LESSON 2 DIGITAL DATA COMMUNICATION NETWORK

Enabling Objectives
By using the information and answering the worksheet in this lesson, you should be able to demonstrate
ability to:
Know and understand the building blocks of digital communication system.
Understand the need for digitizing analog signals.
Know and understand the elements of pulse code modulation.
Explain the role of sampling in the process of transforming of analog to digital signal.
Comprehend Nyquist theorem and know its importance in determining sampling frequency and rate.
Understand the implication of ‘aliasing’ in the quality of sampled signal and in the design of low pass
filter.

1.0 Introduction
The necessity for digitization is to overcome problems that were encountered with the conventional
methods of communication using analog signals. Analog signals suffer from many losses caused by
distortion, interference, and security breach, among others. With digitized signals, communicating
devices allow communication to be more clear and accurate with tolerable losses.
This lesson covers the processes for digitization of signals, the advantages and the elements of digital
communication system, pulse code modulation of signals and Nyquist Theorem.

1.1 Digital Communication Network

To better understand why digital signal is superior to its predecessor analog signal, Fig. 2.0 shows a
comparison of both types of signal. Fig. 2.0a represents the analog signal which appears continuous.
A common example of analog signal is the human voice picked-up by a microphone. The same signal
can be converted into a digital signal (Fig. 2.0b) which consists of 1s and 0s or indicates High and Low
values respectively. The 0s and 1s are aggregated, for example 8 combinations of 0s and 1s, to form
a byte. A byte (consists of bits or 0s and 1s) which represents a character (e.g. letters of the alphabet
a –z, A-Z, numbers and other special characters). Once signal is expressed in binary digits, that defines
digital signal’s unique advantages over the analog signals.

(a) (b)
Figure 2.0: Representation of (a) analog signal, and (b) digital signal.

1.2 Advantages of Digital Communication

There are several advantages of digital communication when signals are digitized over its analog
communication counterpart. Among them are:
 The effect of distortion, noise, and interference is much less in digital signals as they are less
affected.
 Digital circuits are more reliable.

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  Digital circuits are easy to design and cheaper than analog circuits.
 The hardware implementation in digital circuits, is more flexible than analog.
 The occurrence of cross-talk is very rare in digital communication.
 The signal is un-altered as the pulse needs a high disturbance to alter its properties, which is very
difficult.
 Signal processing functions such as encryption and compression are employed in digital circuits
to maintain the secrecy of the information.
 The probability of error occurrence is reduced by employing error detecting and error correcting
codes.
 Spread spectrum technique is used to avoid signal jamming.
 Combining digital signals using Time Division Multiplexing (TDM) is easier than combining analog
signals using Frequency Division Multiplexing (FDM).
 The configuring process of digital signals is easier than in analog signals.
 Digital signals can be saved and retrieved more conveniently than analog signals.
 Many of the digital circuits have almost common encoding techniques and hence similar devices
can be used for a number of purposes.
 The capacity of the channel is effectively utilized by digital signals.

2.3 Elements of Digital Communication System

Figure 2.1 depicts the key elements of a digital communication system. It is presented as a block
diagram to facilitate understanding.

Figure 2.1: Key elements of a digital communication system.

1. Source. The source can be an analog signal. Example of this type of signal is a sound signal.
2. Input Transducer. This is a transducer which takes a physical input (e.g. voice feed to microphone)
and converts it to an electrical signal (see Fig. 2.0a). This block also consists of an analog to
digital converter where a digital signal is needed for further processes. As already mentioned, a
digital signal is generally represented by a binary sequence (0s and 1s).

