Stable System:

A stable system satisfies the BIBO boundedinputforboundedoutput condition. Here,

bounded means finite in amplitude. For a stable system, output should be bounded or finite,
for finite or bounded input, at every instant of time.

Some examples of bounded inputs are functions of sine, cosine, DC, signum and unit step.

Examples

a) y(t)=x(t)+10

Here, for a definite bounded input, we can get definite bounded output i.e. if we put
x(t)=2,y(t)=12 which is bounded in nature. Therefore, the system is stable.

b) y(t)=sin[x(t)]

In the given expression, we know that sine functions have a definite boundary of values,
which lies between -1 to +1. So, whatever values we will substitute at xt, we will get the
values within our boundary. Therefore, the system is stable.

Unstable System:

Unstable systems do not satisfy the BIBO conditions. Therefore, for a bounded input, we
cannot expect a bounded output in case of unstable systems.

Examples

a) y(t)=tx(t)

Here, for a finite input, we cannot expect a finite output. For example, if we will put
x(t)=2⇒y(t)=2t. This is not a finite value because we do not know the value of t. So, it can be
ranged from anywhere. Therefore, this system is not stable. It is an unstable system.

b) y(t)=x(t)sint

The sine function has a definite range from -1 to +1; but here, it is present in the
denominator. So, in worst case scenario, if we put t = 0 and sine function becomes zero,
then the whole system will tend to infinity. Therefore, this type of system is not at all stable.
Obviously, this is an unstable system.

Bilinear Transform:
The bilinear transform is used in digital signal processing and discrete-time control theory to
transform continuous-time system representations to discrete-time and vice versa.

The bilinear transform is a special case of a conformal often used to convert a transfer
function Ha(s) of a linear, time-invariant (LTI) filter in the continuous-time domain to a
transfer function Hd(z) of a linear, shift-invariant filter in the discrete-time domain.
Discrete-time approximation:

Properties of the Bilinear Transform:

The Bilinear Transform maps an   -plane transfer function Ha(s) to a   -plane transfer

function:

We can observe the following properties of the bilinear transform:

 Analog dc  (  ) maps to digital dc (  )

 Infinite analog frequency (  ) maps to the maximum digital frequency (

 The entire jw axis in the   plane is mapped exactly once around the unit circle in

the   plane rather than summing around it infinitely many times, or aliasing as it
does in ordinary sampling.
 Stability is preserved (when   is real and positive)
 Order of the transfer function is preserved
 Choose   to map any particular finite frequency (such as a resonance frequency)
from the jwa axis in the   plane to a particular desired location on the unit
circle   in the   plane. Other frequencies are ``warped''.
Fourier transform (FT):

A Fourier transform is a mathematical technique for converting a time function into one
expressed in terms of frequency. OR A Fourier transform is a circuit analysis technique that
decomposes or separates a waveform or function into sinusoids of different frequency
which sum to the original waveform.

Properties of Fourier Transform:

Linearity: Addition of two functions corresponding to the addition of the two frequency
spectrum is called the linearity. If we multiply a function by a constant, the Fourier
transform of the resultant function is multiplied by the same constant. The Fourier
transform of sum of two or more functions is the sum of the Fourier transforms of the
functions.

Case I. If h(x) -> H(f) then ah(x) -> aH(f)

Case II. If h(x) -> H(f) and g(x) -> G(f) then h(x)+g(x) -> H(f)+G(f)

Scaling: Scaling is the method that is used to the change the range of the independent
variables or features of data. If we stretch a function by the factor in the time domain then
squeeze the Fourier transform by the same factor in the frequency domain.

If f(t) -> F(w) then f(at) -> (1/|a|)F(w/a)

Differentiation: Differentiating function with respect to time yields to the constant multiple
of the initial function. If f(t) -> F(w) then f'(t) -> jwF(w)

Convolution: It includes the multiplication of two functions. The Fourier transform of a

convolution of two functions is the point-wise product of their respective Fourier
transforms.

If f(t) -> F(w) and g(t) -> G(w) then f(t)*g(t) -> F(w)*G(w)

Frequency Shift: Frequency is shifted according to the co-ordinates. There is a duality

between the time and frequency domains and frequency shift affects the time shift.

If f(t) -> F(w) then f(t)exp[jw't] -> F(w-w')

Time Shift: The time variable shift also effects the frequency function. The time shifting
property concludes that a linear displacement in time corresponds to a linear phase factor in
the frequency domain. If f(t) -> F(w) then f(t-t') -> F(w)exp[-jwt']..
What is filter?

