SRI VENKATESWARAA COLLEGE OF TECHNOLOGY

BHB Nagar, Vadakal Village, Sriperumbudur-602 105.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

Subject:EC8553 – Discrete Time Signal Processing Year: III

UNIT I DISCRETE FOURIER TRANSFORM

PART B
1. Compute 8 point DFT of the sequence x(n) = {0, 1, 2, 3, 4, 5, 6, 7} using radix 2 DIF
algorithm.
2. i) Determine the circular convolution of the following sequence �(�) = {1,1,2,1}, ℎ(�) =
{1,2,3,4} using DFT and IDFT method? Verfy the result using Matrix method
ii) Compute the DFT of the sequence �(�) = 𝑐𝑜� �� 2 where � = 4.Plot magnitude and phase
Spectrum
3. Compute 8 point DFT of the sequence x(n) = {0.5, 0.5, 0.5, 0.5, 0, 0, 0, 0} using radix - 2 DIT
algorithm.
4. Compute the linear convolution of finite duration sequence
h(n) = {1, 2} and x(n) = {1,2,-1,2,3,-2,-3,-1,1,1,2,-1}
by overlap add and overlap save method.
5. Summarize the following properties of DFT: Periodicity, Conjugation, Circular frequency
shifting & Multiplication
PART C
6. Evaluate the 8 point for the given sequence using DIT FFT algorithm
�(�) = { 1 𝑓𝑜� − 3 ≤ � ≤ 3
0 𝑜�ℎ𝑒�𝑤𝑖�𝑒
7. Given X(k) = {28,-4+j9.656,-4+j4,-4+j1.656,-4,-4-j1.656,-4-j4,-4-j9.656} ,find x(n) using
inverse DIT FFT algorithm
UNIT II DESIGN OF IIR FILTERS

PART B
1. Apply Impulse Invariant transform to determine �(�) for Butterworth filter satisfying the
following specifications.
0.8 ≤ |�(𝑒 �w)| ≤ 1 0 ≤ � ≤ 0.2�
|�(𝑒 ��)| ≤ 0.2 0.6� ≤ � ≤ �
2. (i) Discuss the steps in the design of digital IIR filter for any one type of filter
(ii) Explain the impulse invariant method of IIR filter design. Explain the poles and zeros
mapping procedure clearly.
3. (i) Obtain the direct form I & II, cascade realization of the following system functions.
y(n) = 0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2).
(ii) Explain the bilinear transform method of IIR filter design. What is warping effect?
Explain the poles and zeros mapping procedure clearly.
4. Analyze a digital Chebyshev filter to satisfy the constraints
0.707 ≤ |�(𝑒 ��)| ≤ 1 0 ≤ � ≤ 0.2�
|�(𝑒 ��)| ≤ 0.1 0.5� ≤ � ≤ �
using Bilinear transformation and assuming � = 1�𝑒c
5. i) Explain the procedure for designing analog filters using the Chebyshev approximation.

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 ii) Convert the following analog transfer function in to digital using impulse invariant
mapping with T=1sec

PART-C
6. Design a high pass filter with a passband cutoff frequency of 1000 Hz and down 10 dB at 350
Hz and 5000 Hz sampling frequency
7. (i) The normalized transfer function of an analog filter is given by,
Ha (Sn) = 1/Sn 2+1.414Sn+1 with a cutoff frequency of 0.4 π, using IIT
(ii) Given the specification ∝�= 3𝑑�; ∝�= 16𝑑�; 𝑓� = 1𝐾��; 𝑓� = 2𝐾��. Solve for H(s) using
Chebyshev approximation.

UNIT III DESIGN OF FIR FILTERS

PART B
1. How to design a FIR band stop filter to reject frequencies in the range 1.2 to 1.8 rad/sec using
hamming window, with length � = 6
2. Design a low pass filter using rectangular window by taking 9 samples of w(n)and with a
cutoff frequency of 1.2 radians/sec.
3. Design a FIR filter having the following specifications using Hanning window

Assume N = 11
4. Explain the designing of FIR filters using windows, Fourier Series and Frequency Sampling
method.
5. A band reject filter of length 7 is required it is to have lower and upper cut off frequencies of
3kHz and 5 kHz respectively. The sampling frequency is 20 kHz. Discover the filter coefficient
using hanning window.

PART-C
6.(i) Prove that an FIR filter has linear phase if the unit sample response satisfies the condition
ℎ(�) = ℎ(� − 1 − �). Also discuss symmetric and antisymmetric case of FIR filter when N is
odd.
(ii) Realize the linear phase FIR filter with the following impulse response
ℎ(�) = �(�) + 1 2 �(� − 1) + �(� − 4) − 1 4 �(� − 2) + 1 2 �(� − 3)
7. Construct a low pass filter using frequency sampling method with the following specifications;
cut off frequency �� = π/4 and N=15 and plot the magnitude response.

UNIT-IV FINITE WORD LENGTH EFFECTS

PART B
1. Explain how signal scaling is used to prevent overflow limit cycle in the digital filter
implementation with an example.
2. Explain the limit cycle oscillations due to product round off and overflow errors.

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3. Analyze the behavior of limit cycle oscillation with respect to the system described by the
following equation y(n) = 0.87 y(n − 1) + x(n). Determine the dead band of the system when
x(n) = 0 and y(−1) = 0.61.
Assume that the product is quantized to 4 bits by rounding.
4. Draw the product quantization noise model of second order IIR system and describe about
product quantization error with an example
5. (i) For the second order IIR filter, the system function is,
H(z) = 1/ (1 − 0.5z −1 )(1 − 0.45z −1)
Examine the effect of shift in pole location with 3 bit coefficient representation in direct and
cascade form.
(ii) Explain in detail about the quantization due to ADC
PART-C
6. Determine the variance of the round of noise power at the output of cascade realization of the
filter is as described by the transfer function �(�) = �1 (�) �2(�).Where

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7. The output of an A/D converter is applied to a digital filter with the system function;
H(z) = 0.5z/(z-−0.5) . Formulate the output noise power from the digital filter when the input
signal is quantized to have 8 bits.

UNIT-V
DIGITAL SIGNAL PROCESSORS

PART B
1. Draw and describe the architecture diagram of TMS320C5X processor.
2. Explain the various types of addressing modes of digital signal processor with suitable example.
3. List and Explain the Various instructions in TMS320C5X processor
4. a)Write short Notes on MAC
b) Write short notes on pipelining.
5. Draw the block diagram of Harvard, Von-Neumann and VLIW architecture of a DSP
and explain its blocks
PART-C
6. With neat figures explain the Architecture for one type of Digital Signal Processor with specifying the
special function registers.
7. (i) Write the applications of digital signal processing.

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(ii) Compose a simple program to generate square and saw tooth wave form.