DSP

2017
Q1 (a). Define the discrete time processing of a signal.
Ans: A discrete-time signal is a sequence of values that correspond to particular instants in
time.
Discrete time processing of a signal refers to the manipulation and analysis of a signal that
is represented by a sequence of discrete values taken at specific time intervals. In this context,
the signal is typically represented as a sequence of numbers, each corresponding to the
amplitude of the signal at a specific time instant.

The discrete time processing involves applying various operations or algorithms to the signal
sequence to achieve desired outcomes. These operations may include filtering, amplification,
modulation, demodulation, sampling, quantization, compression, and many others. The
purpose of these operations is to extract or enhance specific characteristics of the signal,
remove unwanted noise, or transform the signal in some meaningful way.

Discrete time processing is commonly used in digital signal processing (DSP) applications,
where signals are digitized and processed using computers or digital hardware. It offers
several advantages over continuous time processing, including the ability to implement
complex algorithms, perform accurate mathematical operations, and facilitate storage and
transmission of signals in digital form.

Overall, discrete time processing of a signal involves manipulating and analyzing a sequence
of discrete values representing the signal at specific time intervals, enabling various
operations and transformations to be applied to the signal for different purposes.

Q2(a). Describe the Stability and causality of LTI system.

Ans: If a system has both the linearity and time invariance properties, then this system is
called linear time invariant (LTI) system.
Stability and causality are important properties of linear time-invariant (LTI) systems, which
are widely used in signal processing and control systems. Let's explore each property:
1. Stability:
Stability refers to the behaviour of an LTI system over time. An LTI system is considered
stable if, for bounded input signals, the output remains bounded or does not grow
indefinitely. Stability ensures that the system's response remains within certain bounds and
does not exhibit unbounded or oscillatory behaviour.
LTI system is stable if its impulse response is absolutely summable i.e., finite.

∑k=−∞∞ |h(k)| < ∞

There are two types of stability for LTI systems:

a. BIBO Stability (Bounded Input, Bounded Output): An LTI system is said to be BIBO
stable if, for any bounded input signal, the output remains bounded. In other words, if the
magnitude of the input signal is limited, the magnitude of the output signal will also be
limited. BIBO stability is a desirable property for LTI systems in many practical applications.

b. Internal Stability: Internal stability refers to the stability of the system when all initial
conditions are set to zero. In an internally stable LTI system, the response to an input signal
converges to zero as time approaches infinity, assuming zero initial conditions. Internal
stability is closely related to the eigenvalues of the system's transfer function, where all
eigenvalues must have negative real parts for the system to be internally stable.

2. Causality:
 Causality is a property of an LTI system that implies the output of the system depends
only on the current and past values of the input signal, not on future values. In other
words, the output at any given time depends only on the input values up to that point in
time.
 Mathematically, an LTI system is causal if the impulse response of the system is zero
for negative time instants. This means that the output of the system at any given time
depends only on the past and current values of the input signal.

 Causality is an important property for practical LTI systems since it ensures that the
system's behaviour is physically realizable and avoids any reliance on future
information, which is often not available or predictable.
 h(t)=0; for t<0
To summarize, stability refers to the boundedness of the system's response, while causality
ensures that the output of the system only depends on past and current inputs. Both properties
play crucial roles in the analysis and design of LTI systems in various fields of engineering
and signal processing.

Q4. Discuss the basic structure for FIR and IIR filters as per following –
(i) Cascade structure:
The cascade structure is a common implementation for both Finite Impulse Response (FIR)
and Infinite Impulse Response (IIR) filters. It involves connecting multiple smaller sub-filters
in series, forming a cascade of stages. Each stage typically consists of a smaller filter with its
own transfer function.
For FIR filters, the cascade structure is straightforward. Each stage represents a linear phase
FIR filter, and the overall transfer function is obtained by multiplying the transfer functions
of all the stages.
For IIR filters, the cascade structure is achieved by breaking down the overall transfer
function into a product of smaller transfer functions, each representing a stage in the cascade.
Each stage is implemented using a smaller IIR filter.