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 3. Source Encoder. The source encoder compresses the data into minimum number of bits. This
process helps in effective utilization of the bandwidth. It removes the redundant
bits (unnecessary excess bits), i.e., zeroes.
4. Channel Encoder. The channel encoder, does the coding for error correction. During the
transmission of the signal, due to the noise in the channel, the signal may get altered and hence
to avoid this, the channel encoder adds some redundant bits to the transmitted data. These are
the error correcting bits.
5. Digital Modulator. The signal to be transmitted is modulated (modified) here by a carrier. A
carrier signal, also called carrier frequency, simply carry the signal. The signal is also converted to
analog from the digital sequence, in order to make it travel through the channel or medium such
as coaxial cable.
6. Channel. The channel or a medium, allows the analog signal to transmit from the transmitter end
to the receiver end.
7. Digital Demodulator. This is the first step at the receiver end. The received signal is demodulated
as well as converted again from analog to digital. The signal gets reconstructed here.
8. Channel Decoder. The channel decoder, after detecting the sequence, does some error
corrections. The distortions which might occur during the transmission, are corrected by adding
some redundant bits. This addition of bits helps in the complete recovery of the original signal.
9. Source Decoder. The resultant signal is once again digitized by sampling and quantizing so that
the pure digital output is obtained without the loss of information. The source decoder recreates
the source output.
10. Output Transducer. This is the last block which converts the signal into the original physical form,
which was at the input of the transmitter. It converts the electrical signal into physical output (for
example in a loud speaker).
11. Output Signal. This is the output which is produced after the whole process. For example, the
sound signal received.

To illustrate how a source signal goes through the whole process, let us use a human voice picked-
up from a laptop or a cellphone microphone ‘say’ during a synchronous (or on-line) class discussion.
1. Source. When a teacher talks in front of the microphone, his voice is the source analog signal. It
will be fed into the input transducer.
2. Input Transducer. The signal from the source (teacher’s voice) will be converted into an electrical
signal in the form of a sinewave. The teacher’s voice will be converted from analog to digital
signal using the analog to digital circuit (A to D). The output of this block is now in the form of
digital signal and it is fed to the source encoder.
Note: You will continue this in your learning activity worksheet to complete the whole process.

2.4 Pulse Code Modulation (PCM)

Modulation is the process of varying one or more parameters of a carrier signal in accordance with
the instantaneous values (sinewave) of the message signal. The message signal is the signal which is
being transmitted for communication and the carrier signal is a high frequency signal which has no
data, but is used for long distance transmission.

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There are many modulation techniques, which are classified according to the type of modulation
employed. For digital modulation, Pulse Code Modulation (PCM) technique is used. A signal is pulse
code modulated to convert its analog information into a binary sequence, i.e., 1s and 0s. The output
of a PCM will resemble a binary sequence. The figure below shows an example of PCM output with
respect to instantaneous values of a given sine wave (often the source signal).

Instead of a pulse train (represent by sinewave), PCM produces a series of numbers or digits,
represented by the square wave (see PCM output), and hence this process is called as digital. Each
one of these digits, though in binary code or number, represent the approximate amplitude (values
on the sinewave) of the signal sample at that instant. Recall that sinewave values are instantaneous.
In Pulse Code Modulation, the message signal is represented by a sequence of coded pulses. This
message signal is achieved by representing the signal in discrete form in both time and amplitude.

Basic Elements of PCM

The transmitter section of a PCM circuit consists of Sampling, Quantizing and Encoding, which are
performed in the analog-to-digital (A to D) 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 (i.e. weakened) signals,
decoding, and reconstruction of the quantized pulse train. Figure 2.2 is the block diagram of PCM
which represents the basic elements of both the transmitter and the receiver sections.
1. 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.
2. 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.
3. 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.
4. Encoder. The digitization of analog signal is done by the encoder. It designates each quantized
level by a binary code. The sampling done in this block 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.

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 5. 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.
6. Decoder. The decoder circuit decodes the pulse coded waveform to reproduce the original signal.
This circuit acts as the demodulator.

Figure 2.2: Key elements of a PCM.

7. Reconstruction Filter. After the digital-to-analog conversion is done by the regenerative circuit
and the decoder, a low-pass 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.

2.5 Digital Communication – Sampling

Sampling is defined as, “The process of measuring the instantaneous values of continuous-time signal
in a discrete form.” Sample is a piece of data taken from the whole data which is continuous (e.g. in
sinewave) in the time domain.
When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e., High or
Low, the signal has to be discretized in time. This discretization of analog signal is called as Sampling.
Fig. 2.3 indicates a continuous-time signal x(t) and a sampled signal xs(t). When x (t) is multiplied by
a periodic impulse train, the sampled signal xs (t) is obtained.
As noted in Fig. 2.3, the sampled signals, represented by the vertical lines, are now discretized in
time.

