Chapter 1: Introduction

1.1 Digital and Analog Sources and Systems

DEFINITION: A digital information source produces a finite set of possible messages. A
telephone touchtone pad is a good example of a digital source. There are a finite number of
characters (messages) that can be emitted by this source. DEFINITION: An analog information
source produces messages that are defined on continuum. A microphone is a good example of an
analog source. The output voltage describes the information in the sound, and it is distributed over
a continuous range of values.

DEFINITION: A digital communication system transfers information from a digital source to the
intended receiver (also called the sink). DEFINITION: An analog communication system transfers
information from an analog source to the sink.

A digital waveform is defined as a function of time that can have only a discrete set of amplitude
values. If the digital waveform is a binary waveform, only two values are allowed. An analog
waveform is a function of time that has a continuous range of values.

Digital communication has a number of advantages:

• Relatively inexpensive digital circuits may be used.
• Privacy is preserved by using data encryption.
• Greater dynamic range (the difference between the largest and smallest values) is possible.
• Data from voice, video, and data sources may be merged and transmitted over common digital
transmission system.
• In long-distance systems, noise does not accumulate from repeater to repeater.
• Errors in detected data may be small, even when there is a large amount of noise on the received
signal.
• Errors may often be corrected by the use of coding.

Digital communication also has disadvantages:

• Generally, more bandwidth is required than that for analog systems.
• Synchronization is required.

The advantages of digital communication systems usually outweigh their disadvantages.

Consequently, digital systems are becoming dominant.

1.2 DETERMINISTIC AND RANDOM WAVEFORMS:

In communication systems, we are concerned with two broad classes of waveforms: deterministic
and random (or stochastic).
DEFINITION: A deterministic waveform can be modeled as a completely specified function of
time. For example, if
w(t) = A cos (wot + φ0)
Where A, ω0, and φ0 are known constants, this waveform is said to be deterministic because, for
any value of t, the value w (t) can be evaluated. If any of the constants are unknown, then the value
of w (t) cannot be calculated, and consequently, w (t) is not deterministic.
DEFINITION: A random waveform (or stochastic waveform) cannot be completely specified as
a function of time and must be modeled probabilistically.

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1.3 Block Diagram of Communication System
The elements of basic communication system are as follows:
➢ Information or input signal
➢ Input Transducer
➢ Transmitter
➢ Communication channel or medium
➢ Noise
➢ Receiver
➢ Output Transducer

Fig.1 Block diagram of Communication system

1. Information or input signal
• The communication systems have been developed for communicating useful information from one
place to other
• The information can be in the form of sound signal like speech or music or it can be in the form of
pictures.

2. Input Transducer
• The information in the form of sound, picture or data signals cannot the transmitted as it is.
• First it has to be converted into a suitable electrical signal.
• The input transducers commonly used in the communication systems are microphones, TV etc.

3. Transmitter
• The function of the transmitter block is to convert the electrical equivalent of the information to a
suitable form
• It increases the power level of the signal. The power level should be increased in order to cover a
large range. The transmitter consists of the electronics circuits such as amplifier, mixer, oscillator,
and power amplifier.

4. Communication channel or medium

• The communication channel is the medium used for the transmission of electronic signals from
one place to another.
• The communication medium can be conducting wires, cables, optical fibers or free space.
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 • Depending upon the type of the communication medium, two types of the communication system
will exist
a. Wire communication or line communication
b. Wireless communication or radio communication

5. Noise
• Noise is an unwanted electrical signal which gets added to the transmitted signal when it is
travelling towards receiver.
• Due to noise, the quality of the transmitted information will degrade. One added the noise cannot
be separated out from the information
• Hence noise is a big problem in the communication systems.

6. Receiver
• The reception is exactly the opposite process of transmission. The received signal is amplified and
demodulated and converted in a suitable form.
• The receiver consists of the electronic circuits like mixer, oscillator, detector and amplifier.

Figure: Digital Data Communication

• In this diagram three basic signal processing operations have been included. They are:

1. Source coding:
• In source coding the encoder converts the digital signal generated at the source output into another
signal in digital form.
• Different source coding techniques are PCM (Pulse code modulation) DM (Delta modulation).
2. Channel coding:
• Channel encoding is done to minimize the effect of channel noise.
• This will reduce the number of errors in the received data and will make the system more reliable.
3. Channel Modulation:
• Modulation is used for providing an efficient transmission of the signal over the channel.
• The detector is used for demodulation channel decoder and source decoder has exactly the opposite
roles to play as compared to the channel encoder and source encoder respectively.
4. Channel Demodulation: It converts received electrical signal into sequence of bits with minimum
error and maximum efficiency.
5. Channel decoder:
▪ The output sequence of digits from the channel demodulator is fed to the channel decoder. It uses
the knowledge from channel encoder and produces the output with few errors as possible.
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6. Source decoder:
▪ Source decoder converts the binary output of the channel decoder into its original analog form or
sequence of symbols.
7. Channel:
▪ The Communication channel is a physical media (wired or wireless) for communication purpose
where unwanted noise is usually added in the path.
• If the information rate is maximum Digital modulation technique can be used because due to the
digital nature of the signal, it is possible to use the advanced processing techniques such as digital
signal processing, image processing, and data compression
8. Output Transducer:
• It consists of the electrical signal at the output of the receiver back to the original form i.e. sound
or TV pictures.
• The typical example of the output transducers are loud speakers, picture tubes etc.

Figure to be added
1.4 Propagation of Electromagnetic Wave
The propagation characteristics are the result of changes in the radio-wave velocity as a function of altitude
and boundary conditions.
Earth’s atmosphere has several layers. These layers play an important role in the wireless communication.
These are mainly classified into three layers. Troposphere This is the layer of the earth, which lies just
above the ground. We, the flora and fauna live in this layer. The ground wave propagation and LOS
propagation take place here.
Stratosphere:
This is the layer of the earth, which lies above Troposphere. The birds fly in this region. The airplanes
travel in this region. Ozone layer is also present in this region. The ground wave propagation and LOS
propagation takes place here.
Ionosphere:
This is the upper layer of the Earth’s atmosphere, where ionization is appreciable. The energy radiated by
the Sun, not only heats this region, but also produces positive and negative ions. Since the Sun constantly
radiates UV rays and air pressure is low, this layer encourages ionization of particles. Importance of
Ionosphere The ionosphere layer is a very important consideration in the phase of wave propagation
because of the following reasons –
➢ The layer below ionosphere has higher amount of air particles and lower UV radiation. Due to this,
more collisions occur and ionization of particles is minimum and not constant.
➢ The layer above ionosphere has very low amount of air particles and density of ionization is also
quite low. Hence, ionization is not proper.
➢ The ionosphere has good composition of UV radiation and average air density that does not affect
the ionization. Hence, this layer has most influence on the Sky wave propagation.
The ionosphere has different gases with different pressures. Different ionizing agents ionize these at
different heights. As various levels of ionization are done at each level, having different gases, few layers
with different properties are formed in the ionosphere.
The number of layers, their heights, the amount of sky wave that can be bent will vary from day to day,
month to month and year to year. For each such layer, there is frequency, above which if the wave is sent
upward vertically, it penetrates through the layer. The function of these layers depends upon the time of
the day, i.e., day time and night time. There are three principal layers- E, F1 and F2 during day time. There
is another layer called D layer, which lies below E layer. This layer is at 50 to 90kms above the
troposphere.

