CS Chapter3 2020

Communication
Systems
Chapter 3
Signal Transmission and Filtering
Dr. Le Dang Quang

Department of Telecommunications (113B3)
Ho Chi Minh City University of Technology
Email: ldquang@hcmut.edu.vn
Telecomm. Dept. CS-2016

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Faculty of EEE HCMUT
Chapter Outline
3.1 Response of LTI Systems
3.2 Signal Distortion in Transmission
3.3 Transmission Loss and Decibels
3.4 Filters and Filtering
3.5 Correlation and Spectral Density
3.6 Probability and Random Variables
3.7 Random Signals and Noise

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3.1 Response of LTI Systems

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Response of LTI Systems
Signal transmission is the process whereby an electrical

waveform gets from one location to another.
In the ideal case there is no distortion in the transmission.
In practice, there is some distortion in the channel which should
be controlled and compensated.
Usually, the channel is modeled by a linear time-invariant (LTI)
system. On the other hand, linear time-invariant filters are used in
different stages of the transmission system to modify the
transmitted signal.
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❑ Impulse response of linear time-invariant (LTI) system
The input signals can be, e.g., current or voltage signals.
The system is linear if
and time-invariant if
Such system is usually called as "filter". Analog filters consist of

inductors, capacitors, and resistors and they can be described by
differential equations.
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In this course we do not discuss about design or implementation of

the filters but they are considered as a black box having a certain
impulse response.
The impulse response h(t) is the response of the system to the

unit impulse:
The response of the system for an arbitrary input signal can be

calculated using the convolution integral (superposition integral):

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❑ Transfer function
The convolution integral, which determines the response of the
system in the time domain is difficult to calculate and it is intuitively
difficult concept.
The transfer function determines the response of system in the

frequency domain and it is easy to calculate and understand.
The transfer function of the linear time-invariant system is

determined by the Fourier transform of the impulse response:
In practice, it is assumed that systems are stable which means

that their impulse responses converge with time.

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Usually, the impulse response of the system is real function.

Therefore, it has the following properties:
According to the convolution theorem, the linear time-invariant

system can be characterize in the frequency domain by
The output spectrum equals the input spectrum multiplied by the

transfer function.

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❑ The 'steady-state' response of the filter
Consider the response of the filter to the complex exponential:
The 'steady-state' response of the system (i.e., the response

after the transients) for the above-mentioned signal is
The output signal is a complex exponential whose frequency is the

same as in the input but the amplitude and phase depend on the
transfer function of the system:

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For the real sinusoidal signals, this can be given by
where Ay and y are as before.

According to the superposition and the results above, there are
only those frequencies in the output signal which are also in
the input signal.
❑ The frequency response of the system

H(f0) is the response of the system for the sinusoidal signal whose
frequency is f0.
H(f) is also called the frequency response of the system which is
usually illustrated by two functions in the frequency domain:
amplitude response: | H(f)|
phase response: arg H(f)

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The effect of the filter to the input signal can be studied by considering both
the frequency response of the system and the spectrum of the input signal:
a. The bandwidth of the filter B is

much higher than the bandwidth W
of the signal, so there is no distortion.
b. B = W → The high frequencies of
the signal are attenuated.
c. The bandwidth of the filter B is
much lower than the bandwidth W of
the signal → the output signal now
looks like the filter’s impulse response
→ we can reasonably model the input
signal as an impulse.

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❑ Block-diagram analysis
Telecommunication systems consists of many blocks which are
connected together. In the case of the
LTI system, the transfer
function of the overall
system can be calculated
from the transfer functions
of the subsystems.
In the figure nearby are the

overall transfer functions of
the blocks which are
connected in:
a) parallel,
b) cascade, and
c) feedback:
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Example: Zero-order Hold circuit
The hold circuit if widely used in telecommunication applications.

The following time and frequency-domain block diagrams can be
given for this circuit:

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The transfer function of the system is:
The amplitude response of the system is sinc-function!
The impulse response is:
The impulse response is a rectangular pulse, which could be also

obtained from the Fourier transform of the above transfer function.

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3.2 Signal Distortion in
Transmission

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Signal Distortion in Transmission
❑ Distortionless channel:
Distortionless transmission means that the output signal has the
same “shape” as the input, but there can be some attenuation.
The received signal is thus
where K are td constants, and x(t) is the transmitted signal. The

spectrum of the received signal is
and therefore, the transfer function of a distortionless channel is
The amplitude and phase responses are

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The amplitude response of the distortionless system is a
constant and the phase response is a linear function of the
frequency.
These conditions have to be met only in those frequency bands

where the transmitted signal has components.
Example: The spectrum of a voice signal

Typical energy spectral density for voice signal:

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❑ Distortions
The are three major types of distortion in transmission:
❖ Amplitude distortion:
❖ Phase distortion:
❖ Nonlinear distortion: Caused by nonlinear elements of the

system.
The first two types are linear distortions, and they can be
analyzed by using the transfer function of the system. Nonlinear
system does not have transfer function, so the analysis of this kind
of system has to be done using some other methods.

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Example
The frequency response of the system:
This system is distortionless in the band 20≤ f ≤30 kHz.