A filter is a “block box” that takes an input signal processes it and then returns an output
signal that in some way modifies the input. Filters are used to refine signals such as
removing noise, cutoff and allowing of signals.

For example: If the input signal is noisy then one would want a filter that removes noise but
otherwise leaves the signal unchanged.

There are two main uses of filters:

i)Signal restoration ii) Signal separation

i)Signal restoration is when two signals are overlapping, we separate signal before
transmitting the signal through a channel.

For example: we have a noisy signal then we separate or filter them from noise.

ii)In signal restoration we perform the degradation/normalization of the signal.

For example: we have a signal with amplitude [1,0,0,1] we degrade/normalize it to

[0.75,0,0,0.8,0.6]. There are two types of filters:

i. Analogue filters (cheap, fast, Huge range)

ii. Digital filters (expensive, accurate, superior performance)

Most popular filters used are:

Low pass filters: filters to block out frequencies

High pass filters: filters to block out low frequencies

Band pass filters: filters which allow only a range of frequencies to pass

Band stop filters: filters which only reject a range of frequencies

Comb filters: a filter designed to reject a specific frequency

All pass filters: filters that allow all frequencies to pass and which modify the phase of a
signal.
Linear filter:

Linear filters are those which process time varying input signals to produce output signals. It
is controlled scaling of the signal components in the frequency domain. Linear filtering hold
the following two properties: Scaling, Superposition

Scaling: The amplitude of the output is proportional to the amplitude of the input.

Superposition: When two signals are added together and fed to the filter, the filter output is
the same as if one had put each signal through filter separately and then added the outputs.

Finite Impulse Response (FIR) filters:

An FIR filter is a filter with no feedback in its equation. This can be an advantage because it
makes an FIR filter inherently stable. Another advantage of FIR filters is the fact that they
can produce linear phases.

So, if an application requires linear phases, the decision is simple, an FIR filter must be used.
The main drawback of a digital FIR filter is the time that it takes to execute.

Since the filter has no feedback, many more coefficients are needed in the system equation
to meet the same requirements that would be needed in an IIR filter. For every extra
coefficient, there is an extra multiply and extra memory requirements for the DSP.

For a demanding system, the speed and memory requirements to implement an FIR system
can make the system unfeasible.

Infinite Impulse Response (IIR) filters:

To reduce the demand on the system while maintaining requirements, an IIR filter can be
used. An IIR filter uses both inputs and past outputs in its equation, allowing it to operate
much more efficiently.

However, a drawback of having feedback is that linear phase is almost impossible to

maintain. So, if phase doesn’t matter, and the designer wants to reduce the number of taps
(coefficients) in the system, an IIR filter is the filter of choice.
Impulse Invariance:

Impulse invariance is a technique for designing discrete-time infinite-impulse-response

filters from continuous-time filters in which the impulse response of the continuous-time
system is sampled to produce the impulse response of the discrete-time system.

The continuous-time system's impulse response, h c(t) is sampled with sampling period T to
produce the discrete-time system's impulse response, h[n].

h[n]=thc(nT)

Correlation:

Correlation is a technique used to determine the similarity between two signals. Correlation
like convolution is a sum of products. When correlating two signals the higher the value of
the result the more similar the signals. A threshold value can be set to determine if the
signal is actually a replica of itself.

Auto Correlation Function

It is defined as correlation of a signal with itself. Auto correlation function is a measure of

similarity between a signal & its time delayed version. It is represented with R(τ).

Auto correlation for discrete time signal.

R11(m)=R(τ)= Σ∞−∞x(n)x∗(n-m)

Cross Correlation Function:

Cross correlation is the measure of similarity between two different signals.
Cross correlation for continuous time signal:

Cross correlation for discrete time signal:

R11(m)= Σ∞−∞x(n)y∗(n-m)
Sampling Theorem:

The sampling theorem essentially says that a signal has to be sampled at least with twice the
frequency of the original signal. Since signals and their respective speed can be easier
expressed by frequencies, most explanations of artifacts are based on their representation
in the frequency domain. The sampling frequency required by the sampling theorem is
called the Nyquist frequency.

Statement: A continuous time signal can be represented in its samples and can be recovered
back when sampling frequency fs is greater than or equal to the twice the highest frequency
component of message signal. i. e. fs≥2fm.

Convolution:

Convolution is a mathematical way of combining two signals to form a third signal. It is the
single most important technique in Digital Signal Processing.

Convolution is a mathematical operation used to express the relation between input and
output of an LTI system. It relates input, output and impulse response of an LTI system as
y(t)=x(t)∗h(t).