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This D layer is responsible for the day time attenuation of HF waves. During night time, this D layer
almost vanishes out and the F1 and F2 layers combine together to form F layer. Hence, there are only two
layers E and F present at the night time.
For efficient radiation, the antenna needs to be longer than one-tenth of a wavelength. For example, for
signaling with a carrier frequency of the wavelength is

Radio wave propagation:

In Radio communication systems, we use wireless electromagnetic waves as the channel. The antennas of
different specifications can be used for these purposes. The mode of propagation of electromagnetic waves
in the atmosphere and in free space may be divided into the following three categories:
• The line of sight (LOS) propagation
• Ground wave propagation
• Sky-wave propagation
In ELF (Extremely low frequency) and VLF (Very low frequency) frequency bands, the Earth, and the
ionosphere act as a wave-guide for electromagnetic wave propagation. In these frequency ranges,
communication signals practically propagate around the world. The channel bandwidths are small.
Therefore, the information is transmitted through these channels has slow speed and confined to digital
transmission.
The line of Sight (LOS) Propagation
Among the modes of propagation, this line-of-sight propagation is the one, which we would have
commonly noticed. In the line-of-sight communication, as the name implies, the wave travels a minimum
distance of sight. Which means it travels to the distance up to which a naked eye can see. Then we need
to employ an amplifier cum transmitter here to amplify the signal and transmit again.
The line-of-sight propagation will not be smooth if there occurs any obstacle in its transmission path. As
the signal can travel only to lesser distances in this mode, this transmission is used for infrared or
microwave transmissions.

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Ground Wave Propagation
Ground wave propagation of the wave follows the contour of the earth. Such a wave is called a direct
wave. The wave sometimes bends due to the Earth’s magnetic field and gets reflected the receiver. Such
a wave can be termed as a reflected wave. The following figure depicts ground wave propagation.

The wave then propagates through the Earth’s atmosphere is known as a ground wave. The direct wave
and reflected wave together contribute the signal at the receiver station. When the wave finally reaches
the receiver, the lags are cancelled out. In addition, the signal is filtered to avoid distortion and amplified
for clear output.
Sky-Wave Propagation
Sky-wave propagation is preferred when the wave has to travel a longer distance. Here the wave is
projected onto the sky and it is again reflected back to the earth.
The sky wave propagation is well depicted in the above picture. Here the waves are shown to be
transmitted from one place and where it is received by many receivers. Hence, it is an example of
broadcasting. The waves, which are transmitted from the transmitter antenna, are reflected from the
ionosphere. It consists of several layers of charged particles ranging in altitude from 30250 miles above
the surface of the earth. Such travel of the wave from the transmitter to the ionosphere and from there to
the receiver on Earth is known as Sky Wave Propagation. The ionosphere is the ionized layer around the
Earth’s atmosphere, which is suitable for sky wave propagation.

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1.5 Information Measure, Channel Capacity and Ideal Communication
Systems:
As we have seen, the purpose of communication systems is to transmit information from a source to a
receiver. However, what exactly is information, and how do we measure it?
The information sent from a digital source when the jth message is transmitted is given by

Where Pj is the probability of transmitting the jth message

From this definition, we see that messages that are less likely to occur (smaller value for Pj) provide more
information (larger value of Ij). We also observe that the information measure depends on only the
likelihood of sending the message and does not depend on possible interpretation of the content as to
whether or not it makes sense.
DEFINITION: The average information measure of a digital source is

Where m is the number of possible different source messages and Pj is the probability of sending the j th
message (m is finite because a digital source is assumed). The average information is called entropy.

1.5 CHANNEL CAPACITY AND IDEAL COMMUNICATION

SYSTEMS
Shannon showed that (for the case of signal plus white Gaussian noise) a channel capacity C (bits/s) could
be calculated such that if the rate of information R (bits/s) was less than C, the probability of error would
approach zero. The equation for C is

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Where B is the channel bandwidth in hertz (Hz) and S/N is the signal-to-noise power ratio (watts/watts,
not dB) at the input to the digital receiver. Shannon does not tell us how to build this system, but he proves
that it is theoretically possible to have such a system. Thus, Shannon gives us a theoretical performance
bound that we can strive to achieve with practical communication systems. Systems that approach this
bound usually incorporate error-correction coding.
1.6 CODING
Automatic repeat request (ARQ)
Automatic Repeat reQuest (ARQ) is an error control method for data transmission that makes use of error-
detection codes, acknowledgment and/or negative acknowledgment messages, and timeouts to achieve
reliable data transmission. An acknowledgment is a message sent by the receiver to indicate that it has
correctly received a data frame.
Usually, when the transmitter does not receive the acknowledgment before the timeout occurs (i.e., within
a reasonable amount of time after sending the data frame), it retransmits the frame until it is either correctly
received or the error persists beyond a predetermined number of retransmissions.
Three types of ARQ protocols are Stop-and-wait ARQ, Go-Back-N ARQ, and Selective Repeat ARQ.
ARQ is appropriate if the communication channel has varying or unknown capacity, such as is the case
on the Internet. However, ARQ requires the availability of a back channel, results in possibly increased
latency due to retransmissions, and requires the maintenance of buffers and timers for retransmissions,
which in the case of network congestion can put a strain on the server and overall network capacity.[5]
For example, ARQ is used on shortwave radio data links in the form of ARQ-E, or combined with
multiplexing as ARQ-M.
Forward error correction
Forward error correction (FEC) is a process of adding redundant data such as an error correcting code
(ECC) to a message so that it can be recovered by a receiver even when a number of errors (up to the
capability of the code being used) were introduced, either during the process of transmission, or on storage.
Since the receiver does not have to ask the sender for retransmission of the data, a backchannel is not
required in forward error correction, and it is therefore suitable for simplex communication such as
broadcasting. Error-correcting codes are frequently used in lower-layer communication, as well as for
reliable storage in media such as CDs, DVDs, hard disks, and RAM.
Error-correcting codes are usually distinguished between convolutional codes and block codes:
Linear Block Codes
Linear block codes have the property of linearity, i.e. the sum of any two code words is also a code word
and they are applied to the source bits in blocks. Thus, the name linear blocks code. Linear block codes
are summarized by their symbol alphabets (e.g. Binary) and parameters (n,m,dmin) Where ‘n’ is the
length of the codeword in symbols, ‘m’ is the number of source symbols that will be used or encoding at
once and ‘dmin’ is the minimum hamming distance.
Early examples of block codes are repetition codes, Hamming codes and multidimensional parity-check
codes. They were followed by a number of efficient codes, Reed–Solomon codes being the most notable
due to their current widespread use. Turbo codes and low-density parity-check codes (LDPC) are
relatively new constructions that can provide almost optimal efficiency.

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Convolution Codes
The idea behind a convolution code is to make every codeword symbol be the weighted sum of the various
input message symbols. This is like convolution used in LTI systems to find the output of a system, when
you know the input and impulse response. So we generally find the output o the system convolution of
the input bit, against the states of the convolution encoder, registers. The encoder is usually a simple circuit
which has state memory and some feedback logic, normally XOR gates. The decoder can be implemented
in software of firmware. The Veterbi algorithm is the optimum algorithm used to decode convolution
codes.

Chapter 2 study from copy

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Chapter 3: Base-band Pulse and Digital Signaling
3.1 Introduction:
Digital signaling is popular because of the low cost of digital circuits and the flexibility of the digital
approach. This flexibility arises because digital data from digital sources may be merged with digitized
data derived from analog sources to provide a general-purpose communication system.

3.2 Pulse Amplitude Modulation Pulse

Amplitude Modulation (PAM) is an analog modulating scheme in which the amplitude of the pulse carrier
varies proportional to the instantaneous amplitude of the message signal. The pulse amplitude modulated
signal, will follow the amplitude of the original signal, as the signal traces out the path of the whole wave.
In natural PAM, a signal sampled at the Nyquist rate is reconstructed, by passing it through an efficient
Low Pass Frequency (LPF) with exact cutoff frequency.