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❑ Amplitude distortion
In amplitude distortion, some frequency
bands of the signal are attenuated or
amplified. Typical example is the
attenuation / amplification of the low
or high frequencies. In the time domain,
the attenuation of high frequencies can be
seen as a smoothing of the rapid changes.
In the upper figure we have the fundamental
Low frequencies attenuated
frequency and two harmonics, together with
the resultant signal (thick line).
In the lower figures, there are some

examples of the effect of the amplitude
distortion to the rectangular pulse train. High frequencies attenuated
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❑ Phase distortion
Phase distortion causes, e.g., different delays for the different harmonics
of the signal, and therefore, destroys the shape of the signal.
The phase delay of the distortion-less system is a

constant.
It should be pointed out that this is different from the constant phase shift
which in general causes distortion of the signal.
In the example of the previous page,

90o constant phase shift leads to the
result in the figure. This looks more
like triangular pulse than rectangular.
Note also that phase distortion can cause the maximum of the signal to be
much higher than before the distortion.
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❑ Equalization
If the transfer function of the channel that causes linear distortion is
known, it can be compensated by the inverse transfer function.
If the transfer functions of the channel and equalizer are HC(f) and Heq(f),
respectively, then the transfer function of the overall transmission
system is
H(f) = HC(f) Heq(f)

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When the goal is to obtain an ideal channel, the transfer function of the
equalizer is chosen to be
The limitation is the fact that in those frequencies where the

attenuation of the channel is high, equalization does not work. If the
signal is covered by higher level noise, it can not be compensated.

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❑ Transversal filter as equalizer

Transversal filter consists of tapped-delay-line. The output is a weighted
average of delayed signals:
The output in this simple case is:

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Therefore the transfer function of the equalizer is
Generally, the transfer function of transversal filter can be given by
This is a Fourier series whose period in the frequency domain is 1 / Δ.
In most of the cases, the tap gains are readjusted to compensate the
changes in the channel response. Such a system is called as adaptive
equalizer.
Transversal filters are usually implemented digitally, however, there are also
some analog implementations (CCD, SAW).

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❑ Multipath distortion
Radio systems sometimes suffer from multipath distortion caused by two
or more propagation paths between transmitter and receiver. The same
phenomenon can occur in cables caused by impedance mismatch.
For two multipath components, the received signal is
The transfer function of the channel is
where t2 > t1, k = K2 / K1 and t0 = t2 − t1.
The transfer function of the equalizer is
Only a finite number of terms are needed from this series, and this number
depends on k. Equalization can be implemented using transversal filter.
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❑ Nonlinear distortion
Here we consider nonlinear functions: y(t) = T[x(t)] which are
memoryless, i.e., the response at time t does not depend on the
previous signal values (nor on the future values). A nonlinear
function can be illustrated by a curve. Typical example:
Under small-signal input conditions: piece-wise linear

approximation.

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More general: polynomial approximation.
Any non-linear function can be approximated by the polynomial:
y(t) = a1x(t) + a2x2(t) + a3x3(t) + ...
From this polynomial, the spectrum of the output is (using
convolution theorem):
Y(f) = a1X(f) + a2X∗X(f) + a3X∗X∗X(f) +...
If the bandwidth of the input signal is W, then the bandwidth of the
output signal of any linear system contains no frequencies
beyond W.
In a nonlinear system, there can be also some other frequency
components which are not present in the input. In the above
equation, the bandwidth of the term X ∗ X (f) is 2W and, in general,
the bandwidth of the nth term is nW. The frequency components
caused by nonlinearity can be filtered out if they are situated outside
the frequency band of the input signal.
However, if, for example, X ∗ X (f) contains components for | f | < W,
these components are overlapping with those of the original input
signal X(f). They form the nonlinear distortion.
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If the input signal is a pure sinusoidal signal, then nonlinearity
produces its harmonics. According to the equations in the previous
page, we can write, if the input is a cosine wave:
This is called as harmonic distortion. Harmonic distortion can be

determined by the amplitude ratio of the harmonics and the
fundamental (i.e., the coefficients of cosω0t term). Most significant
are usually the second-harmonic distortion terms.
Now, if the input signal consists of the sinusoidals having the

frequencies of f1 and f2, then the output consists of the harmonics
as well as the linear combinations of these frequencies:

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❑ Companding
Companding = Compressing + Expanding
It is needed to deal with the non-linearities in the system (very
useful concept later on when discussing the exponential
modulations).
Sometimes nonlinear blocks are used in telecommunication

systems. For example, high signal levels can be decreased by
compressing them using the following nonlinear function:
Compressing decreases the effect of

nonlinearities of the channel. The
compressor has a higher amplification
at low signal levels than at high signal
levels.

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The inverse operation, expanding, is done in the receiver.
The use of companding allows signals with a large dynamic range

to be transmitted over facilities that have a smaller dynamic range
capability. Companding reduces the noise and crosstalk levels at
the receiver.

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3.3 Transmission Loss and
Decibels

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Transmission Loss and Decibels
❑ Power gain, decibels
In addition to the distortions mentioned above, the power level
of the signal is reduced in the transmission system (transmission
power loss). This attenuation is not a problem, it can be
compensated by amplification of the received signal. The
problem is, however, that the noise is amplified at the same
time.
The power gain is determined by
where Pout and Pin are the powers of the output and input
signals.

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The power gain is often expressed in decibels (dB):
g gdB
1000 30
1 0
0.1 -10
0.01 -20
Decibels are also used for amplitude responses:
Decibels represent power ratios.

