Department of Electrical & Electronic Engineering

Lab Report

Course Title: Laboratory on Digital Signal Processing

Course Code: EEE 712
Experiment No: 04
Experiment Name: Write a MATLAB program for: (a) N point DFT of sequence (b) N - point
IDFT of sequence and (c) Linear convolution by DFT.

Course conducted by:

Abu Zahed Chowdhury

Associate Professor Department of EEE, CU

Submitted by:

Name: Ananna Dev Aishi

ID 20702074
B.Sc. (EEE) 7th Semester
Session: 2019-2020
Date of Submission: 31 October ,2024

Experiment Name: Write a MATLAB program to obtain (a) N-point DFT of sequence, (b) N-
point IDFT of sequence and (c) Linear convolution by DFT.
  To obtain N-point DFT of sequence
 To obtain N-point DFT of sequence
 To obtain Linear convolution by DFT

 Hardware: Personal computer.

 Software: Matlab 2021a
Theory: The N-point Discrete Fourier Transform (DFT) is a mathematical operation
that allows us to convert an N-point time-domain sequence x(n) into an N-point
frequency domain sequence X(k). The primary objective of employing the DFT is to
transform a discrete-time signal into a frequency domain representation, which then
enables various signal processing tasks, including filtering and spectral analysis.
In essence, the DFT provides a bridge between the time and frequency domains,
allowing us to examine and manipulate signals in both domains. This transformation is a
fundamental step in digital signal processing, as it facilitates tasks such as isolating
specific frequency components, filtering out unwanted noise or interference, and gaining
insights into the spectral characteristics of a signal. The N-point DFT serves as a critical
tool for analyzing and processing digital signals in various applications, making it a
cornerstone in signal processing and analysis.

Where:
• N is the length of the DFT.
• x[n] is the input sequence in the time domain.
• X[k] is the DFT coefficient at frequency index k.

1. Start the MATLAB program.

 2. Open new M-file
3. Type the program.
4. Save in current directory.
5. Compile and Run the program.
6. If any error occurs in the program correct the error and run it again.
7. For the output see command window/ Figure window.
8. Stop the program.

FlowChart:

N – Point DFT of a Sequence: Program Code:

clc; clear all;

close all; N = input('Enter the length of sequence: ');
 x=
input('
Enter
the % Initialize DFT coefficients as zeros
X = zeros(1, N);

% Compute the N-point DFT using direct computation for k

= 0:N-1 for n =
0:N-1
X(k+1) = X(k+1) + x(n+1) * exp(-1j * 2 * pi * k * n / N);
end end

% Display the DFT coefficients disp('N-point

DFT
coefficients:'); disp(X);

% Frequency axis for

plotting f = (0:N-1);

% Plot the magnitude frequency

subplot(2, 1, 1); stem(f, abs(X));
title('Magnitude Frequency of
DFT'); xlabel('Frequency
(cycles/sample)');
ylabel('Magnitude');

% Plot the phase frequency

subplot(2, 1, 2); stem(f,
angle(X));
title('Phase Frequency of DFT');
xlabel('Frequency (cycles/sample)');
ylabel('Phase
(radians)');

% Adjust plot layout

sgtitle('Magnitude and Phase Frequency of DFT')
sequence values (in square brackets): ');
Input:

Output:

% Plot the original and

reconstructed sequences subplot(2,
N – Point IDFT of a Sequence Program Code: 1, 1); stem(0:N-1, real(X));
title('Real Part of DFT
Coefficients');
xlabel('Frequency Index'); ylabel('Real
clc; Part');
close
all;
clear subplot(2, 1, 2); stem(0:N-1, imag(X));
all;
title('Imaginary Part of DFT
Coefficients'); xlabel('Frequency Index');
N = input('Enter the length of sequence:
ylabel('Imaginary Part'); sgtitle('DFT
'); X = input('Enter the DFT coefficients
(in square brackets): '); Coefficients');

% Initialize reconstructed sequence

as zeros x_reconstructed = zeros(1, figure; subplot(2, 1, 1); stem(0:N-
N); 1, real(x_reconstructed));
title('Real Part of Reconstructed
% Compute the N-point IDFT using Sequence'); xlabel('Sample
direct computation for n = 0:N-1 for k = Index'); ylabel('Amplitude');
0:N-1 x_reconstructed(n+1) =
x_reconstructed(n+1) + X(k+1) * exp(1j
subplot(2, 1, 2); stem(0:N-1,
*
imag(x_reconstructed));
2 * pi * k * n / N);
title('Imaginary Part of
end
Reconstructed Sequence');
x_reconstructed(n+1) =
xlabel('Sample Index');
x_reconstructed(n+1) / N;
ylabel('Amplitude');
end

sgtitle('Reconstructed Sequence');
% Display the reconstructed sequence
disp('Reconstructed sequence using
N- point IDFT:');
disp(x_reconstructed);
 Input:

Output:

Linear Convolution by
DFT: Program:

x = input('Enter the first sequence (in square brackets): '); h

= input('Enter the second sequence (in square brackets): ');

% Determine the lengths of sequences

M = length(x);
N = length(h);

% Calculate the length of the linear convolution result

L = M + N - 1;

% Initialize the linear convolution result

y = zeros(1, L);
% Perform linear convolution using direct
computation for n = 1:L for k = 1:M if n - k + 1 > 0
&& n - k + 1 <= N y(n) = y(n) + x(k) * h(n - k + 1);
end
end
end

% Display the linear convolution result

disp('Linear convolution result using
DFT:'); disp(y);

Linear Convolution by
DFT: Program Code:

x = input('Enter the first sequence (in square brackets): '); h

= input('Enter the second sequence (in square brackets): ');

% Determine the lengths of sequences

M = length(x);
N = length(h);

% Calculate the length of the linear convolution result

L = M + N - 1;

% Initialize the linear convolution

result y = zeros(1, L);
% Perform linear convolution using direct
computation for n = 1:L
for k = 1:M
if n - k + 1 > 0 && n - k + 1 <= N y(n)
= y(n) + x(k) * h(n - k + 1);
en
d
end
end

Input
:
Output
:

Result:
Thus, the MATLAB programs for N-point DFT , N-point IDFT and linear convolution
have written.

Discussion : The implementation of the N-point Discrete Fourier Transform (DFT) in

MATLAB can be considered a success, and it opens up various possibilities for signal
analysis and processing.
By computing the N-point DFT of a given sequence, MATLAB efficiently transforms a
time- domain signal into its corresponding frequency domain representation. This not
only enables us to view the signal's spectral components but also allows for various
spectral analysis and manipulation techniques to be applied.
The successful implementation of the N-point DFT in MATLAB signifies that we now
have the capability to perform tasks such as filtering to extract specific frequency
components, identifying the dominant frequencies in a signal, and examining the power
distribution across various frequency bands. It's a crucial step in digital signal processing
and significantly enhances our ability to analyze and manipulate digital signals
effectively.
In summary, the successful implementation of the N-point DFT in MATLAB empowers
us to perform a wide range of spectral analyses and processing operations, making it a
valuable tool in various fields, including communication, audio processing, and image
analysis.

Question & Answer :

Q15. How can DFT be applied to non-periodic signals, and what are the considerations?
Answer: The DFT assumes that the input signal is periodic, meaning it repeats indefinitely.
When applied to non-periodic signals, the DFT treats the signal as if it were periodic, which
can introduce discontinuities at the boundaries and lead to spectral leakage. To handle non-
periodic signals, windowing and zero-padding are commonly applied: Windowing reduces
edge discontinuities by tapering the signal. Zero-padding improves frequency resolution
without altering the original signal content. These techniques help achieve a more accurate
frequency representation of non-periodic signals.

Q2. How is linear convolution of two sequences performed using the Discrete
Fourier Transform (DFT)?
Answer: Linear convolution of two sequences can be performed using the Discrete Fourier
Transform (DFT) through the following steps:
1. Compute the DFT of both input sequences
2. Multiply the DFTs of the two sequences element-wise to obtain
3. Compute the Inverse DFT (IDFT) of Y[k] to obtain the linear convolution result
y[n].
This process is known as the "Convolution Theorem" and allows for faster convolution by
exploiting the efficient algorithms for DFT and IDFT calculations (such as the Fast
Fourier Transform, FFT). It is especially useful when dealing with long sequences, where
direct convolution can be computationally expensive.

Q12. What is the difference between the DFT and the Continuous Fourier Transform
(CFT)?
Answer: The Discrete Fourier Transform (DFT) is applied to discrete sequences (i.e.,
sampled data) and yields a discrete set of frequency components. It is primarily used for
digital signal processing. The Continuous Fourier Transform (CFT), on the other hand, is
applied to continuous signals and yields a continuous frequency spectrum. The DFT can be
thought of as a sampled version of the CFT, tailored for finite and discrete data.

