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Assignment-4-2024

The document outlines an assignment on Digital Signal Processing covering various topics including the design of Butterworth and FIR filters, DFT properties, FFT computation methods, and the application of Fourier and Wavelet transforms. It includes specific questions requiring calculations of transfer functions, poles, zeros, and filter responses, as well as practical implementation scenarios. Additionally, it discusses the use of MATLAB for generating Short-Time Fourier Transform (STFT) and analyzing time-frequency trade-offs.

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Sanjeev Achar
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16 views2 pages

Assignment-4-2024

The document outlines an assignment on Digital Signal Processing covering various topics including the design of Butterworth and FIR filters, DFT properties, FFT computation methods, and the application of Fourier and Wavelet transforms. It includes specific questions requiring calculations of transfer functions, poles, zeros, and filter responses, as well as practical implementation scenarios. Additionally, it discusses the use of MATLAB for generating Short-Time Fourier Transform (STFT) and analyzing time-frequency trade-offs.

Uploaded by

Sanjeev Achar
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
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Digital Signal Processing (IN-270 3:0) : Assignment-4

Q1) A Butterworth LPF of fourth order and cut-off frequency of 1KHz is to be designed.
Sampling frequency is 10 KHz. Transform this filter into discrete time domain by bilinear
transformation such that the resulting filter preserves the gain at 1 KHz. Find H(z), its poles
and zeros.

Q2) A simple averaging filter is defined as


1
𝑦[𝑛] = (𝑥[𝑛 − 1] + ⋯ + 𝑥[𝑛 − 𝑁])
𝑁
This is clearly an FIR Filter.
a) Let N=4. Determine the transfer function, its zeros and poles;
b) Determine a general form for zeros and poles for any N;
c) By comparing y[n] and y[n-1] determine a recursive implementation. Also the transfer
function, together with its zeros and poles of the recursive implementation. Looking at this
example, can we say that "any" recursive filter is IIR?

Q3) We want to design a Low Pass FIR Filter with the following characteristics:
a) Passband 10kHz,
b) Stopband 11kHz, with attenuation of 50dB,
c) Sampling frequency 44kHz
Determine the causal impulse response h[n], and an expression for the phase within the
passband. Plot the magnitude response. The LP-FIR should be designed using Blackman and
Kaiser window.

Q4) The first five values of the 9-point DFT of a real-valued sequence x[n] are given by
{4, 2 − 3𝑖, 3 + 2𝑖, −4 + 6𝑖, 8 − 7𝑖}
Without computing IDFT and then DFT, only using the DFT properties determine the DFT of
each of the following sequence:
a) x1[n] = x[((n+2))9],
b) x2[n] = 2x[((2-n))9],
c) x3[n] = x[n]x[((-n))9],
d) x4[n] = x2[n]
−𝑖4𝜋𝑛
e) x5[n] = x[n]𝑒 9

Q5) When a very large FFT of a very large data set is required (e.g., of size 216 or larger), it may
be computed in stages by partially decimating the time data down to several data sets of
manageable dimension, computing their FFTs, and then rebuilding the desired FFT from the
smaller ones.
In this context, suppose you want to compute a (4N)-point FFT but your FFT hardware can
only accommodate N-point FFTs. Explain how you might use this hardware to compute that
FFT. Discuss how you must partition the time data, what FFTs must be computed, how they
must be combined, and how the partial results must be shipped back and forth from
secondary storage to the FFT processor in groups of no more than N samples. What is the
total number of complex multiplications with your method? Compare this total to the cost of
performing the (4N)-point FFT in a single pass? What is your observation?

Q6) Your DSP chip can accommodate FIR filters of maximum length 129 at audio rates of 44.1
kHz. Suppose such a filter is designed by the Kaiser method.
a) What would be the minimum transition width Δf between passband and stopband that you
can demand if the stopband attenuation is to be 80 dB?
b) If the minimum transition width Δf between passband and stopband is taken to be 2 kHz,
then what would be the maximum stopband attenuation in dB that you can demand? What
would be the corresponding passband attenuation in dB of the designed filter in this case?
c) Suppose your DSP chip could handle length-129 FIR filters at four times the audio rate, that
is, 4 × 44.1 = 176.4 kHz. You wish to use such a filter as a four-times oversampling FIR
interpolator filter for a CD player. The filter is required to have passband from 0 kHz to 19.55
kHz and stopband from 24.55 kHz up to the Nyquist frequency 176.4/2 = 88.2 kHz. Using a
Kaiser design, how much stopband attenuation in dB would you have for such a filter?

Q7) How do you use Fourier and Wavelet transforms to solve the below differential equation?
𝜕2𝑢 2
𝜕2𝑢
= 𝑣
𝜕𝑡 2 𝜕2𝑡
Elaborate on the advantages/disadvantages of using these transform while solving such a
differential equation.

Q8) Generate STFT using Matlab functions? Generate filterbank outputs using the filtering
view of the STFT? Take a speech signal, and show the time-frequency trade-off of using
Fourier transform vs STFT.

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