The power itself can be expressed in dBm if it is divided by one
miliwatt:

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❑ Transmission loss
The inverse of the power gain is the attenuation (transmission
loss) L:
In the case of transmission cables, the output power decreases

exponentially with distance l:

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The attenuation in cables is compensated using repeaters.
The output power of the above system can be calculated as:

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❑ Radio transmission
Signal transmission by radiowave propagation can reduce the
required number of repeaters and has the advantage of
eliminating long cables.
Consider the line-of-sight (LOS) propagation:
The attenuation of this transmission is (free space loss):
where λ is the wavelength, f is the signal frequency, c is the

speed of light, and l is the distance (logarithmic increase of LdB
with distance).
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3.4 Filters and Filtering

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Filtering
Every communication system includes one or more filters to separate
the desired information signal from noise, distortion, and other
information signals.
❑ Ideal filter
An ideal filter has the characteristics of distortionless
transmission over one or more specified frequency bands and zero
response at all other frequencies. The transfer function is thus
The bandwidth of the

filter is B = fu− fl

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Filtering
Ideal filters are physically unrealizable, in the sense that their
characteristics cannot be achieved with a finite number of elements.
For example, the impulse response of an ideal filter is a sinc-function,
which is infinite long and noncausal.
The filters used in practical applications must be causal, that is

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Filtering
The bandwidth is usually defined to be the 3 dB-bandwidth.

The amplitude response at the edge frequencies fl and fu is
then lowered by 3 dB (or 0.707 in the linear scale) compared
to the maximum value of the passband.

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Filtering
❑ Butterworth filter
A Butterworth filter is the simplest of the standard (realizable)
filter types which is also easy to analyse. The amplitude
response of the nth-degree Butterworth filter is:
The amplitude response is maximally flat, meaning that its first

n derivatives equal to zero at f = 0.

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Filtering
Third-order Butterworth LPF

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Filtering
Bode diagram of Butterworth filter

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Filtering
❑ Other filter types
If the amplitude response in the passband and stopband varies
between certain maximum and minimum values, the filter is called
an equiripple filter.
❖ Elliptic filters (Cauer filters) are equiripple in the passband
and stopband. These filters provide the best selectivity in the
sense that their transition bands are the sharpest for a given
filter specifications.
❖ Other filter types are the Chebyshev filters which are
equiripple in the passband and maximally flat in the
stopband (or vice versa.)
❖ The filters mentioned above are based on the approximation of
an ideal amplitude response. When the aim is a good
selectivity, the phase response may be poor. Bessel filters
have a better phase response (linear phase and constant group
delay).
In critical applications, the filter is optimized according to both the
amplitude and phase response. Frequency transforms may be
used to transform a prototype LPF into HPF, BPF, BSF, etc.

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Filtering
❑ Filter implementation
Implementation of the analog filters can be based on
❖ Passive RLC-circuits: usually used for high frequency
(RF) applications because of the limited frequency range of
active circuits and good behaviour of inductors at high
frequencies.
❖ Active filters: are used at low frequencies (audio and
video) because inductors exhibit a significant resistive
component at low frequencies.
❖ Crystal filters.
❖ There are special implementations structures for microwave
filters.

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Filtering
❑ Pulse response and risetime
The spectrum of a rectangular pulse has high frequency
components. This applies also to other signals having sharp
changes, such as the unit step function. Filtering these signals
smoothes the sharp corner. This smoothing effect has to be
studied in the time domain.
For example, the step response of a first order RC filter is
Another example is the step response of an ideal lowpass

filter:
Here μ = 2B and

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Filtering
The step responses for the first order filters and for the ideal
lowpass filters:
The first-order filters do not have a good attenuation in high

frequencies, and thus, their step responses rise fast. For
example, the step response of a higher-order Butterworth filter is
closer to the step response of the ideal lowpass filter.

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Filtering
The risetime is a measure of the “speed” of the step response. It
is usually defined in the time interval where the output signal rises
from 10% to 90% of the final value.
The risetimes for the first order and ideal lowpass filters are tr ≅
0.35 / B and tr ≅ 0.44 / B, respectively.
The following approximation can be used for the risetime of an

arbitrary lowpass filter:
The risetime of a steep slope at corners is faster (shorter rise

time) when more high-frequency harmonics are included in the
Fourier series. We loosely describe any waveform with sharp
corners and short risetime as having “large bandwidth”.

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Filtering
Conversely, when we remove high frequency components from a

waveform via filtering devices (i.e., good attenuation at high
frequencies), we:
❖ reduce the bandwidth
❖ increase the risetime
❖ change the waveform to some extent
Pulse response
The pulse response is the filter response to a rectangular pulse (make
distinction with impulse response!). The rectangular pulse can be
constructed from two unit-step functions.

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Filtering
The pulse response of the ideal lowpass filter is
When Bτ > 2 the pulse response is close to the rectangular pulse. For
lower values of Bτ , the pulse response becomes flat.

