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Higher Technological Institute

Electrical Engineering Department Course Instructor:

Lectures series in
Dr.Eng. Ehab Helmy Mohamed Elshazly.
Assistant Professor at Egyptian Atomic Energy Authority
Communications Engineering 4
B.Sc. Electronic Engineering, Menoufya University.
Principles of Digital Communication M.Sc. Communication Engineering, Menoufya University.
Ph.D. Communication Engineering, E-JUST.
By
Dr. Eng. Ehab Helmy El Shazly Research area: Computer vision & Image processing.

Mobile: 01009448624
Email: ehab.elshazly@ejust.edu.eg
ihab.helmy311@hti.edu.eg

Course Outlines Course Outlines

Text Book:  Basic Digital Communication system.
“Modern Digital And Analog Communication Systems”  Basic binary modulation, ASK, FSK, PSK.
By B. P Lathi, 3rd Edition.  Differential Phase Shift Keying (PSK): DPSK,
 MSK Multi-level modulation and signal constellations, QPSK, QAM-
Reference books: principles.
1. Communication Systems 4th ed.  Modulation schemes evaluation.
By Simon Haykin.  Multiplexing, multiple access, duplexing, time division, frequency division.
&  Principles of cellular mobile communication systems
2. Analog and Digital Communication  Cellular network organization, cell size, frequency reuse, operation of
By Simon Haykin. cellular systems.
 Mobile radio propagation effects, path loss calculations
Evaluation:  Satellite communication systems.
Assignments+ Tests + Quizzes 60%  Line coding schemes.
Final Term Exam 40%  Digital data transmission techniques

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Lecture Outlines Communications Systems

• Communication means to share information; To

 Overview of general Communication System transmit or receive information or data.
 Digital vs. Analog Communications
 Analog to Digital conversion.
 Why Digital Communication systems is Different ? • Communication may be established between people,
 Design goals of any communication system. Computer to Computer, near distance far distance
 System capacity. (Telecommunication)

• The communication system , consist of different

components which helps in transmitting or receiving
data, between two entities or devices.

Signal: The actual entity (electrical, optical, mechanical, etc.) that is

transmitted from sender to receiver and convey information.
Components of communication system
Message: Information that is transmitted.

Information: Knowledge communicated or received concerning some

fact or circumstance.

The source: Originates a message, which could be a human voice, a

television picture or data. The source is converted by an input
transducer into an electrical waveform referred to as the baseband
signal or message signal.

The transmitter: Modifies the baseband signal for efficient

transmission. The transmitter generally consists of one or more of the
following subsystems: a pre-emphasizer, a sampler, a quantizer, a coder
and a modulator.

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Pre-emphasis: Refers to a system process designed to

increase (within a frequency band) the magnitude of some
Receiver converts the received signal back into a readable message, The
(usually higher) frequencies with respect to the magnitude of
receiver output is fed to the output transducer, which converts the
other (usually lower) frequencies in order to improve the electrical signal to its original form.
overall signal-to-noise ratio
The receiver output is fed to the output transducer, which converts the
The channel: Medium through which the transmitter’s electrical signal to its original form.
output is sent, which could be a wire, a coaxial cable, an
A channel acts as a filter that attenuate the signal and distort its
optical fiber, or a radio link, etc.
waveform.
The signal attenuation increases with the length of the channel.
Based on the channel type, modern communication systems
are divided into two categories: wired communication
systems and wireless communication systems.

Analog versus Digital

Noise: random and unpredictable modifications of the signal amplitude
Digital messages are constructed with a finite number of symbols. For
example,
 Thermal noise or Internal noise: caused by random movements of
printed language consists of 26 letters, 10 numbers, a space, and several
electrons in conductors punctuation marks. Thus, a text is a digital message constructed from
about 50 symbols.
 External noise: caused e.g. by interference with other channels.
Human speech is also a digital message, because it is made up from a
A high signal-to-noise ratio (SNR) is required for a high quality finite vocabulary in a language.
communication Similarly, a Morse-coded telegraph message is a digital message
SNR = power of signal / power of noise. constructed from a set of only two symbols mark and space. It is
therefore a binary message, implying only two symbols.
However in a realistic channel the noise is accumulating along the A digital message constructed with M symbols is called an M-ary
channel's path and the signal amplitude is decreasing. message.

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Analog versus Digital (cont…) Comparison of analog and digital signals

Analog messages, on the other hand, are characterized by data whose

values vary over a continuous range.

For example, the temperature or the atmospheric pressure of a certain

location can vary over a continuous range and can assume an infinite
number of possible values.

Similarly, a speech waveform has amplitudes that vary over a

continuous range.

Over a given time interval, an infinite number of possible different

speech wave forms exist, in contrast to only a finite number of possible
digital messages.

Analog Signals
• An analog signal is continuous and it can have an infinite
number of values in a range. The primary short comings of
analog signals is the difficulty to separate noise from the original Note
waveform.
Phase describes the position of the
• An example is a sine wave which can be specified by
three characteristics: θ(t) = A sin (2 π f t + Φ)
waveform relative to time 0.
A: amplitude f : frequency Φ : phase

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Figure 3.5 Three sine waves with the same amplitude and frequency,
but different phases

Example 3.3

A sine wave is offset 1/6 cycle with respect to time 0.