The advantages of the cascade structure include:

- Flexibility: The cascade structure allows the design of complex filters by combining
simpler sub-filters in a modular manner. Each stage can be designed independently and
optimized for specific frequency response characteristics.
- Stability: In the case of IIR filters, the cascade structure helps ensure stability by breaking
down the overall transfer function into smaller, stable sub-filters.
- Efficiency: The cascade structure can be computationally efficient since it allows for
parallel processing of different stages and reduces the number of operations required
compared to implementing the entire filter in a single stage.

(ii) Lattice Structure:

The lattice structure is a specific form of filter implementation that can be used for both FIR
and IIR filters. It provides an alternative representation of the filter's transfer function by
decomposing it into a series of lattice stages.

In the lattice structure, the signal passes through a series of reflection coefficients and lattice
ladder sections. Each section consists of a delay element and a reflection coefficient
multiplier. The delay elements store the previous signal values, while the reflection
coefficient multipliers scale and feed back the delayed signal.

For FIR filters, the lattice structure is particularly useful for linear phase designs. Each lattice
stage represents a tapped delay line with a corresponding reflection coefficient. The overall
transfer function is obtained by combining the reflection coefficients of all stages.

For IIR filters, the lattice structure represents the recursive part of the filter as well as the
feedforward part. The reflection coefficients capture the feedback paths, while the
feedforward paths represent the direct contribution of the input to the output.

The lattice structure offers several advantages:

- Numerical stability: The lattice structure is inherently stable for IIR filters since the
reflection coefficients can be constrained within a stable range.
- Efficient adaptation: The lattice structure is well-suited for adaptive filtering algorithms,
such as the least mean squares (LMS) or recursive least squares (RLS), as it simplifies the
adaptation process.
- Reduced computational complexity: The lattice structure can lead to reduced
computational complexity compared to other filter structures, especially for high-order filters.

Overall, the cascade and lattice structures provide different approaches for implementing FIR
and IIR filters, offering advantages in terms of flexibility, stability, efficiency, and numerical
precision. The choice of structure depends on the specific requirements and constraints of the
application at hand.
Q7. Discuss the following important properties of DFT
(i) Time Reversal Property:
The time reversal property of the Discrete Fourier Transform (DFT) states that if we compute
the DFT of a time-reversed sequence, the resulting DFT coefficients will be complex
conjugates of the original sequence's DFT coefficients. Mathematically, if x[n] is the input
sequence and X[k] is its DFT, then the time reversal property can be expressed as:
X[k] = DFT(x[n]) => X'[k] = DFT(x[-n]) = X*[k]
Where X'[k] represents the DFT coefficients of the time-reversed sequence x[-n], and X*[k]
denotes the complex conjugate of X[k].
This property is useful in various applications, such as analyzing symmetric signals, filtering
in the frequency domain, and detecting symmetry or anti-symmetry in signals.

(ii) Convolution of Two Sequences:

The convolution property of the DFT relates to the multiplication of the DFT coefficients of
two sequences. Specifically, if we compute the DFT of two sequences and multiply their
corresponding DFT coefficients element-wise, it is equivalent to the DFT of their
convolution. Mathematically, if x[n] and y[n] are two input sequences with their respective
DFTs X[k] and Y[k], then the convolution property can be expressed as:
DFT(x[n] * y[n]) = X[k] * Y[k]
Where * denotes the convolution operation and X[k] * Y[k] represents element-wise
multiplication of the DFT coefficients.

This property is particularly valuable in signal processing applications where convolution is

involved, such as linear filtering and system analysis. It allows us to perform convolutions
efficiently by transforming the sequences into the frequency domain, multiplying their DFT
coefficients, and then transforming them back using the inverse DFT.

