UNIT 4 SIGNAL TRANSMISSION THROUGH LINEAR

SYSTEMS
13 May 2019 13:58

• Linear system
• Impulse response
• Response of a linear system, linear time-invariant (LTI) system, linear time variant (LTV) system
• Transfer function of a LTI system.
• Filter characteristics of linear systems.
• Distortion less transmission through a system
• Signal bandwidth, system bandwidth
• Ideal LPF, HPF and BPF characteristics
• Causality and Paley-Wiener criterion for physical realization
• Relationship between bandwidth and rise time.
• Energy and Power Spectral Densities

14-15 classes including problems of system response in CTFT & DTFT

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 Linear Systems & Linear Time Invariant (LTI) Systems
27 May 2019 08:58

Non-Linear systems constitute a major percentage of real-world systems. Analysis of linear systems is very
important because most non-linear systems can be approximated by linear systems over a limited range
Ex: Ohm's law, Hooke's law

For almost all active networks, the input and output signals bear linear relationship for small signals
To directly analyse a non-linear system is complicated as
• There can be no general solution
• Each situation & boundary condition has a unique solution

Linear system can be characterized by linear algebraic equations, difference equations or differential
equations and its analysis is highly developed

Linear System:
Def: A linear system is one whose response to the sum of weighted inputs is same as the sum of weighted
responses
i.e., a system is linear if it satisfies the superposition principle
Ex: filters, communication channel etc
•

• x(t) ------> SYSTEM, f ------------> y(t)

Driving fn response fn

Classification of Linear Systems

1. Lumped & Distributed Systems

Lumped Systems:
• It is a collection of individual elements interconnected in a particular way
• Consists of Lumped elements ( R, L, C ) where the energy in the system is stored or dissipated
• It is also assumed that the disturbance initiated at any point is propagated instantaneously at every
point in the system i.e., the dimensions of the elements are very small compared to the transmitted
signal wavelengths
• Ex: V=IR
• Lumped systems are described by ordinary differential equations

Distributed Systems:
• Not possible to describe the system by lumped parameters
• Ex: Transmission lines, waveguides, antennas, semiconductor devices, etc
• It takes a finite amount of time for a disturbance at one point to propagate to the other point. We have
to deal with independent variable , time t and space variable x
• They are described by partial differential equations

2. Time invariant & time variant systems

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2. Time invariant & time variant systems

Time invariant systems are those whose parameters do not change with time. Hence, they are also called
CONSTANT PARAMETER SYSTEMS
Most systems observed belong to this category
If f(t) yields a response r(t), then the same f(t) delayed by time T yields same response r(t) but delayed by
time T
Symbolically,

Systems that do not satisfy this condition are called Time Variant Systems

Systems that satisfy both the properties of linearity and time invariance are called Linear Time Invariant (LTI)
systems
• Characterized by linear equations with constant coefficients ( algebraic, difference or differential)
• Ex: Circuits using passive elements

Systems that satisfy the property of linearity but are time variant are called Linear Time Variant (LTV)
systems
• Characterized by linear equations with time dependent coefficients

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Impulse response of LTI Systems
27 May 2019 08:58

The LTI systems exhibit an important characteristic i.e., applicability of superposition theorem to find the
response y(t) to a given excitation x(t)

Following steps may be adopted to find the response of an LTI system using superposition theorem
• Resolve the input function f(t) in terms of simpler basic functions like impulse functions for which
response can be easily evaluated
• Determine individually the response of LTI system for the simpler input impulse functions
• Using superposition theorem, find the sum of the individual responses which will become the overall
response y(t) of function f(t)

In simple terms, to find the response of a LTI system to any function we have to find the response of the LTI
system to an unit impulse i.e., UNIT IMPULSE RESPONSE of the LTI system

Unit Impulse Response:

Def: It is the response of the CT (or DT) LTI system to an unit impulse input applied at time t = 0 (or n = 0)

----------> CT LTI System ------------>

Unit Impulse input Unit Impulse Response

----------> DT LTI System ------------>

Any LTI system can be completely characterized in terms of its unit impulse response

Representation of a function f(t) as a continuous sum of impulse functions:

Consider an arbitrary CT input signal f(t).