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Note: Though the PAM signal is passed through an LPF, it cannot recover the signal without distortion.
Hence to avoid this noise, flat-top sampling is done as shown in the above figure.
Flat-top sampling is the process in which sampled signal can be represented in pulses for which the
amplitude of the signal cannot be changed with respect to the analog signal, to be sampled. The tops of
amplitude remain flat. This process simplifies the circuit design. The Flat top PAM is most popular and is
widely used because during the transmission of signal, the noise interfere with the top of the transmission
pulse and noise can be easily removed in this case
Sample and hold circuit generating flat top sampled PAM The figure below represents the basic circuit
for the sample and hold circuit:

In the sample mode of the circuit, the switch present is closed, and so this charges capacitor C, with the
instantaneous value of the applied input signal. However, in the hold mode of the circuit, the switch now
gets open, and so no further charging is possible. But now at the hold mode, the capacitor holds the charge
that was initially being stored at the time of sample mode. It should be understand that the stored charge
is held by capacitor rather being dissipated. So, this is because the circuit has no path for the dissipation
of the stored charge through it.
Basically, the sample and hold circuit, samples the analog signal and the capacitor present holds these
samples. This sampled value when provided to the ADC; it generates a discrete signal from an analog one.
Let’s first see the input and output response of a sample and hold circuit:

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The first diagram in the above figure represents an analog signal that is applied at the input of the sample
and hold circuit. The sampling of the applied input signal in the closed switch condition is shown in the
next image. While hold input of the analog signal under open switch condition is represented in the last
image of the above figure.
Following are the benefits or advantages of PAM:
➨In PAM, amplitudes of regularly spaced pulses are varied in proportion to corresponding sample values
of continuous message signal. Hence system is lowest in complexity to implement. Hence generation and
detection is easy.
Following are the disadvantages of PAM:
➨Noise interference is higher.
➨It is difficult to remove noise, as this will affect amplitude part which carries information.
➨It has lowest power efficiency among all three types.
➨Instantaneous power of transmitter varies.
➨Transmission bandwidth is too large.
Pulse Width Modulation
Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time Modulation (PTM)
is an analog modulating scheme in which the duration or width or time of the pulse carrier varies
proportional to the instantaneous amplitude of the message signal. The width of the pulse varies in this
method, but the amplitude of the signal remains constant. Amplitude limiters are used to make the
amplitude of the signal constant. These circuits clip off the amplitude, to a desired level and hence the
noise is limited
Pulse Position Modulation
Pulse Position Modulation (PPM) is an analog modulating scheme in which the amplitude and width of
the pulses are kept constant, while the position of each pulse, with reference to the position of a reference
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pulse varies according to the instantaneous sampled value of the message signal. The transmitter has to
send synchronizing pulses (or simply sync pulses) to keep the transmitter and receiver in synchronism.
These sync pulses help maintain the position of the pulses.

3.3 Pulse Coded Modulation

PCM is essentially analog to digital conversion (ADC) of special type where the information contain in
the instantaneous sample of an analog signal is represented by digital word serial bit stream. The
transmitter section of a Pulse Code Modulator circuit consists of Sampling, Quantizing and Encoding,
which are performed in the analog-to-digital converter section. The low pass filter prior to sampling
prevents aliasing of the message signal. The basic operations in the receiver section are regeneration of
impaired signals, decoding, and reconstruction of the quantized pulse train. Following is the block
diagram of PCM which represents the basic elements of both the transmitter and the receiver sections.

Low Pass Filter

This filter eliminates the high frequency components present in the input analog signal which is greater
than the highest frequency of the message signal, to avoid aliasing of the message signal.
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Sampler
This is the technique which helps to collect the sample data at instantaneous values of message signal, so
as to reconstruct the original signal. The sampling rate must be greater than twice the highest frequency
component W of the message signal, in accordance with the sampling theorem.

Quantizer
Quantizing is a process of reducing the excessive bits and confining the data. The sampled output when
given to Quantizer, reduces the redundant bits and compresses the value.

Encoder
The digitization of analog signal is done by the encoder. It designates each quantized level by a binary
code. The sampling done here is the sample-and-hold process. These three sections LPF, Sampler, and
Quantizer will act as an analog to digital converter. Encoding minimizes the bandwidth used.

Regenerative Repeater
This section increases the signal strength. The output of the channel also has one regenerative repeater
circuit, to compensate the signal loss and reconstruct the signal, and also to increase its strength. Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original signal. This circuit acts
as the demodulator.

Reconstruction Filter
After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a lowpass filter
is employed, called as the reconstruction filter to get back the original signal. Hence, the Pulse Code
Modulator circuit digitizes the given analog signal, codes it and samples it, and then transmits it in an
analog form. This whole process is repeated in a reverse pattern to obtain the original signal.

Following are the benefits or advantages of PCM:

➨It is robust against noise and interference.
➨ Uniform transmission quality.
➨Efficient SNR and bandwidth trade off.
➨It provides secure data transmission.
➨It offers efficient regeneration.
➨It is easy to add or drop channels.

Following are the drawbacks or disadvantages of PCM:

➨Overload appears when modulating signal changes between samplings, by an amount greater than the
size of the step.
➨Large bandwidth is required for transmission.
➨Noise and crosstalk leaves low but raises attenuation.
➨An IDN (Integrated Digital Network) can only be realized by gradual extension of noise.
➨The difference between original analog signal and translated digital signal is called quantizing error.

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The quality of a Quantizer output depends upon the number of quantization levels used. The discrete
amplitudes of the quantized output are called as representation levels or reconstruction levels. The
spacing between the two adjacent representation levels is called a quantum or step-size.

Quantization:
The digitization of analog signals involves the rounding off of the values which are approximately equal
to the analog values. The method of sampling chooses a few points on the analog signal and then these
points are joined to round off the value to a near stabilized value. Such a process is called as Quantization.
Types of Quantization
There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.
The type of quantization in which the quantization levels are uniformly spaced is termed as a Uniform
Quantization.
The type of quantization in which the quantization levels are unequal and mostly the relation between
them is logarithmic, is termed as a Non-uniform Quantization.

❖ Companding in PCM
The word Companding is a combination of Compressing and Expanding, which means that it does both.
This is a non-linear technique used in PCM which compresses the data at the transmitter and expands the
same data at the receiver. The effects of noise and crosstalk are reduced by using this technique.
There are two types of Companding techniques. They are –
➢ A-law Companding Technique
• Uniform quantization is achieved at A = 1, where the characteristic curve is linear and no compression
is done.
• A-law has mid-rise at the origin. Hence, it contains a non-zero value.
• A-law companding is used for PCM telephone systems.
➢ µ-law Companding Technique
• Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and no compression
is done.
• µ-law has mid-tread at the origin. Hence, it contains a zero value.
• µ-law companding is used for speech and music signals. µ-law is used in North America and Japan.

3.4 Differential PCM

For the samples that are highly correlated, when encoded by PCM technique, leave redundant information
behind. To process this redundant information and to have a better output, it is a wise decision to take a
predicted sampled value, assumed from its previous output and summarize them with the quantized values.
Such a process is called as Differential PCM .

DPCM Transmitter
The DPCM Transmitter consists of Quantizer and Predictor with two summer circuits. Following is the
block diagram of DPCM transmitter.

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The signals at each point are named as –
• x(nTs) is the sampled input
• xˆ(nTs) is the predicted sample
• e(nTs) is the difference of sampled input and predicted output, often called as prediction error
• v(nTs) is the quantized output
• u(nTs) is the predictor input which is actually the summer output of the predictor output and the
quantizer output
The predictor produces the assumed samples from the previous outputs of the transmitter circuit.
The input to this predictor is the quantized versions of the input signal x(nTs) Quantizer Output is
represented as –

Where q (nTs) is the quantization error

Predictor input is the sum of quantizer output and predictor output,

The same predictor circuit is used in the decoder to reconstruct the original input.

DPCM Receiver
The block diagram of DPCM Receiver consists of a decoder, a predictor, and a summer circuit. Following
is the diagram of DPCM Receiver.
The notation of the signals is the same as the previous ones. In the absence of noise, the encoded receiver
input will be the same as the encoded transmitter output.

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As mentioned before, the predictor assumes a value, based on the previous outputs. The input given to the
decoder is processed and that output is summed up with the output of the predictor, to obtain a better
output.

3.5 Delta Modulation

The sampling rate of a signal should be higher than the Nyquist rate, to achieve better sampling. If this
sampling interval in Differential PCM is reduced considerably, the sample to-sample amplitude difference
is very small, as if the difference is 1-bit quantization, then the step-size will be very small i.e., Δ delta.
The type of modulation, where the sampling rate is much higher and in which the stepsize after
quantization is of a smaller value Δ, such a modulation is termed as delta modulation.
Features of Delta Modulation
Following are some of the features of delta modulation.
• An over-sampled input is taken to make full use of the signal correlation.
• The quantization design is simple.
• The input sequence is much higher than the Nyquist rate.
• The quality is moderate.
• The design of the modulator and the demodulator is simple.
• The stair-case approximation of output waveform.
• The step-size is very small, i.e., Δ deltadelta.
• The bit rate can be decided by the user.
• This involves simpler implementation.
Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit DPCM scheme. As the
sampling interval is reduced, the signal correlation will be higher.