(iii) Circular Correlation:

Circular correlation is a property of the DFT that allows us to compute the correlation
between two sequences using circular shifting. The circular correlation property is similar to
convolution, but it involves circular shifts rather than linear shifts. Mathematically, if x[n]
and y[n] are two input sequences with their respective DFTs X[k] and Y[k], then the circular
correlation property can be expressed as:
DFT(x[n] ⊗ y[-n]) = X[k] * Y*[k]
Where ⊗ denotes the circular correlation operation, and Y*[k] represents the complex
conjugate of Y[k].
This property is particularly useful in applications such as signal synchronization, time delay
estimation, and pattern recognition. It enables us to efficiently compute correlations by
exploiting the circular shift property of the DFT.
2016
Q1(a). With the Help of block Diagram explain the function of digital signal processing
system. Mention the advantages and disadvantages of this system.
Ans: A digital signal processing (DSP) system is designed to process and analyze digital
signals using various algorithms and techniques. Here's a block diagram illustrating the
general function of a DSP system:

Analog ADC DSP DAC Analog

signal signal

Input -----> | DSP Processor | -----> Output

The main components of a DSP system are as follows:

1. Input: The input to the DSP system is a digital signal, typically represented as a sequence
of discrete values. The signal can come from various sources such as sensors, analog-to-
digital converters, or pre-recorded data.
2. DSP Processor: The DSP processor is the heart of the system and performs the signal
processing operations. It executes algorithms and mathematical computations on the input
signal to extract information, enhance the signal, or achieve a desired objective. The DSP
processor may include features like multiplication, addition, filtering, modulation,
demodulation, and more, depending on the specific application.
3. Output: The processed signal or the desired outcome is obtained as the output of the DSP
system. It could be a modified version of the input signal, a transformed representation, or
any other relevant result based on the application requirements.

Advantages of a DSP System:

1. Flexibility: DSP systems provide great flexibility in implementing complex algorithms
and signal processing operations. The software-based nature of DSP allows for easy
modification and reconfiguration of the system to adapt to changing requirements.
2. Precision: Digital processing enables high precision and accuracy in computations. Signal
values can be represented with a high resolution, leading to improved accuracy in analysis
and manipulation.
3. Reproducibility: DSP systems offer reproducibility of results since the same input signal
will always yield the same output. This attribute is crucial for testing, validation, and
maintaining consistency in signal processing tasks.
4. Noise Immunity: Digital signals can be more immune to noise and distortions compared
to analog signals. DSP systems can employ techniques such as error correction codes,
filtering, and noise reduction algorithms to enhance the signal quality.
Disadvantages of a DSP System:
1. Conversion Limitations: Analog signals need to be converted into digital form for
processing, and the analog-to-digital conversion introduces quantization errors and
limitations due to sampling. These limitations can affect the accuracy and fidelity of the
processed signals.
2. Processing Speed: Some complex DSP algorithms can require significant computational
resources and processing time. Real-time processing of high-speed signals or real-time
applications may demand specialized hardware or optimized algorithms to meet the required
processing speed.
3. Complexity: DSP systems can be complex to design and implement, especially for
advanced signal processing tasks. Developing efficient algorithms and managing
computational resources can be challenging and require expertise in DSP theory and
implementation.
4. Cost: DSP systems can involve higher costs compared to analog signal processing
systems. The need for specialized hardware or dedicated DSP processors, as well as software
development, can contribute to increased expenses.

Despite these disadvantages, the advantages and advancements in digital signal processing
have led to its widespread adoption in various fields, including telecommunications, audio
and video processing, biomedical applications, control systems, and more.

Q1(b). (ii) define z transform and inverse z transform.

The Z-transform is a mathematical transform used in the analysis and processing of discrete-
time signals and systems. It is analogous to the Laplace transform, which is used for
continuous-time signals and systems. The Z-transform provides a powerful tool for studying
the frequency response, stability, and transfer function of discrete-time systems.
The Z-transform of a discrete-time signal x[n] is defined as the sum of the signal's samples
multiplied by a complex exponential sequence:
X(z) = Z{x[n]} = ∑[x[n] * z-n]
where z is a complex variable, and the sum is taken over all values of n.
The Z-transform can be seen as a generalization of the discrete Fourier transform (DFT) for
signals that may not be periodic. It provides a representation of the signal in the complex z-
plane, allowing us to analyze its frequency content and system properties.

