Showing posts with label ECE. Show all posts
Showing posts with label ECE. Show all posts
ECE
ELECTRONICS AND COMMUNICATION ENGINEERING




ALL semester Notes You can Download Here...Notes will be very simple.It Contains Important Concept , Block Diagram and Sub headings....It is very useful during exam time to study....

Semester
Syllabus
Semester – 1
HS6151 Technical English
MA6151 Mathematics
PH6151 Engineering Physics
CY6151 Engineering Chemistry
GE6151 Computer Programming
GE6152 Engineering Graphics
Semester – 2
HS6251 Technical English
MA6251 Mathematics
PH6251 Engineering Physics
CY6251 Engineering Chemistry
EC6201 Electronic Devices
EE6201 Circuit Theory
Semester – 3
MA6351 Transforms and Partial Differential Equations EE6352 Electrical Engineering and Instrumentation EC6301 Object Oriented Programming and Data Structures
EC6302 Digital Electronics
EC6304 Electronic Circuits- I
Semester – 4
MA6451 Probability and Random Processes
EC6401 Electronic Circuits II
EC6402 Communication Theory
EC6403 Electromagnetic Fields
EC6404 Linear Integrated Circuits
EC6405 Control System Engineering
Semester – 5
EC6501 Digital Communication
EC6503 Transmission Lines and Wave Guides
GE6351 Environmental Science and Engineering
EC6504 Microprocessor and Microcontroller
Semester – 6
MG6851 Principles of Management
CS6303 Computer Architecture
CS6551 Computer Networks
EC6601 VLSI Design
EC6602 Antenna and Wave propagation
Elective I
Semester – 7
EC6701 RF and Microwave Engineering
EC6702 Optical Communication and Networks
EC6703 Embedded and Real Time Systems
Elective II
Elective III
Elective IV
Semester – 8
EC6801 Wireless Communication
EC6802 Wireless Networks
Elective V
Elective VI
Elective -I
EC6001 Medical Electronics
EC6002 Advanced Digital Signal Processing
CS6401 Operating Systems
EC6003 Robotics and Automation
Elective -II
EC6004 Satellite Communication
EC6005 Electronic Testing
EC6006 Avionics
CS6012 Soft Computing
IT6005 Digital Image Processing
CS6013 Foundation Skills in Integrated Product Development
Elective -III
EC6007 Speech Processing
EC6008 Web Technology
EC6009 Advanced Computer Architecture
EC 6010 Electronics Packaging
EC6011 Electro Magnetic Interference and Compatibility
Elective -IV
EC6012 CMOS Analog IC Design
EC6013 Advanced Microprocessors and Microcontrollers
EC6014 Cognitive Radio
EC6015 Radar and Navigational Aids
EC6016 Opto Electronic Devices
Elective -V
EC6017 RF System Design
CS6003 Ad hoc and Sensors Networks
GE6082 Indian Constitution and Society
EC6018 Multimedia Compression and Communication
GE6075 Professional Ethics in Engineering
GE6083 Disaster Management
Elective -VI
EC6019 Data Converters
CS6701 Cryptography and Network Security
GE6757 Total Quality Management
MG6071 Entrepreneurship Development
MG6088 Software Project Management
GE6084 Human Rights


EC6303          SIGNALS AND SYSTEMS                                 L T P C                   3 1 0 4
OBJECTIVES:
· To understand the basic properties of signal & systems and the various methods of classification
· To learn Laplace Transform &Fourier transform and their properties
· To know Z transform & DTFT and their properties
· To characterize LTI systems in the Time domain and various Transform domains

UNIT I CLASSIFICATION OF SIGNALS AND SYSTEMS                          9

UNIT II ANALYSIS OF CONTINUOUS TIME SIGNALS                           9

UNIT III LINEAR TIME INVARIANT- CONTINUOUS TIME SYSTEMS 9
Differential Equation-Block diagram representation-impulse response, convolution integrals-Fourier and Laplace transforms in Analysis of CT systems

UNIT IV ANALYSIS OF DISCRETE TIME SIGNALS                                   9
Baseband Sampling - DTFT – Properties of DTFT - Z Transform – Properties of Z Transform

 UNIT V LINEAR TIME INVARIANT-DISCRETE TIME SYSTEMS 9
Difference Equations-Block diagram representation-Impulse response - Convolution sum- Discrete Fourier and Z Transform Analysis of Recursive & Non-Recursive systems
 TOTAL (L:45+T:15): 60 PERIODS