This input excitation function can be represented in terms of ELEMENT AREAS

• nth element area can be considered as a rectangle of width and height
• Ex: the shaded area has a width of and height . The area of the shaded portion is given by
" "
Now, in the Lt , this element area approaches function of strength located at
this delata function can be represented as
Hence the function f(t) is a continuous sum of such impulse functions and may be represented as

Let h(t) be the unit impulse response of LTI system for an input function of with unit strength
the response of LTI system for an impulse function of strength located at is

Using the superposition theorem, the response y(t) of LTI system for the input f(t) is

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 Using the superposition theorem, the response y(t) of LTI system for the input f(t) is

As , (2) becomes an integral

which is the CONVOLUTION INTEGRAL for continuous time LTI systems

The output of any general input may be found by convolving the given input signal f(t) with LTI system's unit
impulse response h(t)

For Discrete Time LTI systems,

Let an arbitrary input be f(n) and h(n) be impulse response of the discrete time LTI system
Then the discrete time output signal of the system is

This is called the convolution sum and expressed as

The response y(n) of a discrete time LTI system is given by the convolution of discrete time input signal f(n) and
discrete time unit impulse response h(n)

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Properties of LTI systems
27 May 2019 08:58

LTI system is completely characterized by its impulse response and has a number of properties not exhibited by other
systems
1. Commutative Property
• The output of a CT LTI system having input f(t) and unit impulse response h(t) is identified to the output of a
CT LTI system having input h(t) and unit impulse response f(t)

f(t)→ h(t) →y(t)

h(t)→ f(t) →y(t)

• Holds good for Discrete time LTI system also

2. Distributive Property
• A parallel combination of LTI systems may be replaced by a single LTI system whose impulse response is the
sum of the individual unit impulse responses in the parallel combination

3. Associative Property
• The impulse response of the cascade of two LTI systems is the convolution of their individual impulse
responses

4. Static & Dynamic LTI Systems

• A system is known to be static if its output at any time depends only on the value of the input at the same
time
• For CT LTI system, it is static if its unit impulse response h(t)=0 for t≠0. (no o/p when there is no i/p)
• These memoryless LTI systems are characterized by the response y(t)=k*f(t) where k is constant --(1) and its
impulse response h(t)=k*δ(t) ---(2)
• If k=1, then it is IDENTITY SYSTEM
• Any system whose impulse response is not zero for t≠0 is called dynamic or system with memory

5. Invertibility of LTI Systems

• A system is known as invertible only if an inverse system exists which, when cascaded with the original
system, produces an output equal to the input at the first system

h(t) - impulse response of original system

h1(t) - impulse response of inverse system
• Cascade interconnection of original system with its inverse system is an identity system i.e., the overall

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 • Cascade interconnection of original system with its inverse system is an identity system i.e., the overall
impulse response of system cascaded with its inverse system is given by h(t)

6. Causality of LTI Systems

• A CT LTI system is called a causal system if its impulse response i.e., causality of an LTI
system is equivalent to its impulse response being a causal system
is an exaample of causal system
• A pure time shift with a unit impulse response is a causal LTI system for . Here the
time shift is DELAY
• When the time shift is an ADVANCE, , then the system is non-causal
7. Stability of LTI Systems
• A stable system is the one which produces bounded output for every bounded input
▪ If the impulse response h(t) is absolutely integrable, then the output of a CT LTI system is bounded in
magnitude and the system is BIBO stable
▪ A sufficient & necessary condition for stability of CT LTI system is

If the system is causal, then the condition reduces to

8. Unit Step response of LTI Systems

• It is the output of LTI system when the input is unit step function or unit step sequence i.e., Unit step
response of CT LTI system is found by

• Unit step response

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Transfer Function of an LTI system
28 May 2019 08:43

If f(t) is the input signal and h(t) is the unit impulse response of an LTI system, then its output y(t) is
expressed as a convolution integral

For any general input, the output of the LTI system can be found by convolving the signal with the
system's unit impulse response

The convolution integral can be applied only in time domain and involved complex manipulations. To
Ease the analysis, we use Fourier transform where simpler equations can be obtained in the
frequency domain.