The inverse Z-transform, denoted as Z-1, is the operation that transforms the Z-domain
representation of a signal or system back into the time domain. It allows us to recover the
original discrete-time signal from its Z-transform representation.
The inverse Z-transform is typically performed using techniques such as partial fraction
expansion, power series expansion, or residue calculus. The choice of method depends on the
specific Z-transform function and its complexity.
Mathematically, if X(z) is the Z-transform of a signal x[n], the inverse Z-transform is given
by:
x[n] = Z-1{X(z)}
The inverse Z-transform allows us to obtain the time-domain representation of a signal or the
impulse response of a discrete-time system from its Z-domain transfer function.

Both the Z-transform and inverse Z-transform are important tools in digital signal processing
for system analysis, filter design, signal representation, and solving difference equations.
They play a fundamental role in understanding and manipulating discrete-time signals and
systems.

Q2(a) What is Region of Convergence (ROC)? Mention important properties of ROC.

Ans:
In the context of the Z-transform, the Region of Convergence (ROC) is a set of values in the
complex plane where the Z-transform of a discrete-time signal or system converges. It
defines the range of values for which the Z-transform is well-defined and converges to a
finite value.

The ROC is an essential concept in Z-transform analysis as it determines the stability and
causality of a discrete-time system. The properties of the ROC include:

1. Causal ROC: For a causal system, the ROC lies outside a circle in the complex plane. It
may extend to infinity, excluding a finite radius. The causality of the system ensures that its
impulse response is zero for negative time indices.

2. Non-causal ROC: In the case of a non-causal system, the ROC lies inside a circle in the
complex plane. It may extend to the origin, excluding a finite radius. The non-causality
implies that the system's impulse response has non-zero values for negative time indices.

3. Right-sided ROC: The right-sided ROC extends from the outermost pole of the Z-
transform to infinity in the complex plane. It includes all values to the right of the outermost
pole and signifies a right-sided or causal sequence or system.

4. Left-sided ROC: The left-sided ROC extends from the origin to the innermost pole of the
Z-transform in the complex plane. It includes all values inside the innermost pole and
signifies a left-sided or anti-causal sequence or system.
5. Two-sided ROC: A two-sided ROC consists of two separate regions, one to the left and
one to the right of a pole. It signifies a two-sided or bilateral sequence or system that has both
causal and anti-causal components.

6. ROC and system stability: The ROC provides valuable information about the stability of
a discrete-time system. A system is stable if and only if its ROC includes the unit circle in the
complex plane.

7. ROC and system inversion: The ROC also determines the range of z-values for which the
inverse Z-transform can be obtained. The inverse Z-transform is possible within the ROC.

8. ROC and system causality: The ROC indicates whether a system is causal, non-causal, or
two-sided. It determines the time-domain behavior and the relationship between past and
future values of the sequence or system.

Understanding and analyzing the properties of the ROC is crucial for determining system
stability, causality, and the range of valid Z-transform operations. It allows us to identify the
regions where the Z-transform exists and provides insights into the behavior and
characteristics of discrete-time signals and systems.

Q3(b). Explain direct form realization for FIR and IIR Filter.
Ans: http://www.ee.cityu.edu.hk/~hcso/ee3202_9.pdf
Direct form realizations are commonly used structures for implementing finite impulse
response (FIR) and infinite impulse response (IIR) filters. Let's discuss the direct form
realization for both types of filters:

1. Direct Form Realization for FIR Filters:

FIR filters have a finite impulse response, meaning their impulse response is of finite
duration. The direct form realization for FIR filters involves implementing the filter in a
straightforward manner using a series of delay elements and coefficients.

The basic structure of a direct form FIR filter is as follows:

+--------+ +--------+ +--------+ +--------+
x[n] --->| b0 |---->| b1 |----> | b2 |----> | bN |----> y[n]
| | | | | | | |
+---------+ +--------+ +--------+ +--------+

In this structure, each delay element represents a unit delay of one sample, and the
coefficients b0, b1, ..., bN represent the filter taps or impulse response coefficients. The input
signal x[n] is multiplied by each coefficient, and the results are summed to obtain the output
signal y[n].
The direct form realization for FIR filters is straightforward to implement and has a linear
phase response. However, it can be computationally intensive for longer filter lengths, as the
number of multipliers and adders increases proportionally to the filter order.