OUTCOMES: Upon the completion of the course, students will be able to:
·Analyze the properties of signals & systems
· Apply Laplace transform, Fourier transform, Z transform and DTFT in signal analysis
·Analyze continuous time LTI systems using Fourier and Laplace Transforms


·Analyze discrete time LTI systems using Z transform and DTFT 38 





IMPORTANT QUESTIONS
ANNA UNIVERSITY QUESTIONS
Subject Related PDF
Subject Related PPT


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EC6502 Principles of Digital Signal Processing
Question Bank
UNIT V
Part A

  1. What is Multirate digital signal processing?(May-2014,Nov-2011)
  2. Write the different applications of Multirate DSP.(Nov-2013,2014)
  3. What are the different areas in which Multirate DSP is used?
  4. Give the advantages of Multirate DSP.
  5. Define Decimation.(May-2012,Nov-2014,Nov-2010,May-2014)
  6. Define Interpolation. .(May-2012)
  7. What are the upsampling and downsampling process?(Nov-2011)
  8. What is the need for sampling rate conversion?
  9. What is the need for Multirate signal processing?
  10. Give some examples of Multirate Digital system.
  11. With an example explain the sampling process.
  12. Draw the block diagram of Multirate stage decimator and interpolator.
  13. Draw the block diagram of a general poly phase frame work for decimator and interpolator.
  14. What is the need for Multistage filter implementation?
15.  What are the drawbacks in Multistage filter implementation. Obtain the poly phase structure of the filter with the transfer function.
                 H(z) = (1-3z-1) / (1+4z-1)
  1. What are the advantages of poly phase decomposition

PART-B

  1. State the applications of  Multirate Digital signal processing.(May-2014)
  2. Explain with the block diagram, the general poly phase framework for interpolator and decimator.(Nov-2010,Nov-2014,May-2014,Nov-2013)
  3. Explain about different applications of Multirate digital signal processing.
  4. Explain about sampling rate conversion by a factor I/D.(May-2012)
  5. Explain the poly phase decomposition for
    (i) FIR filter structure
    (ii) IIR filter structure
  6. Explain in detail about Interpolation and decimation with Examples.(Apr-2011)
  7. Explain in detail about Adaptive Channel Equalization.
  8. Discuss on LMS adaptive filter.
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EC6502 Principles of Digital Signal Processing
Question Bank
UNIT IV
Part A

  1. What do you understand by a fixed point number? (Apr-2011)
  2. Express the fraction 7/8 and -7/8 in sign magnitude, 2s complement and 1’s  complement.                         
  3. What are the quantization errors due to finite word length registers in digital filters?                                     
  4. What are the different quantization methods?                               
  5. What are the different types of fixed point number representation?
  6. What do you understand by sign-magnitude representation?
  7. What do you understand by 2s complement representation?
  8. Write an account on floating point arithmetic?                                      
  9. What is meant by block floating point representation? What are its advantages?
  10. What are advantages of floating point arithmetic? (Nov-2011)
  11. Compare the fixed point and floating point arithmetic.                       
  12. What are the three quantization errors due to finite word length registers in digital filters?                                                                                       
  13. How the multiplication and addition are carried out in floating point arithmetic?
  14. Brief on co-efficient inaccuracy.
  15. What do you understand by input quantization error? (Nov-2013)
  16. What is product quantization error? (Nov-2010)
  17. What is meant by A/D conversion mode?
  18. What is the effect of quantization on pole locations?
  19.  What are the assumptions made concerning the statistical independence of  various noise sources that occur in realizing the filter?
  20. What is zero input limit cycle overflow oscillation    (Nov-2011)     
  21. What is meant by limit cycle oscillations? (Nov-2012,Apr-2011)
  22. Explain briefly the need for scaling digital filter implementation?
  23. Why rounding is preferred than truncation in realizing digital filter?
  24. Define the deadband of the filter?      (Nov-2014,May-2012)           
  25. Determine the dead band of the filter with pole at 0.5 and the number of bits used  for quantization is 4(including sign bit)
  26. Draw the quantization noise model for a first order IIR system
  27. What is meant by rounding? Draw the pdf of round off error
  28. What is meant by truncation? Draw the pdf of round off error (Nov-2010)
  29. What do you mean by quantization step size?
  30. What is scaling?(Nov-2014)
  31. State the methods used to prevent overflow.(Nov-2013)
  32. Find the quantization step size of the quantizer with 3 bits
  33. Give the expression for signal to quantization noise ratio and calculate the   improvement with an increase of 2 bits to the existing bit.
  34. Express the following binary numbers in decimal
(a) (100111.1110)2       (b) (101110.1111)2    (c) (10011.011)2




