Digital communication

Objectives:
1. To understand the building blocks of digital communication system.
2. To understand and analyze the signal flow in a digital communication system.
3. To understand basic digital modulation schemes
4. To analyze error performance of a digital communication system in presence of noise and
other interferences.

Chapter 1: Introduction to digital communication

I. Analog to digital
The conventional methods of communication used analog signals for long distance
communications, which suffer from many losses such as distortion, interference, and other losses
including security breach. In order to overcome these problems, the signals are digitized using
different techniques. The digitized signals allow the communication to be more clear and accurate
without losses. The following figure indicates the difference between analog and digital signals.
The digital signals consist of 1s and 0s which indicate High and Low values respectively.

1.1. Advantages of digital communication

As the signals are digitized, there are many advantages of digital communication over
analog communication, such as:

• The effect of distortion, noise, and interference is much less in digital signals as
they are less affected.
• Digital circuits are more reliable.
• 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.

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 • 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 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.

1.2. Elements of digital communication

The elements which form a digital communication system is represented by the following
block diagram:

1.2.1. Source
The source can be an analog signal like sound for example
1.2.2. Input transducer
This is a transducer which takes a physical input and converts it to an
electrical signal (Example: microphone). This block also consists of an analog
to digital converter where a digital signal is needed for further processes. A
digital signal is generally represented by a binary sequence.

1.2.3. 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)

1.2.4. Channel encoder

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 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.

1.2.5. Digital modulator

The signal to be transmitted is modulated here by a carrier. The signal is also
converted to analog from the digital sequence, in order to make it travel through
the channel or medium.

1.2.6. Channel
The channel or a medium, allows the analog signal to transmit from the
transmitter end to the receiver end.

1.2.7. 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.

1.2.8. 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.
1.2.9. 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.

1.2.10. 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 (Example: loud speaker).
1.2.11. Output signal
This is the output which is produced after the whole process. Example
− The sound signal received. This unit has dealt with the introduction, the
digitization of signals, the advantages and the elements of digital
communications.

II. Pulse code modulation

Modulation is the process of varying one or more parameters of a carrier signal in
accordance with the instantaneous values of the message signal. The message signal is the signal

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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.
There are many modulation techniques, which are classified according to the type of
modulation employed. Of them all, the digital modulation technique used is Pulse Code
Modulation (PCM). 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
following figure shows an example of PCM output with respect to instantaneous values of a
given sine wave.

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this
process is called as digital. Each one of these digits, though in binary code, represent the
approximate amplitude of the signal sample at that instant.
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.

2.1. Basic elements of PCM

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.

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 2.1.1. 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.
2.1.2. 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. 2.1.3. 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.
2.1.4. 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. 2.1.5. 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.
2.1.6. Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original signal.
This circuit acts as the demodulator.

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 2.1.7. 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.

2.2. 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 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. The following figure 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.

2.2.1. Sampling rate

To discretize the signals, the gap between the samples should be fixed. That gap can be
termed as a sampling period Ts.

SamplingFrequency fs
Where,

Ts is the sampling time and fs is the sampling frequency or the sampling rate

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 Sampling frequency is the reciprocal of the sampling period. 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.

2.2.2. Nyquist rate

Suppose that a signal is band-limited with no frequency components higher than W Hertz.
That 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.
Which means, fS=2W Where fS is the sampling rate and W is the highest frequency. This
rate of sampling is called as Nyquist rate. A theorem called, Sampling Theorem, was stated on
the theory of this Nyquist rate.

2.2.3. The 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.”
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
For the continuous-time signal x (t), the band-limited signal in frequency domain, can be
represented as shown in the following figure.

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 We need a sampling frequency, a frequency at which there should be no loss of
information, even 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(t) is:
Xs(w)=1Ts∑n=−∞∞X(w−nw0)
Where Ts = Sampling Period and w0=2πTs
Let us see what happens if the sampling rate is equal to twice the highest frequency that
means, fs=2W Where, fs is the sampling frequency and W is the highest frequency

The result will be as shown in the above figure. 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 following
figure.

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 We can observe from the above pattern that the over-lapping of information is done,
which leads to mixing up and loss of information. This unwanted phenomenon of over-lapping is
called as 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.”
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.

2.3. 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.

2.3.1. Quantizing an analog signal

The analog-to-digital converters perform this type of function to create a series of digital
values out of the given analog signal. The following figure represents an analog signal. This
signal to get converted into digital, has to undergo sampling and quantizing.

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 The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels. Quantization is representing the sampled values of the amplitude by a finite
set of levels, which means converting a continuous-amplitude sample into a discrete-time signal.
The following figure shows how an analog signal gets quantized. The blue line represents analog
signal while the brown one represents the quantized signal.

Both sampling and quantization result in the loss of information. 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.

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 The following figure shows the resultant quantized signal which is the digital form for the
given analog signal.

This is also called as Stair-case waveform, in accordance with its shape.

2.3.2. 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.

There are two types of uniform quantization. They are Mid-Rise type and MidTread type.
The following figures represent the two types of uniform quantization.

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of uniform quantization.

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 • The Mid-Rise type is so called because the origin lies in the middle of a raising part of
the stair-case like graph. The quantization levels in this type are even in number.
• The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
• Both the mid-rise and mid-tread type of uniform quantizers are symmetric about the
origin.

2.3.3. Quantization error

For any system, during its functioning, there is always a difference in the values of its
input and output. The processing of the system results in an error, which is the difference of those
values.

The difference between an input value and its quantized value is called a Quantization
Error. A Quantizer is a logarithmic function that performs Quantization (rounding off the value).
An analog-to-digital converter (ADC) works as a quantizer.

Quantization noise is a type of quantization error, which usually occurs in analog audio
signal, while quantizing it to digital. For example, in music, the signals keep changing
continuously, where a regularity is not found in errors. Such errors create a wideband noise.

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