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 Figure 2.3: Representation of sampled signal.

Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed as
a sampling period TS.
Sampling Frequency (fs) =1/TS= fs
Where:
Ts is the sampling time
fs is the sampling frequency or the sampling rate
Sampling frequency is the reciprocal of the sampling period (Ts=1/fs). This sampling frequency, can
be simply called as Sampling rate. The sampling rate denotes the number of samples taken per
second, or for a finite set of values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be highly
considered. The rate of sampling should be such that the data in the message signal should neither
be lost nor it should get over-lapped. Hence, a rate was fixed for this, called as Nyquist rate.
Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher than W Hertz. This
means, W is the highest frequency. For such a signal, for effective reproduction of the original signal,
the sampling rate should be twice the highest frequency. This gives us
fS=2W
Where:
fs is the sampling rate
W is the highest frequency
2W is the rate of sample for band-limited signal
This rate of sampling is called as Nyquist rate.

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A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate. The Nyquist
Theorem states that in order to adequately reproduce a signal it should be periodically sampled at
a rate that is 2X the highest frequency (fs=2W) you wish to record.

This gives a Nyquist sampling formula:

Nyquist Sampling (f) = W/2, where W=the smallest object, or highest frequency we wish to
record.
Illustrative Example
An audio CDs have a sampling rate of 44100 samples/sec. What is the Nyquist frequency?

Solution:
Nyquist Sampling Frequency (fs): fs = W/2
fs = 44100/2
fs = 22050 Hz or 22.050 Khz
The anti-aliasing filter (low-pass filter) must adequately suppress any higher frequencies but
negligibly affect the frequencies within the human hearing range. Theoretically, a filter that preserves
0–20 kHz is more than adequate. The above result suggests that the low-pass filter design can
preserve 0-22.1 KHz.
Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient
sample rate in terms of bandwidth for the class of functions that are bandlimited.

The sampling theorem states that, “a signal can be exactly reproduced if it is sampled at the
rate fs which is greater than twice the maximum frequency W.” In other words, if the sample
frequency is below it, the signal cannot be recovered.
To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal whose value
is non-zero between some –W and W Hertz.

Such a signal is represented as x(f)=0 for ∣f∣ >W. The |f| means absolute value, which
means that the sign of W is disregarded.
For the continuous-time signal x(t), the band-limited signal in frequency domain, can be represented
as shown in the figure below with x(w) is the highest frequency.

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We need a sampling frequency, a frequency at which there should be no loss of information, eve after
sampling. For this, we have the Nyquist rate that the sampling frequency should be two times the
maximum frequency. It is the critical rate of sampling.
If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered, and if it is
sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency domain.

The above figure shows the Fourier transform of a signal xs t. Here, the information is reproduced
without any loss. There is no mixing up and hence recovery is possible.
The Fourier Transform of the signal xs ts

Let us see what happens if the sampling rate is equal to twice the highest frequency (2W)
That means, fs=2W

Where,
fs is the sampling frequency
W is the highest frequency
The result is shown in the figure to the right.
The information is replaced without any loss.
Hence, this is also a good sampling rate.

Now, let us look at the condition, fs < 2W

The resultant pattern will look like the figure
on the right. We can notice pattern of over-
lapping of information (the point where wave
intersects), which leads to mixing up and loss
of information. This unwanted phenomenon
of over-lapping is called as Aliasing.
Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a
signal, taking on the identity of a low-frequency component in the spectrum of its sampled version.”

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The corrective measures taken to reduce the effect of Aliasing are:
 In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the sampler,
to eliminate the high frequency components, which are unwanted.

 The signal which is sampled after filtering, is sampled at a rate slightly higher than the Nyquist rate.

This choice of having the sampling rate higher than Nyquist rate, also helps in the easier design of
the reconstruction filter at the receiver.

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