2. Direct Form Realization for IIR Filters:

IIR filters have an infinite impulse response, meaning their impulse response extends
indefinitely. The direct form realization for IIR filters involves feedback loops, where the
output is fed back into the filter's input.

The basic structure of a direct form IIR filter is as follows:

+-------+ +------+ +------+ +-------+
x[n] --->| b0 |---------| b1 |---->| b2 |---------| bN |
| | | | | | | |
| +----+---| a1 |<----+ +----+---| aM |
| | | | | |
+-------------+ +-----+ +-------+
z-1
+--------+
---->| -a1 |
| | |
| +---------+
|
+----> y[n]

In this structure, the input signal x[n] is multiplied by the feedforward coefficients b0, b1, ...,
bN, while the output signal y[n] is obtained by summing the output of the feedforward path
and the feedback path. The feedback coefficients a1, a2, ..., aM determine the recursive
behavior of the filter.

The direct form realization for IIR filters allows for more efficient implementations compared
to FIR filters for certain filter characteristics. However, it may introduce stability challenges
due to the presence of feedback loops. Careful selection of coefficients is necessary to ensure
stability and avoid issues such as filter oscillations or divergent behavior.

It's important to note that these are basic forms of direct realizations, and various
optimizations and variations exist, such as transposed direct form structures and higher-order
structures (e.g., cascade, parallel, or lattice structures), which can provide improved
performance and computational efficiency for FIR and IIR filters based on specific
requirements.
https://uomustansiriyah.edu.iq/media/lectures/5/5_2020_03_25!08_53_05_PM.pdf
Q5(a). What do you mean by digital filter design? Explain any one approach to design
digital filter from analog signal.
Ans:
Digital filter design refers to the process of designing a digital filter, which is a system that
processes discrete-time signals to achieve desired filtering characteristics. The goal of digital
filter design is to determine the coefficients or parameters of the filter that best satisfy certain
specifications, such as the desired frequency response, passband ripple, stopband attenuation,
and other design requirements.

One approach to designing a digital filter from an analog signal is the Analog-to-Digital Filter
Transformation method. This method involves designing an analog filter first and then
converting it into a digital filter using a suitable transformation technique.

Here's a step-by-step explanation of the Analog-to-Digital Filter Transformation method:

1. Specify the analog filter specifications: Determine the desired filter characteristics, such
as the filter type (e.g., low-pass, high-pass, band-pass), cutoff frequencies, passband ripple,
stopband attenuation, and other design requirements.
2. Design the analog prototype filter: Use established analog filter design techniques, such
as Butterworth, Chebyshev, or Elliptic filter design methods, to design an analog filter that
meets the desired specifications. The resulting analog filter will typically have an impulse
response that is continuous and infinite in duration.
3. Choose a transformation technique: Select a suitable transformation technique to convert
the analog filter into a digital filter. The most common transformation techniques are the
impulse invariant method and the bilinear transform method.
4. Apply the transformation: Apply the chosen transformation technique to obtain the
digital filter specifications. The transformation maps the analog filter's continuous-time
variables, such as frequency and impulse response, into their discrete-time counterparts.
5. Modify the digital filter: Adjust the parameters of the transformed digital filter to meet
the desired digital filter specifications. This may involve manipulating the filter coefficients
or applying additional corrections to achieve the desired frequency response, passband ripple,
stopband attenuation, or other design requirements.
6. Implement the digital filter: Once the digital filter specifications are determined,
implement the digital filter in software or hardware using appropriate techniques. This
involves programming the filter coefficients and performing the necessary computations to
process the discrete-time input signals.