PART-B
  1. Draw the quantization noise model for a second order system and explain  and find its steady state output noise variance
  1. Consider the transfer function H(z)=H1(z)H2(z) where

       Find the output round off noise power. Assume a1=0.5 and a2=0.6 and find out the output round off noise power.                                                                                                   
  2. Find the effect of coefficient quantization on pole locations of the given second order IIR system when it is realized in direct form I and in cascade form. Assume a word length of 4-bits through truncation. 
      
  1. Explain the characteristics of Limit cycle oscillations with respect to the system described by the differential equations. y(n)=0.95y(n-1)+x(n) and determine the dead band of the filter.                               
  2. i) Describe the quantization errors that occur in rounding and truncation in 2’s complement.  
ii) Draw a sample/hold circuit and explain its operation.(Nov-2014,May-2014,Nov-2010,May-2011)
   
  1. Express the following decimal numbers in binary form
            (a) 525    (b) 152.1875 (c) 225.3275
  1. Express the decimal values 0.78125 and -0.1875 in (i) 1’s complement form (ii)sign magnitude form (iii) 2’s complement form.
  2. Express the decimal values  -6/8 and 9/8 in (i) Sign magnitude form (ii) 1’s complement form and (iii) 2’s complement form
  3. Study the limit cycle behavior of the following systems
(a)    y(n) = 0.7y(n-1) + x (n)
(b)   y(n) = 0.65y(n-2) + 0.52y (n-1) + x (n)
  1. Derive the quantization input noise power and determine the signal to noise ratio of the system.  (Nov-2014)
  2. Derive the truncation error and round off error noise power and compare both errors.
  3. Explain product quantization error and coefficient quantization error with examples (Nov-2014,April-2011,May-2014)
  4. Derive the scaling factor so that prevents the overflow limit cycle oscillations in a second order IIR system.
  5. 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
(a) 8 bits      (b) 16 bits                                                                                   
  1. Convert the following decimal numbers into binary:                     
(a) (20.675)     (b) (120.75)
  1. Find the steady state variance of the noise in the output due to  quantization of input for the first order filter y(n)=ay(n-1) + x(n)   
  2. The output of an A/D  is fed through a digital system whose system function is
            H(z)=0.6z/z-0.6. Find the output noise power of the digital system=8 bits.
  1. a. Write a short note on limit cycle oscillation.      (Nov-2010,Nov-2011)                                                                                                    b. Explain in detail about signal scaling.  ((Nov-2011,May-2012)
  2. A digital system is characterized by the difference equation
y(n)=0.95y(n-1)+x(n). Determine the dead band of the system when x(n)=0  and y(-1)=13
  1. Two first order filters are connected in cascaded  whose system functions of the
individual sections are H1(z)=1/(1-0.8z-1)  and H2(z)=1/(1-0.9z-1). Determine the
overall output noise power.           
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EC6502 Principles of Digital Signal Processing
Question Bank
Unit – III
Part – A
1.   What are the desirable and undesirable features of FIR filter?
2.   Discuss the stability of the FIR filters.         
3.   What are the main advantages of FIR over IIR?
4.   What are the design techniques of designing FIR filters?
5.   What condition on the FIR sequence h(n) are to be imposed in order that this filter can be called a Linear phase filter?
6.    State the condition for a digital filter to be a causal and stable.
7.   What is Gibbs phenomenon?   (May-2014,Apr-2011)                              
8.   What are the properties of FIR filters?(Nov-2013,Nov-2011,Apr-2011)
9.   Explain the procedure for designing FIR filters using windows.   
10.     What is the principle of designing FIR filters using windows?
11.  What are desirable characteristics of windows? (Nov-2011,Nov-2013)
12.  What is a window and why it is necessary?
13.  Difference between FIR and IIR filters.                     
14.  Give the equation specifying Hanning and Blackman windows. (Nov-2014,2010)
15.  Draw the frequency response of N point Blackman window
16.  Draw the frequency response of N point Hanning window.     
17.     What is the necessary and sufficient condition for linear phase characteristics in FIR filter.
18.     Condition for the impulse response of FIR filters to satisfy for constant phase delay and group delay and only for constant group delay?
19.     Draw the direct form realization structure for FIR system.
20.     Direct form realization structure of a linear phase FIR system for N odd and N even.
21.     What are the advantages and disadvantages of FIR filters?
22.     Write the steps involved in FIR filter design.
23.     What are the possible types of impulse response for linear phase FIR filters?
24.     Write the magnitude and phase function of FIR filter when impulse response is symmetric and N is odd.
25.     Write the magnitude and phase function of FIR filter when impulse response is anti-symmetric and N is odd.
26.     Write the magnitude and phase function of FIR filter when impulse response is symmetric and N is even.
27.     Write the magnitude and phase function of FIR filter when impulse response is anti-symmetric and N is even.
28.     Write the procedure for designing FIR filter using Windows.
29.     What are the desirable characteristics of the frequency response of Window function?
30.     Compare the features of Hamming, Hanning and Blackman windows.
31.     Draw the Linear phse FIR filter for the following system function
       H(z)=1+2z-1-3z-2-4z-3.             (Nov-2010,2014,May-2012)