Q4. What is the relationship between the DFT and the IDFT, and why are they
important in signal processing?
Answer: The DFT and IDFT are closely related mathematical operations. The DFT
transforms a signal from the time domain to the frequency domain, while the IDFT
transforms it back from the frequency domain to the time domain. They are essential
in signal processing for the following reasons:
• Analysis and Synthesis: DFT is used to analyze a signal's frequency
content, while IDFT is used to synthesize the original signal from its
frequency components.
• Filtering: In applications like audio and image processing, you can filter a
signal in the frequency domain (using DFT) and then transform it back to
the time domain (using IDFT).
• Convolution: The convolution theorem relates convolution in the time
domain to multiplication in the frequency domain, making DFT and IDFT
useful for efficient convolution calculations.
Q5. What is the advantage of using the Discrete Fourier Transform (DFT) for
linear convolution?
Answer: The advantage of using the DFT for linear convolution lies in efficiency,
especially when dealing with long sequences:
• Faster Computation: Convolution in the time domain has a time
complexity of O(N^2) for two sequences of length N. DFT-based
convolution, on the other hand, can be calculated in O(N log N) time using
algorithms like the Fast Fourier Transform (FFT).
• Reusability: Once you've computed the DFT of sequences, you can reuse
them for multiple convolutions, saving computational effort.
 • Periodic Convolution: The circular convolution is efficiently implemented
using DFT, which is especially useful in applications involving periodic
signals.
• Overlapping Signals: DFT-based convolution is beneficial when dealing
with signals that overlap or have a large number of zeros. It avoids
unnecessary multiplications and additions of zero values.
• Noise Reduction: DFT-based convolution can be used for filtering noisy
signals more efficiently than direct convolution, making it advantageous in
applications like audio and image demising.

Q7. What is circular convolution, and how does it differ from linear
convolution? Answer: Circular convolution is a type of convolution where the
input sequences are treated as periodic and the result wraps around, meaning that the
output is also periodic. In contrast, linear convolution is the standard convolution
operation where the result is not periodic, and the output sequence has a length
equal to the sum of the lengths of the two input sequences minus one.

Department of Electrical & Electronic Engineering

Lab Report

Course Title: Laboratory on Digital Signal Processing

Course Code: EEE 712
Experiment No: 05
Experiment Name: Write a MATLAB program to generate Spectrum Analysis of a sequence
using DFT.

Course conducted by:

Abu Zahed Chowdhury

Associate Professor Department of EEE, CU

Submitted by:

Name: Ananna Dev Aishi

ID 20702074
B.Sc. (EEE) 7th Semester
Session: 2019-2020
Date of Submission: 31 October, 2024

Experiment Name: Write a MATLAB program to generate Spectrum Analysis of a sequence using
DFT.

Objective:

➢ To write MATLAB program for spectrum analyzing signal using DFT.

Apparatus required:

➢ Hardware: Personal computer.

➢ Software MATLAB 2014a.

Theory: In time domain representation of digital signals describes the signal amplitude
versus the sampling time instant or the sample number. However, in some applications, signal
frequency content is very useful than as digital signal samples. The algorithm transforming
the time domain signal samples to the frequency domain components is known as the discrete
Fourier transform, or DFT. The DFT also establishes a relationship between the time domain
representation and the frequency domain representation. Therefore, we can apply the DFT to
perform frequency analysis of a time domain sequence. In addition, the DFT is widely used in
many other areas, including spectral analysis, acoustics, imaging/ video, audio,
instrumentation, and communications systems.
Spectral analysis:
It involves the calculation of waves or oscillations in a set of sequenced data. These
data may be observed as a function of one or more independent variables such as the
three Cartesian spatial coordinates or time. The results of a spectral analysis are
typical ly presented as a spectral power plot where the ordinate is either an absolute
(amplitude squared, C2n) or relative (amplitude squared normalized by the total
variance, C2n/σ2) measure of the amount of total variance explained by a sinusoidal
waveform at the nth frequency and where the abscissa is frequency or its inverse,
period.
Procedure:
1. Start the MATLAB program.
2. Type the program.
3. Compile and Run the program.
4. For the output see command window\ Figure window.