Table 3.1 Units of period and frequency What is its phase in degrees and radians?

Solution
We know that 1 complete cycle is 360°. Therefore, 1/6
cycle is

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Example 3.8
Figure 3.15 The bandwidth for Example 3.8

A non periodic composite signal has a bandwidth of 200

kHz, with a middle frequency of 140 kHz and peak
amplitude of 20 V. The two extreme frequencies have an
amplitude of 0. Draw the frequency domain of the signal.

Solution
The lowest frequency must be at 40 kHz and the highest at
240 kHz. Figure 3.15 shows the frequency domain and the
bandwidth.

Why digital? Why Digital Communications?

Digital techniques need to distinguish between discrete
symbols allowing regeneration versus amplification Can combine different signal types – data, voice, text, etc.

Data communication in computers is digital in nature

Good processing techniques are available for digital whereas voice communication between people is analog in
signals, such as: nature. The two types of communication are difficult to
combine over the same medium in the analog domain.
Using digital techniques, it is possible to combine both format
Data compression (or source coding) for transmission through a common medium.
Error Correction (or channel coding)(A/D conversion)
Encryption and privacy techniques are easier to implement.
Equalization
Security Better overall performance.

Easy to mix signals and data using digital techniques

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Why Digital Communications? Why Digital Communications?

Hardware is more flexible. Digital hardware implementation is Easy to regenerate the distorted signal. Regenerative
flexible and permits the use of microprocessors, mini-processors, repeaters along the transmission path can detect a digital
digital switching and VLSI. signal and retransmit a new, clean (noise free) signal
Shorter design and production cycle. These repeaters prevent accumulation of noise along the
path . This is not possible with analog communication
Low cost , the use of LSI and VLSI in the design of components systems. Why?
and systems have resulted in lower cost .

Easier and more efficient to multiplex several digital signals. Digital

multiplexing techniques – Time & Code Division Multiple Access -
are easier to implement than analog techniques such as Frequency
Division Multiple Access.

(a) transmitted signal

(b) received distorted
signal without noise.
(c) received distorted
signal with noise.
(d) regenerated signal
The data can be recovered correctly as long as the distortion and the
noise are within limits. In contrast, the waveform in an analog message is
important, and even a slight distortion or interference in the waveform
will cause an error in the received signal.
Clearly, a digital communication system is more rugged than an analog
communication system in the sense that it can better withstand noise and
distortion (as long as they are within a limit).

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The main reason for the superiority of digital systems over analog ones is For analog systems, there is no way to avoid accumulation of noise and
the viability of regenerative repeaters in the former. distortion along the path. As a result, the distortion and the noise
interference are accumulated over the entire transmission path.
Repeater stations are placed along the communication path of a digital Thus, with increasing distance the signal becomes weaker, whereas the
system at distances short enough to ensure that noise and distortion distortion and the noise become stronger.
remain within a limit. This allows pulse detection with high accuracy.
Amplification is of little help, because it enhances the signal and the
At each repeater station the incoming pulses are detected and new clean noise in the same proportion. Consequently, the distance over
pulses are transmitted to the next repeater station. which an analog message can be transmitted is limited by the transmitter
power.
This process prevents the accumulation of noise and distortion along the Despite these problems, analog communication was used widely and
path by cleaning the pulses periodically at the repeater stations. We can successfully in the past.
thus transmit messages over longer distances with greater accuracy. Because of the advent of optical fiber and the dramatic cost reduction
achieved in the fabrication of digital circuitry, almost all new
communication systems being installed are digital.

TRANSMISSION IMPAIRMENT

Signals travel through transmission media, which are not perfect. The imperfection
causes signal impairment.

This means that the signal at the beginning of the medium is not the same as the
signal at the end of the medium.

What is sent is not what is received. Three causes of impairment are attenuation,
distortion, and noise.

Topics discussed in this section:

 Attenuation
 Distortion
 Noise

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Distortion Attenuation

Noise

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RSS is Received Signal Strength

Suppose a signal travels through a transmission medium and its A signal travels through an amplifier, and its power is increased 10
power is reduced to one-half. This means that P2 is (1/2)P1. In this times. This means that P2 = 10P1 . In this case, the amplification
case, the attenuation (loss of power) can be calculated as (gain of power) can be calculated as

A loss of 3 dB (–3 dB) is equivalent to losing one-half the power.