Delta Modulator
The Delta Modulator comprises of a 1-bit quantizer and a delay circuit along with two summer circuits.
Following is the block diagram of a delta modulator.
The predictor circuit in DPCM is replaced by a simple delay circuit in DM

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From the above diagram, we have the notations as –

Using these notations, now we shall try to figure out the process of delta modulation.

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19
Delay unit output is an Accumulator output lagging by one sample. From equations 5 & 6, we get a
possible structure for the demodulator. A Stair-case approximated waveform will be the output of the delta
modulator with the step-size as delta (Δ). The output quality of the waveform is moderate.
Delta Demodulator
The delta demodulator comprises of a low pass filter, a summer, and a delay circuit. The predictor circuit
is eliminated here and hence no assumed input is given to the demodulator. Following is the diagram for
delta demodulator.

A binary sequence will be given as an input to the demodulator. The stair-case approximated output is
given to the LPF.
Low pass filter is used for many reasons, but the prominent reason is noise elimination for out of-band
signals. The step-size error that may occur at the transmitter is called granular noise, which is eliminated
here. If there is no noise present, then the modulator output equals the demodulator input.
Advantages of DM Over DPCM
• 1-bit quantizer
• Very easy design of the modulator and the demodulator However, there exists some noise in DM.
• Slope Over load distortion (when Δ is small)
• Granular noise (when Δ is large)
The delta modulation has two major drawbacks as under:
1. Slope overload distortion
2. Granular or idle noise
Now, we will discuss these two drawbacks in detail.

1. Slope Overload Distortion

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 We can observe from fig.1 , the rate of rise of input signal x(t) is so high that the staircase signal
cannot approximate it, the step size ‘Δ’ becomes too small for staircase signal u(t) to follow the
step segment of x(t).

This distortion arises because of large dynamic range of the input signal.

Fig.1: Quantization Errors in Delta Modulation

Hence, there is a large error between the staircase approximated signal and the original input signal x(t).
This error or noise is known as slope overload distortion.
To reduce this error, the step size must be increased when slope of signal x(t) is high.
2. Granular or Idle Noise
Granular or Idle noise occurs when the step size is too large compared to small variation in the input
signal. This means that for very small variations in the input signal, the staircase signal is changed by large
amount (Δ) because of large step size. Fig.1 shows that when the input signal is almost flat, the staircase
signal u(t) keeps on oscillating by ±Δ around the signal. The error between the input and approximated
signal is called granular noise. The solution to this problem is to make the step size small.
Solution
In order to overcome the quantization errors due to slope overload and granular noise, the step size (Δ) is
made adaptive to variations in the input signal x(t). Particularly in the steep segment of the signal x(t), the
step size is increased. And the step is decreased when the input is varying slowly. This method is known
as Adaptive Delta Modulation (ADM).
The adaptive delta modulators can take continuous changes in step size or discrete changes in step size.

3.6 Line Coding

A line code is the code used for data transmission of a digital signal over a transmission line. This process
of coding is chosen so as to avoid overlap and distortion of signal such as intersymbol interference.
Properties of Line Coding
Following are the properties of line coding –
• As the coding is done to make more bits transmit on a single signal, the bandwidth used is much
reduced.
• For a given bandwidth, the power is efficiently used.
• The probability of error is much reduced.
• Error detection is done and the bipolar too has a correction capability.
• Power density is much favorable.
• The timing content is adequate.
• Long strings of 1s and 0s is avoided to maintain transparency
Types of Line Coding
There are 3 types of Line Coding
• Unipolar
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• Polar
• Bi-polar Unipolar Signaling
It has only one level as shown in figure:

Polar NRZ
In this type of Polar signaling, a High in data is represented by a positive pulse, while a Low in data
is represented by a negative pulse. The following figure depicts this well.

Bipolar Signaling
This is an encoding technique which has three voltage levels namely +, - and 0. Such a signal is called as
duo-binary signal.
An example of this type is Alternate Mark Inversion AM. For a 1, the voltage level gets a transition from
+ to – or from – to +, having alternate 1s to be of equal polarity. A 0 will have a zero-voltage level.

Biphase (Manchester and Differential Manchester ) –

Manchester encoding is somewhat combination of the RZ (transition at the middle of the bit) and NRZ-L
schemes. The duration of the bit is divided into two halves. The voltage remains at one level during the
first half and moves to the other level in the second half. The transition at the middle of the bit provides
synchronization.
Differential Manchester is somewhat combination of the RZ and NRZ-I schemes. There is always a
transition at the middle of the bit but the bit values are determined at the beginning of the bit. If the next
bit is 0, there is a transition, if the next bit is 1, there is no transition.

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3.8 Time Division Multiplexing
Time division multiplexing (FDM) is a technique of multiplexing, where the users are allowed the total
available bandwidth on time sharing basis. Here the time domain is divided into several recurrent slots of
fixed length.
In TDM, the data flow of each input stream is divided into units. One unit may be 1 bit, 1 byte, or a block
of few bytes. Each input unit is allotted an input time slot. One input unit corresponds to one output unit
and is allotted an output time slot. During transmission, one unit of each of the input streams is allotted
one-time slot, periodically, in a sequence, on a rotational basis.

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Types of TDM
1. Synchronous TD
2. Asynchronous TDM
Synchronous TDM (STDM)
1. In synchronous TDM, each device is given same time slot to transmit the data over the link, irrespective
of the fact that the device has any data to transmit or not. Hence the name as Synchronous TDM.
Synchronous TDM requires that the total speed of various input lines should not exceed the capacity of
path.

Asynchronous TDM
1. It is also known as statistical time division multiplexing.
2. Asynchronous TDM is called so because is this type of multiplexing, time slots are not fixed i.e. the
slots are flexible.
3. Here, the total speed of input lines can be greater than the capacity of the path.

E1 and T1 Hierarchy T1-

carrier is used by North America, Canada. It uses 24 channels. Transmission System 1 (T1), was
introduced in 1962 in the Bell System, and could transmit up to 24 telephone calls simultaneously over a
single transmission line of copper wire. The data rate for T1- carrier is given by 1.544 Mbps
The E1 frame defines a cyclical set of 32 time slots of 8 bits. The time slot 0 is devoted to transmission
management and time slot 16 for signaling; the rest were assigned originally for voice/data transport. The
data rate for E1- carrier is given by 2.048Mbps.

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Calculation of T1:
• In telephony system, Voice signal is given by 4khz frequency (where 3100hz for voice and 900hz for
guard band)
• And the required digital sampling rate = 8000 hz
• Each T1 frame contains 1 byte of voice data for each 24 channels. i.e. system needs 8000 frames per
second to maintain those 24 simultaneous voice channels
• Each frames of a T1 is ( 24 Channels×8bits per second+ 1 framing bit = 193)
• Since 8000 frame per second ( 193×8000 = 1.5 Mbps)

Calculation of E1-Carrier
• 32time slot each being allocated of 8 bit in turn.
• (8×8000×32 = 2048000 = 2.048 Mbps) Where 8000 is frames per second for sampling.

3.9 Packet Transmission System

A packet transmission system partitions the source data into data packets, each of which contains a
destination address header. Many users share the high-speed channel by merging their packets into a
high-speed data stream. Routers along the network read the header information on the packets and
route the packet to the appropriate destination. To accommodate high-speed sources, more packets are
sent from these sources over a given time interval, compared with only a few packets that are merged
onto the network from low-speed sources. A packet network efficiently assigns network resources
when the sources have bursty data. However, there is network overhead caused by the transmission of
the packet header information. STM networks are more efficient when the sources have a fixed data
rate (i.e., when they are not bursty). It is emphasized that packet switched transmission systems (such
as the Internet, where internet protocol is used with IP numbers) are dominating transmission systems.
This is especially true when compared with circuit switched transmission systems (The upcoming and
cost-effective way to implement telephone systems is to use VoIP (Voice over the Internet with IP
packets).

