Advanced DSP Final Exam

1) Explain, using Block Diagram, How filtering the Fast FFT Algorithm.
2) Consider the LTI discrete system described by the difference equation
y n = −x n + x n − 2
a) Determine h[n], the impulse response sequence that describes the system
b) If the input system is the sample sequence :
𝑥 𝑛 = 0120
i) Determine y[n] using circular convolution
ii) Determine y[n] using DFT/IDFT
3) A Biomedical signal processing application requires removal of mains hum (50 Hz) from an Electrocardiogram (ECG)
signals that is sampled with frequency of 500 Hz. Determine the transfer function for a notch filter suitable for this
purpose. You may assume that a notch of width 20 Hz (BW) will suffice.
4)
a) Explain the theory of the window technique for designing FIR filters.
b) What is the Gibbs phenomenon, and how it is reduced using smooth windows?
5)
a) Determine the transfer function of a recursive linear phase FIR filter for N=15. The real-valued frequency response of
the filter to be designed has to satisfy the condition:

2𝜋𝑘 1 𝑘 = 0,1,2,3
𝐻𝑟 =
15 0 𝑘 = 4,5,6,7

Realize the designed filter using the frequency sampling method and compare it to the canonical realization

b) Calculate the Impulse response of the designed filter using DIF radix 2 FFT algorithm.
6)
a) Design a band stop IIR filter, with a butterwoth magnitude frequency response that meets the following
requirements:

Lower Passband 0-50 Hz

Upper Passband 450-500Hz
Stopband 200-300Hz
Passband Ripple 3dB
Stopband Attenuation 20dB
Sampling Frequncy 1KHz

b) Determine the following :

i) The order of the prototype LPF
ii) Coefficients, and hence transfer function, of the discrete filter using the BZT
iii) Can we design this filter using the impulse invariant method
iv) Discuss the effect of the finite word length on the system
7) Take Home Questions
a) Design IIR BPF with: PassBand Edge frq =200-300Hz, Stopband Edges are 50 and 450 Hz, Sampling 8KHz, Passband
ripple 3dB, stopband attenuation = 20 dB. Plot its frequency response
b) Generate a zero-mean, WGN using the randn function in MATLAB, with twice the length of the data in the file
“E1.wav”. Use the above filter this noise sequence. Use it in the next question.
c) Explain the operation of noise canceling using adaptive filtering. Discuss why it may perform better than regular
frequency-selective filtering ( FIR,IIR ).
d) For the signal given as “E1.wav”, add first half of the noise sequence to it to generate a signal “noisy.wav”
e) Compare between LMS adaptive filtering (using the second half of the noise source as a reference), with FIR and IIR
filtering for the noise reduction task.
(using MATLAB and show the resulting sound files after noise reduction in each case).