FlowChart:
 Program Code:
clc;
close
all;
clear
all;

N = input('Enter the length of DFT: '); T = input('Enter the sampling

period: '); freq_sinusoidal = input('Enter the frequency of sinusoidal
signal (Hz): ');

% Generate the time

vector t = 0:T:(N-1)*T;

% Generate the input sinusoidal sequence

x=
sin(2*pi*freq_sinusoidal*t);
% Compute the DFT of the input
sequence X = fft(x);
% Frequency axis for
plotting f = (0:N-1) / (N*T);
% Plot the magnitude
spectrum subplot(2, 1,
1); stem(f,
abs(X));
title('Magnitude
Spectrum');
xlabel('Frequency (Hz)');
ylabel('Magnitude');
subplot(2, 1, 2); stem(f, angle(X)); title('Phase Spectrum'); xlabel('Frequency (Hz)');
ylabel('Phase (radians)'); grid on; % Adjust plot layout sgtitle('Spectrum Analysis using
DFT')
Input

Output:

Result: Thus, the Spectrum Analysis of the signal using DFT is obtained using
MATLAB. By analyzing the magnitude and phase spectra, we can gain insights into the
frequency components present in the signal.

Discussion: In this lab, we developed a MATLAB program to conduct Spectrum Analysis

using the Discrete Fourier Transform (DFT). The program generated a discrete sequence,
computed its DFT, and visualized the magnitude and phase spectra. The results showed
accurate identification of frequency components, despite noise presence. The lab highlighted
the significance of understanding a signal's frequency content for various applications such as
audio and image processing. Future improvements could involve exploring advanced spectral
analysis techniques and applying the program to a diverse set of signals for a more
comprehensive understanding.
Question and Answer:

Q1. What is the purpose of the Discrete Fourier Transform (DFT)?

Answer: The purpose of the DFT is to transform a discrete-time signal from the time
domain to the frequency domain, allowing for the analysis of the signal's frequency
content. This transformation helps to identify the individual frequency components that
make up the signal.

Q2. Why is the DFT important in digital signal processing (DSP)?

Answer: The DFT is crucial in DSP because it allows signals to be analyzed in the
frequency domain. Many operations, such as filtering, modulation, and noise reduction,
are easier and more efficient in the frequency domain. The DFT helps break down
complex signals into simpler sinusoidal components for easier manipulation.

Q3. What is the role of windowing in the DFT?

Answer: Windowing is used in the DFT to reduce the effects of spectral leakage, which
occurs when a signal is not perfectly periodic within the observation window. A
windowing function, such as Hamming or Hanning, tapers the signal at its edges to
minimize the discontinuities at the boundaries, leading to a more accurate frequency
representation.

Q4. What are the limitations of the DFT?

Answer: Some limitations of the DFT include:

 Spectral Leakage: Occurs when the signal's frequency components do not align with the
DFT's frequency bins.
 Resolution: The frequency resolution depends on the length of the input sequence; longer
sequences provide better frequency resolution.
 Periodic Assumption: The DFT assumes the signal is periodic within the observation
window, which may not be true for real-world signals.

Q5. What is the significance of the DFT in modern communication systems?

Answer: In modern communication systems, the DFT is used for tasks like modulation,
demodulation, and signal detection. It helps analyze and process signals in the frequency domain,
which is crucial for efficient data transmission, error correction, and spectral analysis in wireless
communication technologies like OFDM (Orthogonal Frequency Division Multiplexing).

Q6. What is the output of the DFT in terms of magnitude and phase?
Answer: The output of the DFT consists of complex numbers, which can be expressed in terms
of magnitude and phase. The magnitude represents the amplitude of the frequency components,
while the phase indicates the phase shift of each frequency component relative to the original
signal
 Department of Electrical & Electronic Engineering

Lab Report

Course Title: Laboratory on Digital Signal Processing

Course Code: EEE 712
Experiment No: 06

Experiment Name: Write a MATLAB program to obtain: a) Partial fraction expansion of rational Z- transform. b)
Z-tran sform from partial fraction expansion. c) Power series expansion of Z-transform. d) Stability test for Z-
transform.

Course conducted by:

Abu Zahed Chowdhury

Associate Professor Department of EEE, CU

Submitted by:

Name: Ananna Dev Aishi

ID 20702074
B.Sc. (EEE) 7th Semester
Session: 2019-2020
Date of Submission: 31 October, 2024

Experiment Name: Write a MATLAB program to obtain:

 1. Stability test for Z-transform
2. Partial fraction expansion of rational Z-transform
3. Z-transform from partial fraction expansion.
4. Power series expansion of Z-transform

Objective:

➢ To write MATLAB programs for calculating Partial fraction expansion of rational Z-

transform
➢ To write MATLAB programs to find out Z-transform from partial fraction expansion.
➢ To write MATLAB programs to obtain power series expansion of Z-transform. ➢
To write MATLAB programs for Stability test of Z-transform.