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Filtering
❑ Required bandwidth for a pulse signal

Rule of thumbs for the required bandwidth of a pulse:
(1) If the shape of the pulse is desired to be preserved, then the
required bandwidth is high:
(2) The pulse is only detected or its amplitude is measured, then

smaller bandwidth can be used:
This is the condition for resolving two

pulses separated by τmin or more.
It is assumed here that the pulse response of the filter is close to

the ideal response. If this is not the case, the pulse contains
more distortion.
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3.5 Correlation and Spectral
Density

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Correlation and Spectral Density
Here we study signals using the time average and signal power
(or energy). This leads to the concept of spectral density
functions. Spectral densities allow us to deal with a broader range
of signal models, not necessarily Fourier transformable (e.g.,
random signals).
❑ Time average
The time average of an arbitrary signal is defined:
It has the properties:
The average power of power signal (i.e., signal with finite

power) is:

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Example: The time average of a sinusoid signal:
The average power of a sinusoid signal:
❑ Scalar product
The scalar product of the power signals v(t) and w(t) is
denoted by v(t)w∗(t). It is real or complex and it measures the
similarity between the two signals. The Schwarz's inequality
relates the scalar product and the signal powers:
Equality holds when v(t) = aw(t).

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Let's calculate the power of the signal:
z(t) = v(t) − aw(t)
which is
when the value of the scalar product is large, the power of signal
difference v(t) − w(t) is small, thus the signals are similar.
❑ Correlation functions of a power signal

The crosscorrelation is defined as:
It is a function of the delay parameter 𝜏 and it has the properties:

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The crosscorrelation measures the similarity between v(t) and
w(t−𝜏) as a function of the time shift 𝜏.
As a special case, we have the autocorrelation function:
Autocorrelation tells us something about the time variation of the

signal. If the autocorrelation is high for some value of 𝜏, then the
original and delayed signals (when delay is 𝜏) are similar.
❖ Properties of the autocorrelation function:
The autocorrelation of real functions is real and even. The

autocorrelation of periodic functions is periodic.
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❖ Autocorrelation of sum and difference signals:
If v(t) and w(t) are uncorrelated that is
then
Superposition of average powers therefore holds for

uncorrelated signals.

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❖ Correlation of complex exponentials:
The scalar product of two complex exponentials is
Let's consider the following two signals (complex exponentials):
where Cv and Cw are complex constants which determine the

amplitude and phase of the signals. The crosscorrelation is
These signals are uncorrelated if their frequencies are not

equal.
The autocorrelation is
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❖ Autocorrelation of sinusoidal signals
The autocorrelation of the sinusoidal signal:
is
Its maximum value is:
Autocorrelation is independent on the phase of the

signal. This emphasizes the fact that the autocorrelation
does not uniquely define a signal.

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❑ Correlation functions of energy signals
If the energy of the signal is finite, its time average is zero. In
this case, the crosscorrelation and autocorrelation are defined
as follows:
and they have the same properties as the correlation functions of

power signals. In the above equations, the power (Pv) is replaced by
the energy (Ev). For example,
In this case, the crosscorrelation can be calculated by using the

convolution:

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Based on the properties of Fourier transform and on Schwarz

inequality, we can write:

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❑ Input-output correlation
For the LTI systems, the crosscorrelation between the input and
output, as well as the autocorrelation of the output, are given by

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❑ Spectral density functions
Spectral density function Gx(f) represents the distribution of
the power or energy in the frequency domain. The area under
Gx(f) equals the average power or total energy.
It can be shown that the autocorrelation function and its

spectral density function form a Fourier transform pair:
that is
For linear time-invariant system, the spectral density of the output is

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❑ Power and energy spectrum
If the Fourier transform of energy signal v(t) is V(f) then its energy
spectral density is.
If the Fourier series coefficients of periodic power signal v(t) are

cn = c(nf0) then
This spectrum consists of impulses and the amplitude of these

impulses represents the power of harmonic components.
For example, in the case of sinusoidal:

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The power in a specified frequency band of signal x(t) is
obtained by integrating its power spectral density over that
frequency band:
Example: Comb filter

The impulse response and transfer function of this filter are

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The square of the amplitude response is
and it is illustrated in the figure in previous slide (that’s why the name
of “comb”).
The power spectral density of the output signal is:
And the autocorrelation function of the output is:
and the output power or energy is

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3.6 Probability and Random
Variables

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Probability and Random Variables
So far we have considered deterministic signals whose behaviour is

known for all possible times.
Random signals occur in communication systems both as unwanted

noise and as desired information-bearing signals.
Example: Discrete-time random signals (a) and the dependence between the
samples (b). See figures in next slide.
Next we will shortly review the basics of probability and random

variables.

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❑ Probability
The probability of event A can be thought as relative frequency
of occurrence of the event for a long set of trials (e.g., when
tossing a coin, the event A could represent heads up):
where N is the number of experiments and NA is the number of

events A.
Example: Probability of throwing a 4 and a 6 simultaneously with 2
dices equals 2/36.
Probability is the likelihood that the event will occur.
Two Conditions:
• Its value is between 0 and 1.
• The sum of the probabilities of all events must be 1.

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The joint probability of events A and B, denoted by P(AB), is
the probability that both events occur.
Events A and B are statistically independent if

P(AB)= P(A)P(B)
Example: Two cards are selected at random from an ordinary
deck of 52 playing cards. After the first card is selected, it is put
back in the deck and then we select the second card. A is the
event that the first card is a heart and B the event that the
second card is also a heart. Note: If the first card is taken out
from the deck of cards after it is selected (and not put back),
then the two events are dependent.
Conditional probability P(BA) is the probability of B

conditioned by the fact that A has occurred:

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❑ Random variables (RV)
RV = A numerical description of the outcome of an experiment.
Here we consider mainly continuous random variables
(random variable can be also discrete, i.e. having a countable
number of distinct values).
For example, one might spin a pointer and measure the final
angle θ. Possible random variables include for example:

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A random variable is determined by its cumulative distribution
function (CDF):
or by its probability density function (PDF):
For instance, the probability

P(a < x ≤ b) corresponds to the
area under the PDF.