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• The packet switching is a switching technique in which the message is sent in one go, but it is divided
into smaller pieces, and they are sent individually.
• The message splits into smaller pieces known as packets and packets are given a unique number to
identify their order at the receiving end.
• Every packet contains some information in its headers such as source address, destination address and
sequence number.
• Packets will travel across the network, taking the shortest path as possible.
• All the packets are reassembled at the receiving end in correct order.
• If any packet is missing or corrupted, then the message will be sent to resend the message. o If the
correct order of the packets is reached, then the acknowledgment message will be sent.

Inter symbol Interference (ISI)

This is a form of distortion of a signal, in which one or more symbols interfere with subsequent signals,
causing noise or delivering a poor output.
Causes of ISI
The main causes of ISI are –
• Multi-path Propagation
• Non-linear frequency in channels
The ISI is unwanted and should be completely eliminated to get a clean output. The causes of ISI should
also be resolved in order to lessen its effect.
To view ISI in a mathematical form present in the receiver output, we can consider the receiver output.
The receiving filter output y(t) is sampled at time ti=iTb(with i taking on integer values), yielding

In the above equation, the first term μai is produced by the ith transmitted bit. The second term represents
the residual effect of all other transmitted bits on the decoding of the ith bit. This residual effect is called
as Inter Symbol Interference.
n the absence of ISI, the output will be –

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This equation shows that the ith bit transmitted is correctly reproduced. However, the presence of ISI
introduces bit errors and distortions in the output.

Nyquist First method (Zero ISI)

Nyquist’s first method for eliminating ISI is to use an equivalent transfer function, He(f), such that the
impulse response satisfies the condition
where k is an integer, Ts is the symbol (sample) clocking period, is the offset in the receiver sampling
clock times compared with the clock times of the input symbols, and C is a nonzero constant.
That is, for a single flat-top pulse of level a present at the input to the transmitting filter at t = 0, the
received pulse would be ahe(t).
It would have a value of aC at t = but would not cause interference at other sampling times because he(kTs
+ ) = 0 for k≠ 0.

Eye Pattern
An effective way to study the effects of ISI is the Eye Pattern. The name Eye Pattern was given from its
resemblance to the human eye for binary waves. The interior region of the eye pattern is called the eye
opening. The following figure shows the image of an eye-pattern.
Jitter is the short-term variation of the instant of digital signal, from its ideal position, which may lead to
data errors. When the effect of ISI increases, traces from the upper portion to the lower portion of the eye
opening increases and the eye gets completely closed, if ISI is very high.

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An eye pattern provides the following information about a particular system.
• Actual eye patterns are used to estimate the bit error rate and the signal-to-noise ratio.
• The width of the eye opening defines the time interval over which the received wave can be sampled
without error from ISI.
• The instant of time when the eye opening is wide, will be the preferred time for sampling.
• The rate of the closure of the eye, according to the sampling time, determines how sensitive the
system is to the timing error.
• The height of the eye opening, at a specified sampling time, defines the margin over noise.
Hence, the interpretation of eye pattern is an important consideration.

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Chapter-4 Principles of Signaling and Circuits

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Chapter-5 AM,FM and Digital Modulation System

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Digital Modulation Techniques

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Chapter 6:
Wire and Wireless Communication system
Introduction

(i) Radiowaves – These are easy to generate and can penetrate through buildings. The sending and
receiving antennas need not be aligned. Frequency Range:3KHz – 1GHz. AM and FM radios and
cordless phones use Radiowaves for transmission.
Further Categorized as
(i)Terrestrial (ii) Satellite.
(ii) Microwaves – It is a line of sight transmission i.e. the sending and receiving antennas need to be
properly aligned with each other. The distance covered by the signal is directly proportional to the height
of the antenna. Frequency Range: 1GHz – 300GHz. These are majorly used for mobile phone
communication and television distribution.

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(iii) Infrared – Infrared waves are used for very short distance communication. They cannot penetrate
through obstacles. This prevents interference between systems. Frequency Range: 300GHz – 400THz. It
is used in TV remotes, wireless mouse, keyboard, printer, etc.

6.1 THE EXPLOSIVE GROWTH OF TELECOMMUNICATIONS

The explosive growth of both wire and wireless telecommunications has been sparked by the need for
personal communication of voice, video, and data and the availability of low-cost integrated circuits and
microprocessors. There is a relentless push for higher speed data transmission over wire and wireless
systems. This is caused not only by the growth of traditional voice, video, and data systems, but also by
the convenience of personal and smart cellular phones, e-mail with file attachments, and the use of the
internet.
6.2 Telephone Systems
Modern telephone systems have evolved from the telegraph and telephone systems of the 1800s.
Telephone companies that provide services for a large number of users over their public switched
telephone networks (PSTN) on a for-hire basis are known as common carriers. The term is applied to
widely diverse businesses, such as mail, airline, trucking, telephone, and data services. The common
carriers are usually regulated by the government for the general welfare of the public, and, in some
countries, certain common carrier services are provided by the government. The information from multiple
users is transmitted over these systems, primarily by using time-division multiplex (TDM) transmission
or packet data transmission.
Historically, telephone systems were designed only to reproduce voice signals that originated from a
distant location. Today, modern telephone systems are very sophisticated. They use large digital
computers at the central office (CO) to switch calls and to monitor the performance of the telephone
system. The modern CO routes TDM PCM voice data, video data, and computer data to remote terminals
and to other central offices. Digital service is provided to the customer in any of three ways:
(1) a dedicated leased circuit, such as a T1 circuit, that is available for use at all times without dialing or
switching,
(2) a circuit-switched service that is available on a dial-up basis, and (3) a packet-switched service such
as DSL, discussed in Section 8–3, that is “always on” and is used only when packets are exchanged.

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The simple two-wire telephone system shown in Fig. 8–1 has three important advantages: (1) it is
inexpensive; (2) the telephone sets are powered from the CO via the telephone line, so no power supply
is required at the user’s location; and (3) the circuit is full duplex. The two-wire system has one main
disadvantage: Amplifiers cannot be used, since they amplify the signal in only one direction.
Consequently, for distant telephone connections, a more advanced technique—called a four-wire circuit—
is required. In a four-wire circuit, one pair (or one optical fiber) is used for signals sent in the transmit
direction, and another is used for arriving signals in the receive direction.
6.3 Digital Subscriber Line
The PSTN and supporting local access networks have been designed with guidelines that transmissions
are limited to an analog voice channel 3400 Hz. For example − Telephones, Modems, Dial Fax Modem
and Private Line Modems have limited their transmissions on local access telephone lines to the
frequency spectrum between 0 Hz and 3400 Hz. The highest information rate possible using 3400 Hz
frequency spectrum are less than 56 Kbps.DSL eliminate the limit of 3400 Hz frequency boundary, much
like the traditional T1 or E1, which uses a much wider range of frequencies than the voice channel.
Digital Subscriber Line (DSL, originally, digital subscriber loop) is a communication medium, which is
used to transfer internet through copper wire telecommunication line. Along with cable internet, DSL is
one of the most popular ways ISPs provide broadband internet access.
• Its aim is to maintain the high speed of the internet being transferred.

• It uses splitters or DSL filters (shown in below diagram).Basically, the use splitter is to splits the
frequency and make sure that they can’t get interrupted.