By employing the Analog-to-Digital Filter Transformation method, an analog filter can be

designed and transformed into a digital filter to achieve the desired filtering characteristics.
This approach leverages the well-established analog filter design techniques and provides a
systematic way to convert the analog design into its digital counterpart.
Q5(b) Design an FIR filter to approximate an ideal low pass filter with pass band gain
of unity , cut off frequency of 850 Hz and working at sampling frequency of Fs = 5000
Hz. The length of impulse response should be 5. Use a rectangular window.
Ans: To design an FIR filter to approximate an ideal low-pass filter with the given
specifications, we can follow these steps:
Step 1: Determine the filter order
The filter order is determined by the length of the impulse response. In this case, the length of
the impulse response is specified to be 5, so the filter order is 4 (N = L - 1).
Step 2: Calculate the normalized cutoff frequency
The normalized cutoff frequency, wc, is calculated by dividing the actual cutoff frequency by
the sampling frequency. In this case, the cutoff frequency is 850 Hz and the sampling
frequency is 5000 Hz.
wc = 850 / 5000 = 0.17
Step 3: Calculate the filter coefficients
To design the filter coefficients, we can use the formula for a windowed FIR filter:
h[n] = (2 * wc * sinc(2 * wc * (n - N / 2))) * w[n]
where h[n] is the filter impulse response, N is the filter order, and w[n] is the window
function. In this case, we'll use a rectangular window, which is simply a sequence of ones of
length N + 1.
Using these formulas, we can calculate the filter coefficients:
h[n] = (2 * 0.17 * sinc(2 * 0.17 * (n - 2))) * 1

For n = 0:
h[0] = (2 * 0.17 * sinc(2 * 0.17 * (0 - 2))) = 0

For n = 1:
h[1] = (2 * 0.17 * sinc(2 * 0.17 * (1 - 2))) = 0

For n = 2:
h[2] = (2 * 0.17 * sinc(2 * 0.17 * (2 - 2))) = 2 * 0.17 = 0.34

For n = 3:
h[3] = (2 * 0.17 * sinc(2 * 0.17 * (3 - 2))) = 0

For n = 4:
h[4] = (2 * 0.17 * sinc(2 * 0.17 * (4 - 2))) = 0

So, the filter coefficients are: [0, 0, 0.34, 0, 0]

These coefficients represent the impulse response of the FIR filter.

Note: The sinc function used here is defined as sinc(x) = sin(pi * x) / (pi * x).

This is one possible design for the FIR filter with a rectangular window to approximate the
desired low-pass filter.
 OR

Q6(i) Give the computational efficiency of FFT over DFT.

Ans: Computationally not found

The Fast Fourier Transform (FFT) algorithm is more computationally efficient than the
Discrete Fourier Transform (DFT) for most practical applications. The computational
efficiency of the FFT can be attributed to several factors:
1. Reduced computational complexity: The FFT algorithm has a significantly lower
computational complexity compared to the DFT. While the DFT requires O(N^2) operations
to compute N-point DFT, the FFT reduces the complexity to O(N log N) operations. This
improvement is achieved by exploiting the inherent symmetry and periodicity properties of
the DFT.

2. Utilization of Cooley-Tukey factorization: The most widely used FFT algorithm, known
as the Cooley-Tukey algorithm, employs a recursive divide-and-conquer strategy. It breaks
down the DFT computation into smaller sub-problems, which are then combined to obtain the
final result. This factorization allows for efficient computation by reducing the number of
operations required.

3. Exploitation of symmetries: The FFT takes advantage of symmetries in the input

sequence to eliminate redundant calculations. For example, for real-valued signals, the DFT
spectrum exhibits symmetry, with the negative frequency components being the complex
conjugates of the positive frequency components. The FFT algorithm can exploit this
symmetry to reduce the number of computations.

4. Use of twiddle factor tables: The FFT precomputes and stores twiddle factors, which are
complex exponential terms used in the computation of frequency components. By using these
precomputed values, the FFT avoids redundant calculations, leading to improved
computational efficiency.

Overall, the FFT algorithm offers a significant speed improvement over the DFT for most
practical applications. This makes it the preferred choice for performing spectral analysis,
filtering, and other frequency domain operations on discrete-time signals. The computational
efficiency of the FFT enables real-time and high-speed processing of large data sets, making
it a fundamental tool in various fields such as telecommunications, audio processing, image
processing, and scientific computing.