As convolution of two signals in time is equivalent to multiplication of their spectra in frequency

domain, we get

where is the Fourier Transform of ,

is the Fourier Transform of ,
is the Fourier Transform of

ratio of transforms of output to input

is called the TRANSFER FUNCTION or SYSTEM FUNCTION of the LTI system.

It is also called the frequency response of the LTI system

If then we have
Here, corresponds to [ ]
Hence, when the input is unit impulse signal, is the response in the frequency domain of
the LTI system

Definition:
The TRANSFER FUNCTION or the frequency response of an LTI system is the transform of its
response when the input is an unit impulse function

• In general is complex and can be expressed as

• For a linear system, exhibits conjugate symmetry

○ amplitude response is symmetrical |
○ phase response is anti-symmetrical
• In log scale,

gain of the system nepers dB [1 neper = 8.69 dB]

phase shift introduced by the system

If is +ve, response > excitation

If is -ve, response < excitation (attenuation)
If is 0, response = excitation

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Filter characteristics of Linear systems
28 May 2019 09:15

For a given system, an input signal f(t) gives rise to a response signal y(t)
i.e., The system processes f(t) in such a way that is CHARACTERISTIC of the system
If the spectral density function of the input signal is given by then the spectral density function
of the response is which implies that the system modifies the spectral density function of
the input (or) THE SYSTEM ACTS AS A KIND OF FILTER TO VARIOUS FREQUENCY COMPONENTS

During transmission through a system,

• Some frequency components are boosted in strength, some are attenuated while some
remain unaffected
• Each frequency component suffers a different amount of phase shift

The system modifies the spectral density function according to its filter characteristics i.e., according
to its transfer function which is the response of the system to various frequency components

acts as a WEIGHTING FUNCTION to different frequencies

• Rising and falling characteristics of the response are controlled by

Example for filter characteristics of a linear system

Consider a Low Pass RC network. Let a square pulse be applied at its input

Transfer function of RC circuit

If RC =1 , then

Magnitude Plot of is

We know that F.T of rectangular pulse is a sinc pulse i.e.,

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We know that,

is a distorted replica of the input function. Here the distortion occurs due to the attenuation of
the high frequency components
• Input signal f(t) rises from 0 to 1 instantaneously at t=0 ⇒ very high frequency
As the system does not pass the HF components, the output cannot rise rapidly and hence it
rises slowly as compared to f(t)
Similarly, the rapid fall of the input signal from 1 to 0 at t=T is seen as a gradual decrease in the
output y(t)

is the spectral function of the response. Comparing | | and

we can say that the higher frequency components in the input have suffered severe attenuation

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we can say that the higher frequency components in the input have suffered severe attenuation
than the Low frequency components

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Distortion less Transmission through a system
29 May 2019 09:55

Def: The transmission of a signal through a system is said to be distortion less if the output signal is
an exact replica of the input signal
• A constant change in magnitude and a constant time delay in the output replica is not treated
as distortion
• The magnitude of the output replica may increase or decrease during transmission depending
on whether the system offers gain or attenuation

Thus, an input function f(t) is said to be transmitted without distortion if the output signal y(t) is
defined as
is a constant representing the change in magnitude
is a constant representing the constant time delay

According to the time shifting property

the frequency domain representation of becomes
But the response is related with input by its transfer function i.e.,

Comparing , we have which is the frequency domain

representation for distortion less transmission
The impulse response
Magnitude of
Phase response

From (5a) and (5b), we see that

• Magnitude should be constant for distortion less transmission
○ All frequency components present in f(t) are equally attenuated or amplified
○ System bandwidth is infinite
• is directly proportional to frequency

No system in practice has infinite bandwidth. Hence distortion less condition are never met.

