(𝑛) + (1/3)𝑥(𝑛 − 1)

Chapter 4 DSP Processors

1. Write a note on DSP processors.


2. Write a note on TMS320C62x processor.
3. What are the features of TMS320C62x processor?
4. What are different types of architectures of DSP? Explain any one in detail.
5. Explain Von Neumann architecture.
6. Explain Harvard architecture.
7. Explain VLIW architecture.
8. Write a note on: Architecture of digital signal processors
9. What are types of DSP processors? Explain in brief.
10. Write a note on : Specifications of TMS320C67x
11. What are the different parameters that influence selection of digital signal processors?
12. Write a note on : selecting digital signal processors
13. Compare Von Neumann architecture with Harvard architecture.
14. How Harvard architecture differs with Von Neumann architecture?
15. What are digital signal processors? Give specifications of any one DSP.

Chapter 5 Amplitude Modulation

1. What are the types of communication systems? Explain in brief.


2. Explain baseband communication.
3. Explain carrier communication.
4. Compare baseband communication with carrier communication.
5. Write a note on: Amplitude Modulation
Shivaji University , Kolhapur
Question Bank For Mar 2022 ( Summer ) Examination

Subject Code : 81532 Subject Name : Digital Signal Processing

6. Explain Double Side Band AM.


7. Explain Single Side Band AM.
8. Explain Vestigial Side Band AM.
9. Write a note on : super heterodyne AM receiver
10. What is Amplitude Modulation? Explain in brief.
11. Explain any one method of Amplitude Modulation.
12. Write a note on : DSB Amplitude Modulation
13. Write a note on : SSB Amplitude Modulation
14. Write a note on : VSB Amplitude Modulation
15. Explain in brief amplitude modulation.

Chapter 6 Angle Modulation

1. Write a note on: Angle Modulation


2. Write a note on: instantaneous frequency
3. Explain the concept of instantaneous frequency.
4. Write a note on: Generation of FM waves
5. Explain the generation of FM waves.
6. What are the features of angle modulation?
7. Write a note on: Demodulation of FM waves
8. How FM waves are demodulated? Explain in brief.
9. Write a note on : FM receiver
10. Write a note on : Interference in angle modulated systems
11. Explain in brief about the interferences in angle modulated systems.
12. What is the effect of nonlinear distortion and interference to angle modulated system?
13. Explain phase modulation and frequency modulation.
14. Write a note on: Bandwidth of angle-modulated waves
15. Explain FM receiver in brief.

Common questions

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Von Neumann architecture simplifies the design with a single bus for data and instructions, reducing hardware complexity but limiting instruction execution throughput due to the bottleneck. Harvard architecture uses separate buses, increasing parallel data and instruction throughput but at the expense of increased complexity and cost, suitable for high-speed DSP tasks .

The bilinear transformation is preferred over impulse invariance because it maps the entire analog frequency range to the digital range without aliasing effects, making it suitable for high-frequency applications. It also maintains the stability by transforming stable analog poles to stable digital poles .

The TMS320C62x DSP processor features include a Very Long Instruction Word (VLIW) architecture, high processing speeds, and specialized hardware for mathematical operations. These features allow fast parallel processing of arithmetic operations and efficient handling of complex DSP tasks, making it suitable for high-performance applications like real-time audio and image processing .

Instantaneous frequency in FM signals refers to the derivative of the phase of the signal with respect to time, providing a real-time measure of frequency variation. This concept allows for the precise characterization of frequency changes over time, enabling better modulation accuracy and signal analysis capabilities, crucial for applications with dynamic signal environments .

The overlap-save method involves dividing the input signal into overlapping segments, computing their DFT using Fast Fourier Transform (FFT), multiplying with the DFT of the impulse response, and finally computing the inverse DFT of the product to get the output segments. Overlap the segments to form the complete output .

The six key properties of the Discrete Fourier Transform (DFT) are Linearity, Symmetry, Periodicity, Time Shifting, Frequency Shifting, and Convolution. Linearity indicates that a linear combination of time sequences results in the same combination of their DFTs. Symmetry shows real-valued signals have symmetric DFTs. Periodicity describes that the DFT is periodic with period N. Time Shifting shifts the DFT in phase, Frequency shifting modulates the sequence, and Convolution between sequences becomes a point-wise multiplication in the frequency domain .

The Kaiser window is parameterized, allowing the designer to adjust the trade-off between main-lobe width and side-lobe level, providing flexibility to optimize filter characteristics for specific requirements. Unlike fixed-width window functions like Hamming or Hanning, the Kaiser window offers more control over the filter’s performance in both the time and frequency domains .

Angle modulation, such as Frequency Modulation (FM), offers improved noise immunity and signal fidelity compared to amplitude modulation, which makes it preferable for applications like radio broadcasting. However, angle modulation requires a larger bandwidth and more complex receiver design, which impacts system cost and complexity .

The Hamming window method for filter design involves defining the desired filter response, selecting the window function to mitigate side-lobes, defining the filter order based on frequency specifications, applying the window to the ideal impulse response, and then transforming to the Z-domain. The Hamming window minimizes the side-lobe levels and reduces amplitude ripple in the passband .

FIR (Finite Impulse Response) filters have a finite duration impulse response, ensuring stability and linear phase but requiring more coefficients. IIR (Infinite Impulse Response) filters can have an infinite impulse response, potentially requiring fewer coefficients for a similar response but may suffer stability issues. FIR filters are preferred in applications needing linear phase, while IIR filters are favored in situations requiring efficient processing .

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