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1. HDSL (high-bit-rate digital subscriber line) uses two twisted pairs (one transmit and one receive) to
support 1.544 Mbps at full duplex for a distance of up to 12,000 ft from the RT.
2. SDSL (symmetrical digital subscriber line) is a one-pair version of HDSL. It provides full duplex to
support 768 kbps in each direction using a hybrid or echo canceller to separate data transmitted from data
received.
3. ADSL (asymmetrical digital subscriber line) uses one twisted pair to support 6 Mbps sent downstream
to the customer and 640 kbps sent upstream over a distance of up to 12,000 ft. The ASDL spectrum is
above 25 kHz.
4. VDSL (very-high-bit-rate digital subscriber line) uses one pair of wires to support 25 Mbps downstream
for distances of up to 3,000 ft from the RT or 51 Mbps downstream for distances of up to 1,000 ft from
the RT. Up to 3.2 Mbps can be supported upstream.
5. ISDN (integrated service digital network) uses one twisted-pair line to provide a subscriber data rate of
up to 144 kbs in each direction at distances of up to 18,000 ft from the RT. This technology has been
available since 1990 and has been popular in Europe and Japan, but not too popular in the United States.
6.4 Capacities of Public Switch Telephone Number
A Public Switched Telephone Network, or PSTN for short, refers to a telecommunications network which
allows subscribers at different sites to communicate by voice. The term plain old telephone service (POTS)
is also frequently used. The features of a PSTN are:
• Subscribers can be connected by entering telephone numbers
• The existing connections are primarily used to transmit speech information
• After hanging up the connection is closed and the resources used become available to other subscribers

The term public switched telephone network is primarily used for public landlines.
Most automated telephone exchanges use digital switching rather than mechanical or analog switching.
A Digital Signal 1 (DS1) circuit carries 24 DS0s on a North American or Japanese T-carrier (T1) line, or
32 DS0s (30 for calls plus two for framing and signaling) on an E-carrier (E1) line used in most other
countries. In modern networks, the multiplexing function is moved as close to the end user as possible,
usually into cabinets at the roadside in residential areas, or into large business premises.
These aggregated circuits are conveyed from the initial multiplexer to the exchange over a set of
equipment collectively known as the access network. The access network and inter-exchange transport
use synchronous optical transmission, for example, SONET and Synchronous Digital Hierarchy (SDH)
technologies, although some parts still use the older PDH technology.
6.5 Satellite communication
In general terms, a satellite is a smaller object that revolves around a larger object in space. For example,
moon is a natural satellite of earth.
If the communication takes place between any two earth stations through a satellite, then it is called as
satellite communication. In this communication, electromagnetic waves are used as carrier signals. These
signals carry the information such as voice, audio, video or any other data between ground and space and
vice-versa.
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Soviet Union had launched the world's first artificial satellite named, Sputnik 1 in 1957.

How a Satellite Works?

A satellite is a body that moves around another body in a particular path. A communication satellite is
nothing but a microwave repeater station in space. It is helpful in telecommunications, radio and television
along with internet applications.
A repeater is a circuit, which increases the strength of the received signal and then transmits it. But, this
repeater works as a transponder. That means, it changes the frequency band of the transmitted signal
from the received one.
The frequency with which, the signal is sent into the space is called as Uplink frequency. Similarly, the
frequency with which, the signal is sent by the transponder is called as Downlink frequency. The
following figure illustrates this concept clearly.

The transmission of signal from first earth station to satellite through a channel is called as uplink.
Similarly, the transmission of signal from satellite to second earth station through a channel is called as
downlink.
Uplink frequency is the frequency at which, the first earth station is communicating with satellite. The
satellite transponder converts this signal into another frequency and sends it down to the second earth
station. This frequency is called as Downlink frequency. In similar way, second earth station can also
communicate with the first one.
The process of satellite communication begins at an earth station. Here, an installation is designed to
transmit and receive signals from a satellite in an orbit around the earth. Earth stations send the information
to satellites in the form of high powered, high frequency (GHz range) signals.
The satellites receive and retransmit the signals back to earth where they are received by other earth
stations in the coverage area of the satellite. Satellite's footprint is the area which receives a signal of
useful strength from the satellite.
Applications of Satellite Communication
Satellite communication plays a vital role in our daily life. Following are the applications of satellite
communication −
• Radio broadcasting and voice communications
• TV broadcasting such as Direct To Home (DTH)
• Internet applications such as providing Internet connection for data transfer, GPS applications,
Internet surfing, etc.
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 • Military applications and navigations
• Remote sensing applications
• Weather condition monitoring & Forecasting

Kepler’s Law
We know that satellite revolves around the earth, which is similar to the earth revolves around the sun.
So, the principles which are applied to earth and its movement around the sun are also applicable to
satellite and its movement around the earth.
Kepler’s First Law
Kepler’s first law states that the path followed by a satellite around its primary (the earth) will be an
ellipse. This ellipse has two focal points (foci) F1 and F2 as shown in the figure below. Center of mass of
the earth will always present at one of the two foci of the ellipse.

If the distance from the center of the object to a point on its elliptical path is considered, then the farthest
point of an ellipse from the center is called as apogee and the shortest point of an ellipse from the center
is called as perigee.
Kepler’s Second Law
Kepler’s second law states that for equal intervals of time, the area covered by the satellite will be same
with respect to center of mass of the earth. This can be understood by taking a look at the following figure.

Assume, the satellite covers p1 and p2 distances in the same time interval. Then, the areas B1 and B2
covered by the satellite at those two instances are equal.
Kepler’s Third Law

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Kepler’s third law states that, the square of the periodic time of an elliptical orbit is proportional to the
cube of its semi major axis length. Mathematically, it can be written as follows –

Earth Orbit Satellites

Satellite should be properly placed in the corresponding orbit after leaving it in the space. It revolves in a
particular way and serves its purpose for scientific, military or commercial. The orbits, which are assigned
to satellites with respect to earth, are called as Earth Orbits. The satellites present in those orbits are
called as Earth Orbit Satellites.
Following are the three important types of Earth Orbit satellites −
• Geosynchronous Earth Orbit Satellites
• Medium Earth Orbit Satellites
• Low Earth Orbit Satellites
Geosynchronous Earth Orbit Satellites
A Geo-synchronous Earth Orbit (GEO) Satellite is one, which is placed at an altitude of 22,300 miles
above the Earth. This orbit is synchronized with a side real day (i.e., 23 hours 56 minutes). This orbit can
have inclination and eccentricity.
It may not be circular. This orbit can be tilted at the poles of the earth. But, it appears stationary when
observed from the Earth. These satellites are used for satellite Television.
The same geo-synchronous orbit is circular and in the plane of equator then it is called as Geo-stationary
orbit. These Satellites are placed at 35,900kms (same as Geosynchronous) above the Earth’s Equator and
they keep on rotating with respect to earth’s direction (west to east).
The satellites present in these orbits have the angular velocity same as that of earth. Hence, these satellites
are considered as stationary with respect to earth since, these are in synchronous with the Earth’s rotation
The advantage of geostationary orbit is that no need to track the antennas in order to find the position
of satellites. In geostationary earth orbit satellite the Earth station antennas to be simplified, since they
are pointed in a fixed direction and do not have to track a moving object. (For communication to the Polar
Regions of the Earth, satellites in polar orbits are used, which require Earth stations with tracking
antennas.) To prevent the satellite from tumbling, one of two spin stabilization techniques—spin
stabilization or three-axis stabilization—is used.

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Medium Earth Orbit Satellites
Medium Earth Orbit (MEO) satellites will orbit at distances of about 8000 miles from earth's surface. Signals
transmitted from a MEO satellite travel a shorter distance. Due to this, the signalstrength at the receiving end
gets improved. This shows that smaller and light weight receiving terminals can be used at the receiving end.
Transmission delay can be defined as the time it takes for a signal to travel up to a satellite and back down to
a receiving station. In this case, there is less transmission delay because the signal travels for a shorter distance
to and from the MEO satellite.
For real-time communications, the shorter the transmission delay, the better will be the communication
system. As an example, if a GEO satellite requires 0.25 seconds for a round trip, then MEO satellite requires
less than 0.1 seconds to complete the same trip. MEOs operates in the frequency range of 2 GHz and above.
These satellites are used for High speed telephone signals. Ten or more MEO satellites are required in order to
cover entire earth.
Low Earth Orbit Satellites
Low Earth Orbit (LEO) satellites are mainly classified into three categories. Those are little LEOs, big LEOs,
and Mega-LEOs. LEOs will orbit at a distance of 500 to 1000 miles above the earth's surface. These satellites
are used for satellite phones and GPS.
This relatively short distance reduces transmission delay to only 0.05 seconds. This further reduces the need
for sensitive and bulky receiving equipment. Twenty or more LEO satellites are required to cover entire earth.