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We can deal with discrete RVs (or combination of discrete and
continuous) by adding impulses to the PDF. If the probability of the
discrete point x0 is PX(x0) then the PDF has a term PX(x0)δ(x − x0).
Uniform distribution:
In the previous example (i.e., spinning a pointer), all angles between
0 ≤ θ < 2π are equally likely. Therefore,
the PDF is a constant in the interval
0 ≤ θ < 2π . It is always required that:
which holds in this case if

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❑ Transformations of random variables
Let X be a known random variable and it is used to derive
another random variable:
Z = g(X)
We will assume that g(X) is a monotonic function so that the
inverse function g−1(Z) exists.
Problem: Given pX(x), find pZ(z).

By using the fact that
we can write:

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In the case of linear transform:
If g(X) is not a monotonic function, it can be divided into

monotonic parts. The overall density function is the sum of the
density functions of the individual parts.
❑ Simple transformations of random variables

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❑ Joint and conditional PDF's
The joint probability density function of X and Y is denoted
by pXY(x,y). For instance
If X and Y are statistically independent:
The conditional PDF is given by
It can be also shown that

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❑ Statistical averages
Here we consider some characteristic numbers of random
variables. The mean or average of the random variable X is
denoted by
and it is calculated by
The mean of the output random variable g(X) after the transform
g() is given by
The nth moment of X is

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Besides the first moment (that is the average) one important

moment is the second moment:
which is the mean square value of X. Note that this is different from
the square of average:
❑ Standard deviation and variance

Standard deviation σX is a measure of the spread of observed
values of X relative to mX. The square of the standard deviation is
called the variance. It is defined as
Variance can be also calculated by using the mean square and

square of average:

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❑ Multivariate expectations
Consider a function of two random variables g(X,Y). Its mean is
If X and Y are independent:
Example: For the sum of two random variables Z = X + Y:
If X and Y are independent:

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❑ Gaussian PDF
Normal or Gaussian distribution has the density function of
where m and σ2 are the mean and variance. This PDF has even
symmetry at the average value. It is “bell shaped”.

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The probabilities of the from
can not be calculated analytically for

normal distribution. This kind of
probabilities can be calculated using
normalized function:
whose values can be found from tables (see Table T6, [1]).

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For large values of k (k > 7), we can use the approximation:

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3.7 Random Signals and Noise

98
Random Signals and Noise
❑ Random signals and noise
All meaningful communication signals are random as viewed
from the receiving end. Furthermore, all communication
systems suffer from noise.
It is therefore of vital importance to know the characteristics and

mathematical foundation of random signals when studying
communication systems.
A random process (or a stochastic process) is a signal

whose behaviour is somehow random.

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Example: An example could be a set of voltage waveforms
generated by thermal electron motion in identical resistors:

100
❑ Random processes
It is not usually known what the time function behind the
random process is. Therefore, at time t1 you could expect any
value from the ensemble of process values (or ensemble of
realizations). The value of the signal at t1 is thus a random
variable V1 and at t2 it is a random variable V2 etc.
Here random processes are denoted like deterministic signals,

e.g., v(t). The context will make it clear when we are talking about
random process rather than deterministic signal.

101
❑ Ensemble averages and correlation functions
Here pV(v;t) denotes the PDF of a random process as a function
of time. The ensemble average of a random process is
Here t is a constant during the expectation operation. In the

general case, this function is dependent on the time.
The autocorrelation function of the random process is (the
signal is assumed to be real)
The above autocorrelation function differs from the

autocorrelation of the deterministic signal in the sense that: we
now deal with ensemble average instead of time average; this
autocorrelation depends on t1 and t2 rather than the time
difference τ.

102
The crosscorrelation of two random processes v(t) and w(t) is
determined by
❑ Stationary processes
The statistical characteristics of a stationary process remain
invariant over time. If this holds for all characteristics, then the
process is stationary in strict sense.
Among other consequences, it follows that:

(1) The mean value must be independent of time:
(2) The autocorrelation function depends only on the time

difference:

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If both of the above conditions hold (i.e., the mean is constant in

time and the autocorrelation depends only on the time
difference), then the process is stationary in wide sense (or
WSS).
It also follows from condition (2) that:
Thus the square average and variance are independent of time.
The autocorrelation function of a stationary signal has the same

properties as a deterministic signal. From now on, by stationary
we mean WSS (unless otherwise indicated).

104
❑ Ergodic random process
If all ensemble averages equal the corresponding time
averages, the process is said to be ergodic. Then
where vi(t) is an arbitrary signal of a random process.
An ergodic process must be strictly stationary, but stationarity

does not guarantee ergodicity.
Here we assume that all random processes are ergodic.

105
❑ Gaussian process
A Gaussian process is a random process for which:
pV(v;t) is a Gaussian PDF for all t
A Gaussian process has the following properties:
❖ The process is completely described by mean and
autocorrelation
❖ If the process satisfies the conditions for wide-sense
stationarity then the process is strictly stationary and ergodic
❖ Any linear operation produces another Gaussian process.
Notes:
1) Gaussian process has many properties that make analytic
results possible.
2) Random processes produced by physical phenomena are
often such that a Gaussian model is appropriate.

