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    Advanced DSP Final Exam

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    Ahmed Samir Mohamed
    This document contains questions for an advanced digital signal processing (DSP) final exam. It covers topics like filtering algorithms, linear time-invariant systems, biomedical signal processing, filter design techniques, finite word length effects, and adaptive noise cancellation. The questions require analyzing systems using block diagrams, determining impulse responses, applying DFT/IDFT, designing IIR and FIR filters, and comparing different noise reduction methods in MATLAB.

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    Advanced DSP Final Exam

    Uploaded by

    Ahmed Samir Mohamed
    289 views1 page
    This document contains questions for an advanced digital signal processing (DSP) final exam. It covers topics like filtering algorithms, linear time-invariant systems, biomedical signal processing, filter design techniques, finite word length effects, and adaptive noise cancellation. The questions require analyzing systems using block diagrams, determining impulse responses, applying DFT/IDFT, designing IIR and FIR filters, and comparing different noise reduction methods in MATLAB.

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    Advanced_DSP_Final_Exam

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    This document contains questions for an advanced digital signal processing (DSP) final exam. It covers topics like filtering algorithms, linear time-invariant systems, biomedical signal processing, filter design techniques, finite word length effects, and adaptive noise cancellation. The questions require analyzing systems using block diagrams, determining impulse responses, applying DFT/IDFT, designing IIR and FIR filters, and comparing different noise reduction methods in MATLAB.

    Copyright:

    Attribution Non-Commercial (BY-NC)
    Download as PDF, TXT or read online from Scribd
    Download as pdf or txt
    289 views1 page

    Advanced DSP Final Exam

    Uploaded by

    Ahmed Samir Mohamed
    This document contains questions for an advanced digital signal processing (DSP) final exam. It covers topics like filtering algorithms, linear time-invariant systems, biomedical signal processing, filter design techniques, finite word length effects, and adaptive noise cancellation. The questions require analyzing systems using block diagrams, determining impulse responses, applying DFT/IDFT, designing IIR and FIR filters, and comparing different noise reduction methods in MATLAB.

    Copyright:

    Attribution Non-Commercial (BY-NC)
    Download as PDF, TXT or read online from Scribd
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    of 1

    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).

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