Part – B
1.        Explain the need for the use of window sequences in the design of FIR filter. Describe the window sequences generally used and compare their properties. 
2.        Derive the frequency response of a linear phase FIR filter when impulse responses symmetric & order M is EVEN and ODD.
3.        Derive the frequency response of a linear phase FIR filter when impulse responses Anti-symmetric & order M is EVEN and ODD.
4.        A low pass filter has the desired response as given below
                 Hd(ω) =      e –j3ω       -π/8 ≤ ω ≤ π/2
                                    0             π/8 ≤ ω ≤ π
Determine the filter coefficients h(n) for M =7 using Hamming and Hanning window.(Nov-2013,Nov-2011)
5.        Design a LPF with 11 coefficients for the following specifications
Passband frequency edge=0.25 KHz
Sampling frequency=1 KHz
Using Hanning and Hamming window.
6.                                    A LPF is to be designed with the following desired frequency response
Hd(ω) =     e –j2ω       -π/4 ≤ ω ≤ π/4
                                 0             π/4 ≤ ω ≤ π
      
Determine the filter coefficients h(n) if the window function is defined as,             W[n] =   1         0≤n≤4
                             0         else
7.                                    Determine a HPF with  M =11 using Hamming  window
Hd(ω) =     1       π/4 ≤ ω ≤ π
                                     0       ω ≤ π/4
8.                                    Realize the FIR filters using sum of the two sub-signal polyphase decomposition
H(z) = 1 + 2z-1 + (1/2)z-2 – (1/2)z-3 – (1/2)z-4+ 2(1/2)z-5 - 3(1/2)z-6
9.                                    Realize the FIR filters in direct form
a.       H(z) = 1 + 2z-1 + (1/2)z-2– (1/2)z-3 – (1/2)z-4
b.      H(z) = 1 + 2z-1 - 4z-2+ z-3 + 3z-4
10.    Design a HPF using hamming window with a cut-off frequency of 1.2radians/sec and M =9.
11.    Design a band pass filter to pass frequencies in the range of 1 to 2 rad/sec using Hanning window with M =5.
12.    Design a band pass filter to pass frequencies in the range of 1.2 to 1.7 rad/sec using Blackman window with M =7.
13.    Obtain the Linear phase realization for the transfer function given below:
a.                     H(z) = 1 + (3/4)z-1 + (17/8)z-2+ (3/4)z-3 + z-4
b.    H(z) = (2/3) + z-1 + (4/5)z-2 + 3z-3 + (7/8)z-4
14.    Design an FIR filter to meet the following specification
Passband edge = 2 KHz
Stopband edge= 5 KHz
Stopband attenuation = 44dB
Sampling frequency = 20 KHz
15.                                If the desired response of a low-pass filter is
Hd(ω) =   e-j3ω       -3π/4 < ω < 3π/4                                                    
                0            3π/4 < | ω | < π
     Determine H(ω) for M=7 using Frequency sampling tech.(Nov-2014,May-2014,2012)
16.                                Design a High pass filter using Hanning window to meet the following specifications.
     Cut-off frequency = 250 Hz                                                                                 
     Sampling frequency = 1 KHz

     Filter length = 7                                                                                          (Nov-2010)


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