106
❑ Central Limit Theorem:

Suppose X1, X2,…, Xn are a set of zero mean independent, identically
distributed random variables with some common distribution.
Consider their scaled sum:
X + X2 ++ Xn
Y= 1 .
n
Then asymptotically (as n →∞):
Y → N (0, 2 ).
The central limit theorem states that a large sum of independent
random variables each with finite variance tends to behave like a
normal random variable. Thus the individual PDFs become
unimportant to analyze the collective sum behavior. If we model the
noise phenomenon as the sum of a large number of independent
random variables (eg: electron motion in resistor components), then
this theorem allows us to conclude that noise behaves like a Gaussian
r.v.

107
❑ Signal power and time average:
Consider a stationary random signal. This kind of signal can not
be timelimited so its energy can not be finite. Therefore, it is a
power signal.
In general, the average power of a random signal is

determined as an ensemble average over the power of the time
function produced by the random process,
For stationary process:

108
If the process is also ergodic, it has the following properties

concerning all time functions of the process:
❖ Its mean is equal to the DC component:
❖ The mean squared equals the DC power:
❖ Variance σV 2 equals the AC power.

109
❑ Noise:
All communication systems include noise which come from e.g.,
atmospheric systems, extraterrestrial radiation and other
physical phenomena. Every conducting media includes thermal
noise, which is the noise due to random motion of electrons.
The white noise is a very useful model in communications.
❑ Thermal Noise:
When a metallic resistance R is at temperature T, a noise
voltage v(t) is produced at the open-circuited terminals. v(t) has
a Gaussian distribution with zero mean and variance (power)

110
❑ White Noise:
Besides thermal resistors, many other types of noise sources
have a flat spectral density over a wide range of frequencies.
Such noise is called white noise, by analogy to the white light.
The spectral density of white noise in general is defined as
where the factor ½ included to

indicate two-sided spectrum.
Thus, the autocorrelation
function is

111
White noise is the least predictable signal: any two different

samples of a Gaussian white noise signal are uncorrelated.
The white noise has a constant power spectral density.
When the white noise is applied to a linear time invariant filter,

the resulting output signal will be described by
Thus, the spectral density of filtered noise has the shape of the
square of the filter amplitude response. We call the filtered white
noise as coloured noise.

112
Example 1: In the case of ideal filter:
So, the output noise power is directly proportional to the filter‘s

bandwidth.

113
❑ Noise Equivalent Bandwidth:
Now, we use the symbol N for the average noise power
The noise equivalent bandwidth is defined as
where is the centre-frequency amplitude ratio,

i.e., the voltage gain
Hence, the filtered

noise power is:
.

114
Signal Transmission with Noise
A baseband communication refers to a system that does not
include modulation. The results obtained for baseband systems
serve as a benchmark for comparison when we arrive to the
modulation systems.
The transmission system is assumed to be linear and the noise is
additive. In practice, the noise adds to the information signal at
various points between the source and destination. For the purpose
of analysis all the noise will be lumped into one source added to the
signal at the input of the receiver.
The model of the received signal with additive noise:

115
❑ Signal-to-Noise Ratio:
The received signal contains the information signal and noise:
In general, the output power is
We make two reasonable assumptions about the additive noise:

❖ The noise comes from an ergodic source with zero mean
and power spectral density Gn(f)
❖ The noise is physically independent of the signal and
therefore uncorrelated with it.
Under these conditions:

116
The signal-to-noise ratio (SNR) is defined as
In the case of white noise (Gn(f) = η / 2), the noise power becomes:
where gR is the receiver power gain and BN is noise equivalent

bandwidth, and η can be seen as average noise power per unit
bandwidth. The above model is called additive white Gaussian
noise model (AWGN).

117
❑ Noise temperature:
The noise density may also be expressed in terms of the noise
temperature TN . The spectral density is then
Typical values are:

118
❑ Analog baseband transmission
The information source generates a message waveform x(t) that will

be reproduced at the destination.
We model the source as an ergodic process characterized by a
message bandwidth W, such Gx( f ) ≈ 0 when | f | > W.
The channel is assumed to be distortionless so that xD(t) = Kx(t − td )

119
With the above assumptions, the average signal power at the

different locations in the system is:
At the receiver, the lowpass filter reduces the noise located outside
the message bandwidth. In the case of an ideal LPF with bandwidth
W, the resulting destination noise power and SNR will be

120
SNR is commonly stated in decibels. In terms of the noise

temperature, SNR can be expressed as
Where T0 is normal room temperature 290°K.
Typical transmission requirements for selected analog signals:

121
❑ Repeaters
SNR can be expressed as
The system performance can be improved by using repeaters.

Suppose that the transmission path is divided into m equal
sections, each having loss L1, and repeaters gain is 1/ L1, SNR
is then
It should be stressed that all of our results have been based on

distortionless transmission, i.e., they represent upper bounds
on SNR.

122
Example:
Practical transmission power values would be in the range of 10 -

20mW to provide a margin of safety.

123
Bandpass Systems and Signals
We now consider transmission systems with modulation. The
purpose here is to present the characteristics and methods of
analysis unique to bandpass systems and signals. This is the
basis for the analysis of modulation methods.
❑ Bandpass Filters:
The simplest bandpass system is the parallel resonant or tuned
circuit:

124
The transfer function of the above bandpass system is:
Here f0 is centre (resonant) frequency and Q is the quality

factor.
The 3-dB bandwidth is
This depends on the Q-value (quality factor) that is typically in
range of 10 < Q < 100.
The fractional bandwidth of bandpass systems is normally
Here, B is bandwidth and fc is carrier frequency.