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The loss in a cable is usually defined in decibels per kilometer

Sometimes the decibel is used to measure signal power in (dB/km). If the signal at the beginning of a cable with −0.3 dB/km
milliwatts. In this case, it is referred to as dBm and is calculated as has a power of 2 mW, what is the power of the signal at 5 km?
dBm = 10 log10 Pm , where Pm is the power in milliwatts. Calculate Solution
the power of a signal with dBm = −30. The loss in the cable in decibels is 5 × (−0.3) = −1.5 dB. We can
calculate the power as
Solution
We can calculate the power in the signal as

Goals in Communication System

Digital Communications Disadvantages
Design
• To maximize transmission rate, R
• Requires reliable “synchronization” • To maximize system utilization, U
• To minimize bit error rate, Pe
• Requires A/D conversions at high rate • To minimize required systems bandwidth, W
•
• Requires larger bandwidth To minimize system complexity, Cx
• To minimize required power, Eb/No

• Performance Criteria
• Probability of error or Bit Error Rate

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Interface of Analog and Digital Systems: A/D and D/A Conversion „

Digital Communication System Shannon's Sampling Theorem, which states that if the highest frequency in the signal
spectrum is B (in hertz), then the signal can be recovered from its samples, taken at a
rate not less than 2B samples per second. „

Quantization each sample is approximated, or round off to the nearest quantized level,
the information is thus digitalized. The quantized signal is an approximation of the
original one. We can improve the accuracy of the quantized signal to any desired degree
by increasing the number of levels. „

Coding „
adc (analog/digital converter): converts the physical analog signal  Source coding Convert the quantized signal into binary sequences. „
into a digital electronic signal  Channel coding Introduce redundancy in a controlled manner to overcome the
source encoder: encodes the data in a format that removes effects of noise and interferences. „
redundancy and irrelevant information.
Mapping Map binary sequence into symbols. „
channel encoder: encodes the signal pulses in a format that is
required by the channel Transmission Symbols are applied to a transmitter filter, which produces a continuous
protocol: controls start, end of transmission and error recovery by signal for transmission over a continuous channel.
adding additional bits for error detection and/or correction

PCM Block Diagram

PCM consists of three steps to digitize an analog signal:

1. Sampling.
2. Quantization.
3. Binary encoding.

Before we sample, we have to filter the signal to limit the

maximum frequency of the signal as it affects the sampling rate.

Filtering should ensure that we do not distort the signal, i.e.

remove high frequency components that affect the signal shape.

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Three different sampling methods

Sampling
for PCM
• Analog signal is sampled every TS sec.
• Ts is referred to as the sampling interval.
• fs = 1/Ts is called the sampling rate or sampling frequency.
There are 3 sampling methods:
1. Ideal - an impulse at each sampling instant.

2. Natural - a pulse of short width with varying amplitude.

3. Flattop - sample and hold, like natural but with single

amplitude value.

The process is referred to as pulse amplitude modulation PAM and

the outcome is a signal with analog (non integer) values.

According to the Nyquist theorem, the sampling rate must be at least 2 Example 2.1
times the highest frequency contained in the signal.

For an intuitive example of the Nyquist theorem, let

us sample a simple sine wave at three sampling rates:
 fs = 4f (2 times the Nyquist rate),
 fs = 2f (Nyquist rate), and
 fs = f (one-half the Nyquist rate).

Figure 4.24 shows the sampling and the subsequent

recovery of the signal.

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Example 2.2
Figure 4.24 Recovery of a sampled sine wave for different sampling rates
It can be seen that
sampling at the Nyquist A complex low-pass signal has a bandwidth of 200 kHz.
rate can create a good
approximation of the
What is the minimum sampling rate for this signal?
original sine wave (part
a).
Oversampling in part b
can also create the same Solution
approximation,
but it is redundant and
The bandwidth of a low-pass signal is between 0 and f,
unnecessary. where f is the maximum frequency in the signal. Therefore,
Sampling below the we can sample this signal at 2 times the highest frequency
Nyquist rate (part c) (200 kHz). The sampling rate is therefore 400,000 samples
does not produce a
signal that looks like the per second.
original sine wave.

Example 2.3
Quantization
A complex bandpass signal has a bandwidth of 200 kHz. Sampling results in a series of pulses of varying amplitude
What is the minimum sampling rate for this signal? values ranging between two limits: a min and a max.
The amplitude values are infinite between the two limits.
We need to map the infinite amplitude values onto a finite set of
Solution known values.
We cannot find the minimum sampling rate in this case This is achieved by dividing the distance between min and max
because we do not know where the bandwidth starts or into L zones, each of height Δ = (max - min)/L
ends. We do not know the maximum frequency in the
The midpoint of each zone is assigned a value from 0 to L-1
signal. (resulting in L values)
Each sample falling in a zone is then approximated to the value
of the midpoint.

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Quantization Zones Assigning Codes to Zones

Each zone is then assigned a binary code.

Assume we have a voltage signal with amplitudes Vmin=
-20V and Vmax=+20V. The number of bits required to encode the zones, or the number of bits
per sample as it is commonly referred to, is obtained as follows:
We want to use L=8 quantization levels. nb = log2 L.
Zone width Δ = (20 - -20)/8 = 5 Given our example, nb = 3
The 8 zones are: -20 to -15, -15 to -10,-10 to -5, -5 to 0, 0 The 8 zone (or level) codes are therefore: 000, 001, 010, 011, 100,
to +5, +5 to +10,+10 to +15, +15 to +20
101, 110, and 111
The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5,
17.5 Assigning codes to zones:
000 will refer to zone -20 to -15
001 to zone -15 to -10, etc.

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