Types of distortions in linear systems

• Amplitude distortion
○ This occurs when is not constant within the frequency band of interest and
different frequency components present in the input signal are transmitted with
different gain or attenuation
• Phase distortion
○ This occurs when is not linearly changing with time and different frequency
components in the input signal are subjected to different time delays

The above distortions change the shape of the output waveform as compared to the input waveform

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System bandwidth & signal bandwidth
29 May 2019 10:23

SYSTEM BANDWIDTH:
The bandwidth of a system is referred to as the interval or range of frequencies over which the
magnitude remains within times its value at midband i.e., within 3dB
• BW specifies the constancy of magnitude in a system.
•

• For distortion less transmission, a system with infinite BW is needed. But due to physical
limitations, it is impossible to construct such a system. In general, a satisfactory distortion less
transmission can be achieved by systems with finite but fairly large bandwidths
• Significance of large BWs lies in the fact that for any physical signal, the energy content
decreases with frequency. It is only necessary to construct a system which will transmit the
frequency components which contain the most of the energy of the signal
• Attenuation of extremely high frequency components would tend to introduce very little
distortion since these components carry very little signal energy

SIGNAL BANDWIDTH:

Signal bandwidth can be defined as the range of frequencies in which most signal energy or power
lies. It is the range of the frequencies spanned by significant components of a signal.

In general, twice the highest frequency component is taken as the signal BW i.e.,

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Ideal LPF, HPF and BPF characteristics
29 May 2019 12:09

Filter:
• ~ is a frequency selective network
• Permits a band of frequencies to pass through it with little attenuation (Pass band)
• Attenuation is severe for remaining frequencies (Stop band)

The transfer functions for ideal LPF, HPF and BPF are illustrated below. Shaded regions represent
passband and non-shaded represent stopband.

Ideal LPF passes all input signal components with radian frequencies below without any
distortion
is the cut-off frequency. The corresponding phase function for distortion less transmission is

The transfer function of ideal CT LPF is defined as

The impulse response of an ideal LPF is given as

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Ideal HPF transmits, without any distortion, all of the signals of frequencies above a certain
frequency radians per second and attenuates completely the signals of frequencies below
radians per second
is called the cutoff frequency. The corresponding phase function for distortion less transmission
is .
The transfer function of an ideal CT HPF is given by or

Ideal BPF transmits, without any distortion, all of the signals of frequencies within a certain
frequency band radians per second and attenuates completely the signals of frequencies
outside this band.
is the bandwidth of the BPF. The corresponding phase function for distortion less
transmission is .
The transfer function of an ideal BPF is given by

Ideal BRF filter rejects totally all of the signals of frequencies within a certain frequency band
radians per second and transmits without any distortion all signals of frequencies outside this
band.
is the rejection band. The corresponding phase function for distortion less transmission is
.
The transfer function of an ideal BPF is given by

In addition to these filters, there is one more filter called an all pass filter which transmits signals of
all frequencies without any distortion, that is, its bandwidth is ∞
The transfer function of an ideal all pass filter is specified by for all frequencies. The
corresponding phase function for distortion less transmission is .

All ideal filters are non-causal systems i.e., . Hence none of them is physically
realizable. However, we can design some filters which are close to ideal filters

A simple realizable LPF, its transfer function & its impulse response

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Impulse response is very similar to that of an ideal filter except that it is causal.