Apparatus required:

➢ Hardware:
Personal computer.
➢ Software: Matlab 2019a.

Theory:

Z-Transform:

The Z-transform is a powerful mathematical tool used to convert linear discrete-time

difference equations into algebraic equations in the z-domain. This transformation is
invaluable for analyzing and solving problems related to linear shift-invariant (LSI) systems,
as it simplifies the process by working in the z-domain. Here's a paraphrased version of your
explanation:

The Z-transform serves as a vital tool in the analysis of linear shift-invariant (LSI) systems,
which are commonly represented by difference equations in the time domain. To solve these
equations effectively, they are first converted into algebraic equations in the z-domain using
the Z-transform. In the z-domain, these equations can be manipulated more easily. The results
obtained are then transformed back into the time domain using the inverse Z- transform. The
partial fraction expansion method, a technique for simplifying algebraic expressions, can be
applied to Ztransforms when the definition is rational, meaning they are expressed as the ratio
of two polynomials.

X(Z) = N(Z) / D(Z)

= 𝑏0 + 𝑏1𝑧−1 + 𝑏2𝑧−2 + ⋯ … + 𝑏𝑀 𝑧−𝑀/𝑎0 + 𝑎1𝑧−1 + 𝑎2𝑧−2 + ⋯ … … . + 𝑎𝑁𝑧−𝑁

𝑏0, 𝑏1, 𝑏2, … … , 𝑏𝑀 = Coefficients of Numerator
𝑎0, 𝑎1, 𝑎2… … ., 𝑎𝑁 = Coefficients of Denominator

M = Degree of numerator
N = Degree of denominator
N(Z) = Numerator polynomial
D(Z) = Denominator
polynomial

For causal sequence, the z-transform G(z) can be expanded into a power series in z −1. In the
series expansion, the coefficient multiplying the term z −n is then the nth sample g[n]. For a
rational G(z), a convenient way to determine the power series is to express the numerator and
the denominator as polynomial in z −1 and then obtained the power series expansion by long
division.

The inverse of a rational z-transform can be readily calculated using MATLAB. The function
impz can be utilized for this purpose. Three versions of this function are as follows:

[h, t] = impz(num, den),

[h, t] = impz(num, den, L)

[h, t] = impz(num, den, L, FT)

where the input data consist of the vectors num and den containing the coefficients of the
numerator and the denominator polynomials of the z-transform given in a descending powers
of z the output impulse response vector h, and the time index vector t. In the first form, the
length L of h is determined automatically by the computer with t = 0:L-1, whereas in the
remaining two forms it is supplied by the user through the input data L. In the last form, the
time interval is scaled so that the sampling interval is equal to 1/FT. The default value of FT
is 1.

Another way of arriving at this result using MATLAB is by making use of the M-file function
y = filter(num, den, x)

where y is the output vector containing the coefficients of the power series representation of
G(z) in increasing power of z −1. The numerator and denominator polynomial of G(z)
expressed in ascending power of z −1 are two of the input data vectors num and den. The
length of x is the same as that of y, and its elements are all zeros except for the first one being
a 1. We present next two examples to illustrate the use of both functions. Let the z-transform
of a causal sequence h[n] be given by
Procedure:

1. Start the MATLAB program.

2. Open new M-file
3. Type the program.
4. Save in current directory.
5. Compile and Run the program.
6. If any error occurs in the program correct the error and run again.
7. For the output see command window/ Figure window.
8. Stop the program.

FlowChart:

Program Code:

(a) Partial fraction expansion of rational Z-

transform

clc;
close
all;
clear
all;
numerator_coeff = input('Enter the coefficients of the numerator polynomial (in
square brackets): '); denominator_coeff = input('Enter the coefficients of the
denominator polynomial (in square brackets): ');

% Compute the partial fraction expansion

[r, p, k] = residue(numerator_coeff, denominator_coeff);

% Display the partial fraction expansion

results disp('Partial Fraction Expansion:');

disp('Residues:');
disp(r);
disp('Poles:');
disp(p);
disp('Direct
Terms:'); disp(k);

Input:

Output:

Program Code:

(b) Z-transform from partial fraction

expansion. Clc;
close
all;
clear
all;

r = input('Enter the residues (in square

brackets): '); p = input('Enter the poles (in
square brackets): '); k = input('Enter the
constant term:

'); [num, den] = residuez(r, p, k);

% Display numerator and denominator polynomial

coefficients disp('Numerator Polynomial Coefficients:');
disp(num); disp('Denominator Polynomial Coefficients:');
disp(den);
Input:

Output:

Program Code:

(c) Power series expansion of Z-transform.

clc;
close
all;
clear
all;
N = input('Enter the length of output vector: ');
num = input('Enter the numerator coefficients:
');
den = input('Enter the denominator coefficients: '); % 1/(1-2(1/z)) x = [1
zeros(1, N-1)]; % Compute the desired number of inverse transform
coefficients y = filter(num, den, x); disp('Coefficients of the power series
expansion'); disp(y) Input:
Output:

Program Code:
(d) Stability test for Ztransform
clc;
close
all;
clear
all;

numerator_coeff = input('Enter the coefficients of the numerator polynomial (in

square brackets): '); denominator_coeff = input('Enter the coefficients of the
denominator polynomial (in square brackets): ');
% Compute the roots (poles) of the denominator
polynomial denominator_roots =
roots(denominator_coeff);

% Test stability by checking if poles are inside the

unit circle is_stable = all(abs(denominator_roots) <
1);
% Display stability test
result if is_stable
disp('The system is
stable.'); else disp('The
system is not
stable.'); end
Discussion: The Z-transform is a powerful mathematical tool used in signal processing and
control theory. It converts discrete-time difference equations into algebraic equations in the
z-domain, simplifying the analysis of linear shift-invariant (LSI) systems. Partial fraction
expansion is employed to decompose complex Ztransform expressions into simpler
fractions, facilitating system analysis. Conversely, the Z-transform from partial fraction
expansion reconstructs the Ztransform from simplified fractions. Power series expansion of
the Z-transform allows the representation of Z-transforms as series of terms, useful for
approximating and analyzing complex expressions. Furthermore, a stability test for Z-
transform is essential to ensure that a discrete-time system remains stable, indicating that
the region of convergence (ROC) includes the unit circle in the Zplane.

Question and Answer:

Q1) What are the basic properties of the Z-transform that make it useful in digital
signal processing?
Answer: The Z-transform has several properties, including linearity, time-shifting,
convolution, and differentiation in the Z-domain, which make it useful for analyzing
discrete-time systems. It allows efficient analysis of system behavior, especially for
difference equations and digital filter design.

Q2)How does the region of convergence (ROC) relate to the causality of a system in
the context of Z-transforms?

Answer: The ROC of a Z-transform determines the causality of a system. For a causal
system, the ROC is typically outside the outermost pole (including infinity), and the
system’s response is dependent only on present and past inputs. For an anti-causal system,
the ROC is inside the innermost pole, and the response depends on future inputs.

Q3)What is the inverse Z-transform, and how is it related to the time-domain

sequence?

Answer: The inverse Z-transform converts a function in the Z-domain back into its
corresponding time-domain sequence. It is done through various methods such as partial
fraction decomposition, power series expansion, or contour integration, and it represents the
original discrete-time signal.

Q4)How can poles and zeros of a Z-transform be used to analyze system behavior?

Answer: The poles of a Z-transform represent the frequencies where the system's response
becomes unbounded, while the zeros represent frequencies where the response is zero. The
arrangement of poles and zeros provides insight into system stability, frequency response,
and transient behavior.

Q5)What role does the partial fraction expansion play in computing the inverse Z-
transform?

Answer: Partial fraction expansion simplifies a complex rational Z-transform into simpler
fractions that correspond to standard time-domain sequences. By decomposing the Z-
transform, you can apply known inverse Z-transform pairs to easily convert each term back
to the time-domain.

Q6)What is the difference between the unilateral and bilateral Z-transform?

Answer: The bilateral (or two-sided) Z-transform considers both positive and negative time
indices in a sequence, while the unilateral (or one-sided) Z-transform considers only non-
negative time indices. The unilateral Z-transform is often used in practical digital signal
processing since many systems are causal and deal only with present and past values.

Q7)How can the long division method be used to find the power series expansion of a
Z-transform?

Answer: The long division method can be used to express a rational Z-transform as an
infinite power series by dividing the numerator by the denominator. This method is useful
when finding a Z-transform's corresponding time-domain sequence for non-standard or
complex expressions, allowing the identification of the first few terms of the sequence.