Newer satellites operate in higher frequency bands, because there are so few vacant spectral assignments
in the 64-GHz band (C band). The Ku band satellites use 14 GHz on the uplink and 12 GHz on the
downlink, with an orbital spacing of 3°. Some new Ku-band satellites have high-power amplifiers that

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feed 120 to 240 W into their transmitting antenna, as compared with 20 to 40 W for low- or medium-
power satellites. High-power satellites— called direct-broadcast satellites (DBS)—provide TV service
directly to the homeowner, who has a small receiving antenna (2 ft or less in diameter). The resulting
system is called the digital satellite system (DSS) by the FCC.
Each satellite has a number of transponders (receiver-to-transmitter) aboard to amplify the received signal
from the uplink and to down-convert the signal for transmission on the downlink. (See Fig. 8–9.) Figure
8–9 shows a “bent-pipe transponder” that does not demodulate the received signal and perform signal
processing but acts as a high-power-gain down converter. Most transponders are designed for a bandwidth
of 36, 54, or 72 MHz, with 36 MHz being the standard used for C-band (64-GHz) television relay service.
As technology permits, processing transponders have come into use since an improvement in error
performance (for digital signaling) is realized.
Each satellite is assigned a synchronous orbit position and a frequency band in which to operate. In the
64-GHz band, each satellite is permitted to use a 500-MHz-wide spectral assignment, and a typical satellite
has 24 transponders aboard, with each transponder using 36 MHz of the 500-MHz bandwidth assignment.
The satellites reuse the same frequency band by having 12 transponders operating with vertically polarized
radiated signals and 12 transponders with horizontally polarized signals.† A typical 64-GHz frequency
assignment for satellites is shown in Fig. 8–10. The transponders are denoted by C1 for channel 1, C2 for
channel 2, and so on. These satellites are used mainly to relay signals for CATV systems.
6.6 Link Budget Analysis

The overall power gain (or power transfer function) of the channel is

where PTx is the signal power into the transmitting antenna, GAT is the transmitting antenna power gain,
GFS is the free-space power gain† (which is orders of magnitude less than one in typical communication
systems), GAR is the receiving antenna power gain, and PRx is the signal power into the receiver.
To use this relationship, these gains should be expressed in terms of useful antenna and free-space
parameters [Kraus, 1986]. Here, GAT and GAR are taken to be the power gains with respect to an isotropic
antenna. The EIRP is

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Where, the power density (measured in Wm2) is evaluated at the same distance, d, for both antennas. The
gains are for some practical antennas is given in the power density (Wm2) of an isotropic antenna at a
distance d from the antenna is

The FCC and others often specify the strength of an electromagnetic field by the field intensity, (V/m),
instead of power density (Wm2). The two are related by

where the power density and the field strength are evaluated at the same point in space and 377 Ω is the
free-space intrinsic impedance
If the receiving antenna is placed at d meters from the transmitting antenna, so that the received power
will be

Where λ= cf is the wavelength, c being the speed of light (3 × 108 ms) and f the operating frequency in
Hz. An antenna is a reciprocal element. That is, it has the same gain properties whether it is transmitting
or receiving.

…………………(a)
Where the free-space gain is

And LFS is the free-space path loss (absolute units). The channel gain, expressed in dB, is obtained by
taking 10 log [·] of both sides of the equation.

The free-space loss is

……….. (b)
For example, the free-space loss at 4 GHz for the shortest path to a synchronous satellite from Earth
(22,300 miles) is 195.6 dB
Note that from Eq. (a), the received power increases as the square of the wavelength (free space condition).
That is, if the carrier frequency is reduced by a factor of 2, then the received power will increase by a
factor of 4. This is equivalent to the loss decreasing by 6 dB as shown by Eq. (b).
6.7 Fiber Optical Communication

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In fiber optic communication, data is transmitted from the source to the destination by sending light pulses
through optical fibers. It changes electrical pulses to light signals and vice versa for communication. Fiber
optic communications are preferred when a huge amount of data needs to be transmitted across large
distances.

Block Diagram of Fiber Communication System

The optical fiber consists of three main elements:
1. Transmitter: An electric signal is applied to the optical transmitter. The optical transmitter consists of
driver circuit, light source and fiber flylead.
o Driver circuit drives the light source.
o Light source converts electrical signal to optical signal.
o Fiber flylead is used to connect optical signal to optical fiber.
2. Transmission channel: It consists of a cable that provides mechanical and environmental protection to
the optical fibers contained inside. Each optical fiber acts as an individual channel.
o Optical splice is used to permanently join two individual optical fibers.
o Optical connector is for temporary non-fixed joints between two individual optical fibers.
o Optical coupler or splitter provides signal to other devices.
o Repeater converts the optical signal into electrical signal using optical receiver and passes it to electronic
circuit where it is reshaped and amplified as it gets attenuated and distorted with increasing distance
because of scattering, absorption and dispersion in waveguides, and this signal is then again converted
into optical signal by the optical transmitter.
3. Receiver: Optical signal is applied to the optical receiver. It consists of photo detector, amplifier and
signal restorer.
o Photo detector converts the optical signal to electrical signal.
o Signal restorers and amplifiers are used to improve signal to noise ratio of the signal as there are chances
of noise to be introduced in the signal due to the use of photo detectors.
For short distance communication only main elements are required.

Source- LED
Fiber- Multimode step index fiber
Detector- PIN detector
For long distance communication along with the main elements there is need for couplers, beam
splitters, repeaters, optical amplifiers.

Source- LASER diode

Fiber- single mode fiber
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Detector- Avalanche photo diode (APD)

Advantages of Optical Fiber

1. The transmission bandwidth of the fiber optic cables is higher than the metal cables
2. The power loss is very low and hence helpful in long-distance transmissions.
3. Fiber optic cables provide high security and cannot be tapped.
4. Fiber optic cables are immune to electromagnetic interference.
5. These are not affected by electrical noise
6. The capacity of these cables is much higher than copper wire cables.
7. Though the capacity is higher, the size of the cable doesn’t increase like it does in copper wire cabling
system.
8. The space occupied by these cables is much less.
9. The weight of these FOC cables is much lighter than the copper ones.
10. Since these cables are di-electric, no spark hazards are present.
11. These cables are more corrosion resistant than copper cables, as they are bent easily and are flexible.
12. The raw material for the manufacture of fiber optic cables is glass, which is cheaper than copper
13. Fiber optic cables last longer than copper cables.

Applications of Fiber Optics

The optical fibers have many applications. Some of them are as follows −
• Used in telephone systems
• Used in sub-marine cable networks
• Used in data link for computer networks, CATV Systems
• Used in CCTV surveillance cameras
• Used for connecting fire, police, and other emergency services.
• Used in hospitals, schools, and traffic management systems.
• They have many industrial uses and also used for in heavy duty constructions.
Types of Fiber

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Based on the differences in the structure of the core, there are three main types of optical fibers.
1. Single-mode optical fiber
2. Multi-mode optical fiber
a. Multimode optical fiber with stepped index
b. Multimode optical fiber with graded-index

1. Single Mode Optical fiber

• As the name suggests, this type of optical fiber transmits only one mode of light. To put it another
way, it can carry only one wavelength of light across its length.
• This wavelength is usually 1310nm or 1550nm.

• These cables can carry only one mode, physically, by having a tiny core. That is to say that the
diameter of the core is essentially of the same order as the wavelength of the light passing through
it.
• Only lasers are used as a light source. To point out, the lights used in single-mode fibers are not in
the visible spectrum.
• Since the light travels in a straight direction, there are fewer losses, and it can be used in
applications requiring longer distance connections.
• A distinct disadvantage of single-mode fiber is that they are hard to couple.

2. Multimode

• As the name implies, these types of optical fibers allow multiple modes of light to travel along
their axis.
• To explain physically, they can do this by having a thicker core diameter

• The wavelengths of light waves in multimode fibers are in the visible spectrum ranging from 850
to 1300 nm.
• The reflection of the waves inside the multimode fiber occurs at different angles for every mode.
Consequently, based on these angles, the number of reflections can vary.
• Since the basis of optical fiber, communication is a total internal reflection, all modes with incident
angles that do not cause total internal reflection get absorbed by the cladding. As a result, losses
are created.
• We can have higher-order modes, waves that are highly transverse to the axis of the waveguide
can reflect many times. In fact, due to increased reflections at unusual angles, higher-order modes
can get completely lost inside the cable.