To distinguish between physically realizable and unrealizable filters we use the Paley-Wiener
criterion

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Causality & Paley-Wiener Criterion for physical realization
30 May 2019 09:21

A simple rule for physical realization of a system is that it cannot have a response before the input
(or excitation or driving) function is applied. This is called the CAUSALITY condition and may be
stated as

• CAUSALITY CONDITION: A unit impulse function of a physically realizable system must be

causal i.e.,

The causality condition is the time domain criterion for physical realizability

The PALEY-WIENER CRITERION gives the frequency domain equivalent for the causality criterion and
states that

• A necessary and sufficient condition for a magnitude function to be physically

realizable is that

Any system whose magnitude function violates eqn.(2) has a non-causal impulse response

Also for Paley-wiener criterion to be valid, the magnitude function must be square integrable
i.e.,

The following conclusions can be drawn from the Paley-wiener criterion:

1. The magnitude function may be zero at some discrete frequencies, but it cannot be
zero over a finite band of frequencies since this will cause the integral in the eqn(2) to become
infinite.

If over any finite band, then over that band leading to violation of
Paley-Wiener criterion. That means ideal filters are not physically realizable.
2. The magnitude function cannot fall off to zero faster than a function of exponential
order i.e., is permissible but is not realizable as it violates
eqn(2).
It implies, a realizable magnitude characteristic cannot have too great a total attenuation.
Hence ideal filters are not realizable (as they have 100% attenuation at cut-off) and are non-
causal

Physically realizable filter characteristics:

• The almost ideal LPF characteristics shown above are physically realizable for small values of
• It has zero values at discrete frequencies but non-zero value anywhere for a band of
frequencies
• This characteristic does not violate Paley-Wiener criterion.

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Relationship between system bandwidth and rise-time
30 May 2019 09:56

Let us consider ideal LPF and a unit step function be applied as the input to the system

The response y(t) does not rise sharply but increases gradually as the sharp rise in input which corresponds to
HF components undergo greater attenuation compared to the LF components

The time taken by the output function to reach its value depends upon the cut-off of the filter or the system
bandwidth.
• Smaller the cut-off, greater the rise time

The RISE TIME is defined as the time taken by the signal to reach the final value from its initial value
The transfer function of a LPF is equivalent to a gate function given by

Also, the F.T of a unit step function is

The F.T of the response y(t) i.e., Y( is obtaained by multiplying with

*
*

Taking Inverse Fourier Transform,

Put ⇒ ⇒
⇒

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The integral of Sampling function is a standard integral known as Sine integral given by

The response,
=

The sine integral , unit step function and the response are shown below

It is obvious that as reduces the response rises more slowly i.e., increases.
Assuming the minimum value of as initial value and maximum value be the final value, the rise time may be
expressed as

The bandwidth of the system (LPF) in Hz

RISE TIME IS INVERSELY PROPORTIONAL TO BANDWIDTH OF THE SYSTEM

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ENERGY SPECTRAL DENSITY
30 May 2019 12:52

An energy signal is one which has finite energy and zero average power i.e.,

For energy signal to finite the signal amplitude must go to 0 as i.e.,

Almost all the practical, non-periodic signals defined over a finite time are energy signals

For CT signal, the energy E is expressed as

PARSEVAL'S THEOREM FOR ENERGY SIGNALS (RAYLEIGH'S ENERGY THEOREM)

The Parseval's theorem states that the energy of a signal may be obtained with the help of its
Fourier transform
PROOF:

Consider a signal with its i.e.,

The energy E of the signal is expressed as

We know that IFT is written as

Replacing one in eqn(2) from eqn(3),

Interchanging the order of integration, we get

The integral in the bracket is equal to

We know that for a real signal the F.T and are complex conjugates i.e.,

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ENERGY SPECTRAL DENSITY:

Consider a signal applied to an ideal LPF. The transfer function of Ideal LPF is defined as

The response of the system is expressed as

Using Parseval's theorem, the energy of the output signal may be expressed as

For an ideal LPF

Also, we may assume that the F.T is constant with frequency for a narrow band to .
The energy of the signal over this narrow band will be

It is clear that represents the energy contribution due to bandwidth of the signal including the
negative components i.e., the frequency components of which lie in the narrow band are
transmitted through the filter

The energy contribution per unit banwidth

represents the energy per unit bandwidth and is therfore called ENERGY DENSITY
SPECTRUM or ENERGY SPECTRAL DENSITY measured in Joules/Hz