125
❑ Bandpass signal:
Consider a real bandpass signal vbp(t) with the spectrum Vbp( f )
located at fc:

126
The corresponding bandpass waveform looks like a sinusoid at
frequency fc with slowly changing amplitude and phase angle.
Formally, we write:
where A(t) is the envelope and φ(t) the phase, functions of time.
The envelope is defined to be nonnegative, A(t) ≥ 0. Negative
amplitudes correspond as ±180° phase changes.
Bandpass signal can be represented by a rotating phasor:

127
This leads to the following alternative representation of
bandpass signal:
where in-phase component vi(t) and quadrature component

vq(t) are:
So, there are two descriptions for bandpass signals:

(1) A(t), φ(t) envelope and phase
(2) vi(t), vq(t) in-phase and quadrature components.
Both of them are useful.
Here is the connection between them:

128
Instead of the terms in-phase and quadrature components, we often use

the terms I and Q components.

129
❑ Lowpass Equivalent Signal
The spectrum of the bandpass signal can be represented by in-
phase and quadrature components:
In order to satisfy the bandpass condition, i.e., fc − W ≤ f ≤ fc + W,

the in-phase and quadrature functions must be lowpass signals with
Lowpass equivalent signal is
In general this is a complex signal whose spectrum is
This equals the positive-frequency portion of the bandpass signal

translated down to the origin (next slide).
130

131
❑ Lowpass-to-Bandpass Transformation
The complex baseband signal can be converted to real
bandpass:
❖ In time domain:
❖ In the frequency domain:
The above transformation is the basis of linear modulation

methods.

132
❑ Bandpass Transmission
Consider a bandpass system whose transfer function is Hbp(f),
Spectrum of input signal Xbp( f ), and spectrum of output signal
Ybp(f).
Obviously,
It is usually easier to work with the lowpass equivalent spectra

related by
where

133
Example: Carrier and envelope (group) delay
Consider a bandpass system having constant amplitude ratio K

but nonlinear phase shift θ( f ):
Lowpass equivalent system is:

134
Assuming, the phase nonlinearities are relatively smooth, we
can write the approximation (Taylor series):
where carrier delay is t0 and envelope (group) delay is t1.
If

135
Transmission Loss

139
Transmission Loss

140
Bài tập 5
Cho hệ thống tuyến tính bất biến có hàm truyền H(f)
như Hình 1.

148
Bài tập 5 (tt)
a) Tính công suất của tín hiệu ngõ ra y(t) khi tín hiệu ngõ vào x(t)
= 10cos20πt – 30sin40πt (t:s).
b) Xác định biểu thức đầy đủ của tín hiệu ngõ vào x(t) để tín hiệu
ngõ ra y(t) = 28cos28πt (t:s).
c) Tìm điều kiện của tín hiệu ngõ vào x(t) để tín hiệu ngõ ra y(t)
của hệ thống trên không méo dạng, nghĩa là y(t) = Kx(t – td).
Xác định giá trị của K và td trong trường hợp này.
d) Xác định giá trị của tín hiệu ngõ vào x(t = 20ms) để tín hiệu
ngõ ra y(t) = 4sin44πt (t:s).

149
Bài tập 6
Cho tín hiệu băng gốc có tần số lớn nhất 5KHz và công suất 10mW
được truyền trực tiếp qua dây cáp có chiều dài 10km và hệ số suy hao
công suất 2dB/km (tuyến tính theo dB).
a) Tính công suất tín hiệu tại đầu cuối cáp truyền theo dBW trong
trường hợp có sử dụng 1 bộ khuếch đại công suất có độ lợi 20dB
và đặt cách đầu vào cáp truyền 2km.
b) Tính tỉ số công suất tín hiệu trên nhiễu tại đầu cuối máy thu (S/N)D
theo dB trong trường hợp mật độ phổ công suất nhiễu AWGN Gn(f)
= N0/2 = /2 = 10-11W/Hz và bộ lọc thu lý tưởng.
c) Tính số bộ lặp lại tối ưu để tỉ số công suất tín hiệu trên nhiễu tại
đầu cuối máy thu (S/N)D là lớn nhất; biết rằng các bộ lặp lại (có
cùng độ lợi bù trừ suy hao từng chặng) cách đều nhau trên đường
truyền và nhiễu AWGN ảnh hưởng như nhau với mỗi bộ lặp lại.

150
Bài tập 7
Cho hệ thống tuyến tính bất biến có hàm truyền H(f) như Hình 1.
Gợi ý: hệ số khai triển chuỗi Fourier phức của chuỗi xung chữ nhật đơn vị (độ
rộng τ) tuần hoàn (chu kỳ T0) có dạng:
cn = (τ/ T0)sinc(nf0τ), trong đó sinc(x)=sin(x)/(x).