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Step index multimode fiber
In step index fiber, the refractive index suddenly changes at the interphase between core and cladding.
The refractive index of the core is slightly greater than that of cladding thus confining the light to the core
by the principle of total internal refraction,
Graded index multimode fiber
In graded index multimode fiber, the refractive index changes gradually from the core to the cladding.
These fibers collect light better than small core single mode fibers and have broader bandwidth than step
index multimode fiber.
6.8 Cellular Telephone (mobile radio) system
This cellular radio concept is illustrated in Fig. (In practice, the cell towers are usually placed at the corner
of the hexagonal cell where three cells intersect. Three 120 degree beam width directional antennas are
placed on the tower—one directional antenna covering each of the three cells. This provides coverage of
the cells using one tower, instead three towers.) Each user communicates via radio from a cellular
telephone set to the cell-site base station. This base station is connected via telephone lines or a microwave
link to the mobile switching center (MSC). The MSC connects the user to the called party. If the called
party is land based, the connection is via the central office (CO) to the terrestrial telephone network. If the
called party is mobile, the connection is made to the cellular site that covers the area in which the called
party is located, using an available radio channel in the cell associated with the called party. Theoretically,
this cellular concept allows any number of mobile users to be accommodated for a given set of radio
channels. That is, if more channels are needed, the existing cell sizes are decreased, and additional small
cells are inserted, so that the existing channels can be reused more efficiently. The critical consideration
is to design the cells for acceptable levels of co-channel interference [Lee, 1986]. As the mobile user
travels from one cell to another, the MSC automatically switches the user to an available channel in the
new cell, and the telephone conversation continues uninterrupted.
The cellular concept has the following advantages:
• Large subscriber capacity. • Efficient use of the radio spectrum. • Service to hand-held portables, as well
as vehicles. • High-quality telephone and data service to the mobile user at relatively low cost
Generation of Cellular Technology
1. First Generation
• Analog Systems
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 • Analog Modulation, mostly FM
• AMPS
• Voice Traffic
• FDMA/FDD multiple access

2. Second Generation (2G)

• Digital Systems

• Digital Modulation
• Voice Traffic
• TDMA/FDD and CDMA/FDD multiple access
• Data facility

3. Third Generation (3G)

• Voice and Data Transmission
• Simultanous voice and data access
• Multi-megabit Internet access
• Interactive web sessions
• Voice-activated calls
• Multimedia Content
• Live music

4. Fourth Generation (4G)

• Data rate of 20mbps is employed.
• Mobile speed will be up to 200km/hr.
• Frequency band is 2-8 GHz
• Features of 4G Wireless Systems
Support for interactive multimedia, voice, streaming video, Internet, and other broadband services
• IP based mobile system
• High speed, high capacity, and low cost per bit
• Global access, service portability, and scalable mobile services
• Ad hoc and multi hop networks
• Better spectral efficiency

Frequency Reuse
• Each cellular base station is allocated a group of radio channels within a small geographic area called a
cell.
• Neighboring cells are assigned different channel groups.
• By limiting the coverage area to within the boundary of the cell, the channel groups may be reused to
cover different cells.
• Keep interference levels within tolerable limits.

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Cell Splitting
• Split congested cell into smaller cells.
– Preserve frequency reuse plan.
– Reduce transmission power.

GSM Architecture
Mobile stations (MS), Base Transceiver Station (BTS) and the Base Station Controller (BSC)
The mobile station contains IMEI (International mobile equipment identity)
The IMSI (International mobile subscriber identity) is stored in the subscriber identity module (SIM),
the HLR, VLR database

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 The IMSI is a unique identity which is used internationally and used within the network to identify the
mobile subscribers.
Provides and manages radio transmission paths between the MS and MSC.
One BSC controls up to several hundred BTSs.
BSC performs handover for MS under the control of same BSC.
MSCs, Visitor Location Register (VLR), Home Location Register (HLR), Authentication Center (AUC)
and Equipment Identity Register (EIR).
Switching of GSM calls between external networks and the BSCs.
HLR : contains subscriber information (International Mobile Subscriber Identity -IMSI) and location
information for each user who resides in the same city as the MSC.
VLR : temporarily stores the IMSI and customer information for each roaming subscriber who is
visiting the coverage area of a particular MSC.

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MULTIPLEXING

Wavelength Division Multiplexing

Wavelength Division multiplexing (WDM) is an analog technique, in which many data streams of
different wavelengths are transmitted in the light spectrum. If the wavelength increases, the frequency of
the signal decreases. A prism which can turn different wavelengths into a single line can be used at the
output of MUX and input of DEMUX.
Example − Optical fiber Communications use the WDM technique, to merge different wavelengths into
a single light for the communication.
Code division multiplexing
Code division multiplexing (CDM), also called spread-spectrum technique, is a multiplexing technique
which has been widely implemented in third generation of wireless network. It takes full advantage of the
available spectrum. Through several concurrent transmissions over the spectrum, this technique has
enhanced the capacity of network 18 times compared to first generation and 6 times compared to the
second generation of wireless communication technologies. For transmitting multiple messages over the
channel simultaneously, the multiplexer assigns a separate spreading code from a set of orthogonal
pseudo-random sequences to each user. The orthogonality of these sequences will help users to recover
their desired signals from the multiplexed signals.
Salient Features of CDMA
CDMA, which is based on the spread spectrum techniques has following salient features −
In CDMA, every channel uses the full available spectrum.

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 Individual conversations are encoded with a pseudo-random digital sequence and then transmitted using
a wide frequency range.

CDMA consistently provides better capacity for voice and data communications, allowing more
subscribers to connect at any given time.

CDMA is the common platform on which 3G technologies are built. For 3G, CDMA uses 1x EV-DO
and EV-DV.

Orthogonal Frequency Division Multiplexing

In data communications and networking, orthogonal frequency-division multiplexing (OFDM) is a
method of digital data modulation, whereby a single stream of data is divided into several separate sub-
streams for transmission via multiple channels.
OFDM uses the principle of frequency division multiplexing (FDM), where the available bandwidth is
divided into a set of sub-streams having separate frequency bands. OFDM was introduced in 1966 by
Chang at Bell Labs and was improved by Weinstein and Ebert in 1971.
Working Principle of OFDM: OFDM is a specialized FDM having the constraint that the sub-streams
in which the main signal is divided, are orthogonal to each other. Orthogonal signals are signals that are
perpendicular to each other. A main property of orthogonal signals is that they do not interfere with each
other.
When any signal is modulated by the sender, its sidebands spread out either side. A receiver can
successfully demod0ulate the data only if it receives the whole signal. In case of FDM, guard bands are
inserted so that interference between the signals, resulting in cross-talks, does not occur. However, since
orthogonal signals are used in OFDM, no interference occurs between the signals even if their sidebands
overlap. So, guard bands can be removed, thus saving bandwidth. The criteria that need to be maintained
is that the carrier spacing should be equal to the reciprocal of the symbol period.
In order that OFDM works, there should be very accurate synchronization between the communicating
nodes. If frequency deviation occurs in the sub-streams, they will not be orthogonal any more, due to
which interference between the signals will occur.
The following diagram plots FDM versus OFDM, to depict the saving in bandwidth obtained by OFDM
–

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Usages
OFDM is used in the following area −
Wi-Fi
DSL internet access
4G wireless communications
digital television
radio broadcast services

6.9 Television
Television (TV) is a method of reproducing fixed or moving visual images by the use of electronic signals.
Television means to see at a distance. For Transmission of pictures, the visual information in the scene is
converted into corresponding electrical signal which modulates an RF carrier for transmission. In the
receiver, the original signal is detected and is made to form the image of the original picture on the
fluorescent screen. In monochrome receivers, the reduced picture has black, white and various shades of
grey. In color television, picture is reproduced in all its natural colors formed by combination of red green
and blue.
The Block schematic arrangement of a television transmission and reception systems are shown.

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