In general, ESD is denoted by

The total energy of the signal may be obtained by integrating over the bandwidth of the signal

For real signals, i.e., the spectrum is symmetrical and contribution of negative
and positive frequencies will be same

Ex:

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RELATIONSHIP BETWEEN ESD's OF INPUT & OUTPUT SIGNALS:

We know that

Let ESD of output be denoted by and that of input be denoted by

[Using eqn.(5)]

PROPERTIES OF ESD:

1. The total area under the ESD function is equal to the total energy of that signal.
Mathematically,
2. If is input to a LTI system with transfer function then the input and output ESD are
related as
3. The auto-correlation function and ESD form a F.T pair. Mathematically

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POWER SPECTRAL DENSITY
31 May 2019 07:10

An power signal is one which has finite average power and infinite energy i.e.,

Almost all the practical periodic signals are power signals since their average power is finite and non-zero

For CT signal, the average power P is expressed as

PARSEVAL'S THEOREM FOR POWER SIGNALS

The Parseval's theorem for power signals states that the power of a signal may be defined in terms of its Fourier Series
coefficients

PROOF:
Consider a signal
We know that where is the complex conjuagte of
The power P of the signal over one cycle is expressed as

Replacing one by its Complex Exponential Fourier Series,

Interchanging the order of integration and summation, we get

Since ,
Using (7) in (6), we get

i.e., Parseval's power theorem states that the power of a signal is equal to the sum of the square of the magnitudes of
various components present in the discrete spectrum

POWER SPECTRAL DENSITY:

The expression for power spectral density may be derived by assuming the power signal as a limiting case of an energy
signal

Consider a power signal extending to infinity. Terminate the signal such that it is 0 outside the interval i.e., the

terminated signal shown in the figure

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The terminated signal is of finite duration and hence it is an energy signal. The energy of this signal, denoted by .
We know that (Parseval's theorem)

By observation, we can say that over the interval is the same as over the interval

Taking Limit on both sides

The LHS represents the average power of

As increases, the energy of also increases. Hence increases with

As , a finite value , (say)

i.e.,

may be treated as average power per unit bandwidth

is Power Spectral Density or Power Density Spectrum

Total power of the signal is obtained by integrating the product of PSD & bandwidth over the entire
bandwidth

Since , power contribution of positive and negative frequencies is identical

Average Power is expressed as

From , it is evident that PSD of a signal retains only magnitude information of the frequency spectrum
So, all signals with identical frequency spectrum magnitude and different phase functions will have the same PSD. Thus
for a given signal, there is a unique PSD. But the converse is not true as there may be a large number of signals with the
same PSD.

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same PSD.

RELATIONSHIP BETWEEN INPUT & OUTPUT PSD's

Similar to ESD's, we have

PROPERTIES OF PSD:

1. The total area under the PSD function is equal to the average power of that signal. Mathematically,

2. If f is input to a LTI system with transfer function then the input and output PSD are related as

3. The auto-correlation function and PSD form a F.T pair. Mathematically

4.
5.
6. If is real, is real

COMPARISON OF ESD & PSD

ESD PSD
ESD gives the distribution of the energy of signal in PSD gives the distribution of the power of signal in
frequency domain frequency domain

Total Energy Total Power

where is the auto-correlation where is the auto-correlation function
function for energy signal for powersignal
Units: Joules/Hz Units: Watts/Hz

CROSS SPECTRAL DENSITY :

CROSS POWER:

PROPERTIES OF CSD:
1.
2. Real part of are even functions of
3. Imaginary part of are odd functions of
4. if are orthogonal

FREQUENCY SHIFTING THEOREM FOR POWER DENSITY SPECTRUM:

Statement:

The multiplication of a signal by a sinusoidal function of frequency shifts its power spectral density by i.e., If
a function has a PSD , then the PSD of the function is given by

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Summary of ESD & PSD
31 May 2019 09:04

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