151
Bài tập 7 (tt)
a) Tìm 1 biểu thức tín hiệu ngõ vào x1(t) khi tín hiệu ngõ ra có
dạng y1(t) = 17sin10t (t:s).
b) Vẽ dạng sóng của tín hiệu ngõ ra y2(t) khi tín hiệu ngõ vào
x2(t) = sin2250t (t:s).
c) Tìm điều kiện của tín hiệu ngõ vào để tín hiệu ngõ ra không
méo dạng, nghĩa là y(t) = Kx(t-td). Xác định giá trị của K và td.
d) Tính công suất của tín hiệu ngõ ra y3(t) khi tín hiệu ngõ vào
x3(t) là chuỗi xung chữ nhật tuần hoàn có dạng sóng minh
họa như Hình 1.
152
Bài tập 8
❑Cho hệ thống tuyến tính bất biến có hàm truyền H(f)
như Hình 1.

153
153
Bài tập 8 (tt)
a) Tìm biểu thức tín hiệu ngõ vào x1(t) khi tín hiệu ngõ ra có dạng
y1(t) = 8sin100t (t:s).
b) Vẽ dạng sóng của tín hiệu ngõ ra y2(t) khi tín hiệu ngõ vào x2(t)
= 10sin2125t (t:s).
c) Tính giá trị của tín hiệu ngõ ra y3(t = 2ms) khi tín hiệu ngõ vào
x3(t) = 20sin400t – 12cos500t (t:s).
d) Tính công suất của tín hiệu ngõ ra y4(t) khi tín hiệu ngõ vào
x4(t) là chuỗi xung chữ nhật tuần hoàn có dạng sóng minh họa
như Hình 1.
154
154
Bài tập 9
Cho các tín hiệu tương tự x1(t) = 2cos22πt (t: s) và x2(t) = 6sin6πt + 7cos7πt +
8sin8πt (t:s) lần lượt đi qua hệ thống tuyến tính bất biến có hàm truyền H(f)
như Hình 1.
a) Xác định biểu thức (theo thời gian) của tín hiệu ngõ ra y1(t).
b) Tính giá trị của tín hiệu ngõ ra y2(t = 0.125s).

155
155
Bài tập 10
❑ Cho hê thống truyền cáp với tổng suy hao L = 200dB có 10
khoảng lặp lại chiều dài bằng nhau (dùng 9 bô lặp lai giống
nhau) và tỉ số tín hiệu trên nhiễu tại đầu cuối (S/N)D = 50dB.
a) Xác định tỉ số tín hiệu trên nhiễu tại đầu cuối (S/N)D theo dB
khi không dùng bộ lặp lại.
b) Xác định tỉ số tín hiệu trên nhiễu tại đầu cuối (S/N)D theo dB
khi dùng 99 bộ lặp lại.
c) Xác định số bộ lặp lại tối thiểu để đảm bảo tỉ số tín hiệu trên
nhiễu tại đầu cuối (S/N)D ≥ 30dB.
d) Xác định số bộ lặp lại tối ưu để tỉ số tín hiệu trên nhiễu tại
đầu cuối (S/N)D lớn nhất có thể.

156
156

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Communication

Dr. Le Dang Quang

Telecomm. Dept. CS-2016

Telecomm. Dept. CS-2016

Telecomm. Dept. CS-2016

Signal transmission is the process whereby an electrical

❑ Impulse response of linear time-invariant (LTI) system

The input signals can be, e.g., current or voltage signals.

The system is linear if

Such system is usually called as "filter". Analog filters consist of

In this course we do not discuss about design or implementation of

The impulse response h(t) is the response of the system to the

The response of the system for an arbitrary input signal can be

Telecomm. Dept. CS-2016

The transfer function determines the response of system in the

The transfer function of the linear time-invariant system is

In practice, it is assumed that systems are stable which means

Telecomm. Dept. CS-2016

Usually, the impulse response of the system is real function.

According to the convolution theorem, the linear time-invariant

The output spectrum equals the input spectrum multiplied by the

Telecomm. Dept. CS-2016

The 'steady-state' response of the system (i.e., the response

The output signal is a complex exponential whose frequency is the

Telecomm. Dept. CS-2016

For the real sinusoidal signals, this can be given by

where Ay and y are as before.

❑ The frequency response of the system

Telecomm. Dept. CS-2016

a. The bandwidth of the filter B is

Telecomm. Dept. CS-2016

In the figure nearby are the

The hold circuit if widely used in telecommunication applications.

Telecomm. Dept. CS-2016

The transfer function of the system is:

The amplitude response of the system is sinc-function!

The impulse response is:

The impulse response is a rectangular pulse, which could be also

Telecomm. Dept. CS-2016

Telecomm. Dept. CS-2016

where K are td constants, and x(t) is the transmitted signal. The

and therefore, the transfer function of a distortionless channel is

The amplitude and phase responses are

Telecomm. Dept. CS-2016

These conditions have to be met only in those frequency bands

Example: The spectrum of a voice signal

Telecomm. Dept. CS-2016

❖ Nonlinear distortion: Caused by nonlinear elements of the

Telecomm. Dept. CS-2016

The frequency response of the system:

This system is distortionless in the band 20≤ f ≤30 kHz.

Telecomm. Dept. CS-2016

In the lower figures, there are some

The phase delay of the distortion-less system is a

In the example of the previous page,

Telecomm. Dept. CS-2016

The limitation is the fact that in those frequencies where the

Telecomm. Dept. CS-2016

❑ Transversal filter as equalizer

The output in this simple case is:

Telecomm. Dept. CS-2016

Generally, the transfer function of transversal filter can be given by

This is a Fourier series whose period in the frequency domain is 1 / Δ.

Telecomm. Dept. CS-2016