Q6(ii). Explain decimators and interpolators with suitable diagram and derivation.
Ans:
Decimators and interpolators are digital signal processing techniques used to change the
sampling rate of a discrete-time signal. Let's discuss each of them with suitable diagrams and
derivations:
1. Decimators:
Decimation is the process of reducing the sampling rate of a signal by a factor of M, where M
is an integer greater than one. The decimation process involves two main steps: low-pass
filtering and downsampling.
Here is the block diagram illustrating the decimation process:
+-----------+ +---------------------+ +-------+
| x[n] | | Low-Pass Filter | | y[n] |
-->| |----->| (LPF) |----->| |
| | | | | |
+-----------+ +---------------------+ +-------+
↑ ↓
Original signal Decimated signal
The decimation process can be mathematically described as follows:
a. Low-Pass Filtering: The original signal x[n] is first passed through a low-pass filter (LPF)
to remove high-frequency components that exceed the new desired Nyquist frequency. The
cutoff frequency of the LPF is determined based on the new sampling rate after decimation.
b. Downsampling: After low-pass filtering, the signal is downsampled by a factor of M. This
means that only every Mth sample is kept, while the rest are discarded.

The decimation process effectively reduces the sampling rate and bandwidth of the signal
while preserving the essential information within the new Nyquist frequency. The low-pass
filtering step prevents aliasing, ensuring that the signal is properly reconstructed after
downsampling.

2. Interpolators:
Interpolation is the process of increasing the sampling rate of a signal by a factor of L, where
L is an integer greater than one. The interpolation process involves two main steps:
upsampling and interpolation filtering.
Here is the block diagram illustrating the interpolation process:
+---------+ +----------------------+ +-------+
| x[n] | | Interpolation Filter | | y[n] |
-->| |----->| (LPF) |----->| |
| | | | | |
+---------+ +----------------------+ +-------+
↑ ↓
Original signal Interpolated signal

The interpolation process can be mathematically described as follows:

1. Upsampling: The original signal x[n] is first upsampled by inserting L-1 zeros between
each sample. This increases the sampling rate of the signal by a factor of L.
2. Interpolation Filtering: The upsampled signal is then passed through an interpolation
filter, typically a low-pass filter (LPF), to reconstruct the continuous-time waveform. The
LPF removes the spectral replicas created during upsampling and smooths the signal.

The interpolation process effectively increases the sampling rate while maintaining the
original signal's frequency content. The interpolation filter aids in reconstructing the
continuous-time waveform and eliminating artifacts caused by upsampling.

Derivation:
The interpolation process can be mathematically derived using the ideal interpolation
formula:
y(t) = Σ x(n) * sinc(t - n)
where y(t) is the interpolated signal, x(n) is the original signal, and sinc(t - n) is the sinc
function.
By upsampling the original signal by a factor of L and applying the ideal interpolation
formula, the interpolated signal y(n) can be obtained:
y(n) = Σ x(k) * sinc(n/L - k)
where k takes integer values.
The interpolation filter is designed to approximate the ideal sinc function, effectively
reconstructing the continuous-time

Q7 Short Notes –
(i) Stability of LTI system:
Stability is an important property of Linear Time-Invariant (LTI) systems. An LTI system is
considered stable if its output remains bounded for any bounded input. In other words, a
stable system does not exhibit unbounded or oscillatory behavior.
There are two types of stability for LTI systems:
- BIBO Stability (Bounded-Input Bounded-Output): An LTI system is BIBO stable if
every bounded input signal produces a bounded output signal.
- Internal Stability: An LTI system is internally stable if its impulse response is absolutely
summable.
For BIBO stability, a common criterion is that the system's transfer function's poles should lie
within the unit circle in the complex plane. This criterion ensures that the system's output
does not grow without bounds. In terms of the frequency response, a stable system will have
a bounded magnitude response for all frequencies.

(ii) Properties of LTI system:

LTI systems possess several key properties that make them mathematically and practically
convenient to analyze:
- Linearity: LTI systems exhibit linearity, which means they satisfy the properties of
superposition and homogeneity. Superposition states that the system's response to the sum of
multiple inputs is equal to the sum of the system's responses to individual inputs.
Homogeneity states that scaling the input signal results in a proportional scaling of the output
signal.
- Time-Invariance: LTI systems are time-invariant, meaning that their behavior does not
change with respect to a time shift in the input signal. A delayed input signal will result in a
delayed output signal, without any other changes.
- Convolution: The input-output relationship of LTI systems is described by convolution.
The output of an LTI system is obtained by convolving the input signal with the system's
impulse response.
- Frequency Response: LTI systems have a frequency response that is the Fourier transform
of their impulse response. The frequency response characterizes how the system behaves for
different frequencies.
- Stability: LTI systems can exhibit stability, which ensures that the output remains bounded
for any bounded input. Stability is crucial for predictable and reliable system behavior.

(iii) Chebyshev Filter:

Chebyshev filters are a type of analog or digital filter designed to have a specific frequency
response characteristic. They are named after Pafnuty Chebyshev, a Russian mathematician
who made significant contributions to the field of mathematical analysis.
Chebyshev filters are primarily known for their ability to achieve a steeper roll-off in the
stopband compared to other filter types, such as Butterworth filters. This comes at the cost of
having ripples in the passband, which is a trade-off to achieve a sharper cutoff. The
magnitude response of a Chebyshev filter exhibits ripples in either the passband or the
stopband, depending on the type of Chebyshev filter (Type I or Type II).
Chebyshev filters are widely used in applications where a sharp transition between the
passband and stopband is desired, such as in telecommunications, audio processing, and
instrumentation. They offer a flexible design approach that allows for adjusting the filter's
characteristics based on the design requirements, including the passband ripple and stopband
attenuation.

(iv) Kaiser Window:

The Kaiser window is a type of window function used in digital signal processing for various
applications, including filter design, spectral analysis, and signal processing algorithms. It is
named after James Kaiser, who introduced the window in 1980.
The Kaiser window is a tapered window function that is designed to achieve a trade-off
between the mainlobe width and the sidelobe level of the window's frequency response. It
offers adjustable parameters that allow the user to control the width and sidelobe
characteristics of the window, making it a versatile tool for signal processing applications.
The Kaiser window is commonly used in filter design, especially for finite impulse response
(FIR) filters. By applying the Kaiser window to the desired impulse response of the filter, the
windowed impulse response is obtained. This helps to control the filter's frequency response,
achieving desired specifications such as sharp cutoff, low sidelobes, or narrow transition
bands.
The Kaiser window is parameterized by two main factors: the beta parameter (β) and the
window length (N). The beta parameter determines the trade-off between mainlobe width and
sidelobe levels, with higher values of beta leading to narrower mainlobes but higher sidelobe
levels. The window length determines the overall width of the window.

(v) Properties of DFT: The Discrete Fourier Transform (DFT) is a fundamental tool in
digital signal processing for analyzing the frequency content of discrete-time signals. It has
several important properties, including:
- Linearity: The DFT is a linear transformation. It satisfies the properties of superposition
and homogeneity, similar to LTI systems.
- Periodicity: The DFT assumes that the input sequence is periodic, with a period equal to
the length of the sequence. This periodicity property can be useful when analyzing signals
with periodic behavior.
- Time and Frequency Reversal: The DFT exhibits time and frequency reversal properties.
Reversing the input sequence results in time reversal in the frequency domain, and vice versa.
- Circular Convolution: The DFT performs circular convolution, which is a circular shift in
the time domain equivalent to a multiplication in the frequency domain.
- Parseval's Theorem: Parseval's theorem states that the energy of a signal in the time
domain is equal to the energy of its frequency domain representation. It establishes the
conservation of energy between the time and frequency domains.
- Symmetry: The DFT exhibits symmetry properties in both time and frequency domains.
For real-valued input sequences, the DFT is conjugate symmetric, and for even-length
sequences, it is also symmetric about the midpoint.
- Fast Computation: The Fast Fourier Transform (FFT) algorithm is an efficient
implementation of the DFT, reducing the computational complexity from O(N^2) to O(N log
N). The FFT algorithm is widely used to compute the DFT efficiently.

These properties make the DFT a powerful tool for spectral analysis, filtering, and various
other signal processing applications.