Chapter4 OK

Chapter 4: Digital Modulation 83
Chapter 4
Digital Modulation
Chapter Outline
4.1 Introduction
4.2 Digital Communication System
4.3 Digital Data, Digital Signal
4.3.1 Line coding
4.3.2 Block Coding
4.4 Analog Data, Digital Signal
4.4.1 Sampling
4.4.2 Quantization
4.4.3 Pulse Amplitude Modulation (PAM)
4.4.4 Pulse Code Modulation (PCM)
4.4.5 Delta Modulation (DM)
4.5 Spread Spectrum
4.5.1 Basic Principle
4.5.2 Direct Sequence Spread Spectrum (DSSS)
4.5.3 Frequency Hopping Spread Spectrum (FHSS)
4.5.4 Time Hopping Spread Spectrum (THSS)
4.6 Key Points
4.7 Review Questions
84 Fundamentals of Communication
4.1 Introduction
Digital modulation techniques are essential to many digital communication systems,
whether it is a telephone system, a cellular communication system, or a satellite
communication system. In the past twenty years or so, researchers in digital modulation
techniques were very active and able to find out lot of positive contributions. Objective of
this chapter is to present the technique for generation of digital signal by modulating
analog and digital data. Application of spread spectrum technologies are expanding fast
which is also included in this chapter.
4.2 Digital Communication System

Figure 4.1 is the block diagram of a typical digital communication system. The message
to be sent may be from an analog source (e.g., voice) or from a digital source (e.g.
computer data). The analog-to-digital (A/D) converter samples and quantizes the analog
signal and represents the samples in digital form (bit 1 or 0). The source encoder accepts
the digital signal and encodes it into a digital signal. This is called source encoding,
which reduces the redundancy hence the transmission speed. This in turn reduces the
bandwidth requirement of the system. The channel encoder accepts the output digital
signal of the source encoder and encodes it into a digital signal.
Analog A/D Source Channel Modulator Power

Source Encoder Encoder Amplifier
Digital
Channel
Source
Analog D/A Source Channel De- Low Noise

User Decoder decoder modulator Amplifier
Digital
User
Figure 4.1: Block diagram of a typical digital communication system.

Redundancy is deliberately added into the coded digital signal so that some of the errors
caused by the noise or interference during transmission through the channel can be
corrected at the receiver. Then the encoded digital symbols are converted into a
waveform suitable for transmission. Usually there is a power amplifier following the
modulator. For high-frequency transmission, modulation and demodulation are usually
performed in the intermediate frequency (IF). For wireless systems an antenna is the final
stage of the transmitter. The transmission medium is usually called the channel, where
noise adds to the signal and fading and attenuation effects can occur.
At the receiving end first the received weak signal is amplified and demodulated. Then
the added redundancy is taken away by the channel decoder and the source decoder
recovers the signal to its original form before being sent to the user. A digital-to- analog
(D/A) converter is needed for analog signals.
Both analog and digital information can be encoded as either analog or digital signal. In
previous chapter we discussed how analog signal can be encoded both from analog data
and digital data. Here we will concentrate on how digital signal can be encoded from
digital data as well as analog data.
4.3 Digital Data, Digital Signal

A digital signal is a sequence of discrete, discontinuous voltage pulses. Each pulse is a
signal element. Binary data are transmitted by encoding each data bit into signal
elements. In the simplest case, there is a one-to-one correspondence between bits and
signal elements. Binary 1 and 0 are represented by different voltage levels. A variety of
encoding schemes having various combinations of voltage levels are also used.
4.3.1 Line coding

Line coding is the process of converting binary data, a sequence of bits, to a digital sig -
nal. For example, data, text, numbers, graphical images, audio, and video that are stored
in computer memory are all sequences of bits. Line coding converts a sequence of bits to
a digital signal. Figure 4.2 shows the concept of line coding.
Some Characteristics of Line Coding
Before we discuss the different types of line coding, we need to understand some
characteristics of line coding such as signal level versus data level, pulse rate versus
bit rate, dc components, and self-synchronization.
011010110 Line
Coding
Binary Data Digital Signal
Figure 4.2: Line coding.
Signal level and Data Level

As discussed previously, a digital signal can have a limited number of values. However,
only some of these values can be used to represent data; the rest are used for other
purposes, as we will see shortly. We refer to the number of values allowed in a particular
signal as the number of signal levels; we refer to the number of values used to represent
data as the number of data levels. Figure 4.3 shows two examples of digital signals. The
signal of Figure 4.3(a) has two signal levels and two data levels. The signal of Figure
4.3(b) has three signal levels and two data levels.
Amplitude
0 1 0 0 1 1 0 1
Time
(a) Two Signal level and two data Level
Amplitude
0 1 0 0 1 1 0 1
Time
(b) Three Signal level and two data Level
Figure 4.3: Signal level versus data level Amplitude

Bit Rate and Pulse rate

The pulse rate defines the number of pulses per second. A pulse is the minimum amount
of time required to transmit a symbol. The bit rate defines the number of bits per second.
If a pulse carries only 1 bit, the pulse rate and the bit rate are the same. If the pulse carries
more than 1 bit, then the bit rate is greater than the pulse rate. In general we have the
following formula, in which L is the number of data levels of the signal.
Bit Rate  Pulse Rate  log 2 L .
DC Components
Some line coding schemes leave a residual direct-current (dc) component (zero
frequency). This component is undesirable for two reasons. First, if the signal is to pass
through a system (such as a transformer) that does not allow the passage of a dc
component, the signal is distorted and may create errors in the output. Second, this
component is extra energy residing on the line and is useless.
Amplitude
0 1 0 0 1 1 0 1
(a) Signal with DC Components Time
Amplitude
0 1 0 0 1 1 0 1
Time
(b) Signal without DC Components
Figure 4.4: Signal with and without DC component

Figure 4.4 shows two line coding schemes. Figure 4.4 (a) has a dc component; the
positive voltages are not canceled by the negative voltages. Figure 4.4 (b) has no dc
component; the positive voltages are canceled by any negative voltages. The first does
not pass through a transformer properly; the second does.
Self-Synchronization
In order to correctly interpret the signals received from the sender, the receiver's bit
intervals must correspond exactly to the sender's bit intervals. If the receiver clock is
faster or slower, the bit intervals are not matched and the receiver might interpret the
signals differently than the sender intended. Figure 4.5 shows a situation in which the
receiver has shorter bit duration. The sender sends 01001101, while the receiver receives
0010000111011 (exaggerated situation).
Amplitude
0 1 0 0 1 1 0 1
Sent Time
Amplitude
0 0 1 0 0 0 0 1 1 1 0 1 1
Received Time
Figure 4.5: Lack of synchronization
A self-synchronizing digital signal includes timing information in the data being

transmitted. This can be achieved if there are transitions in the signal that alert the
receiver to the beginning, middle, or end of the pulse. If the receiver's clock is out of
synchronization, these alerting points can reset the clock.
Line coding schemes

We can divide line coding into three broad categories- Unipolar, Polar, Bipolar. Figure
4.6 shows the basic types of line coding.
Line Coding
Unipolar Polar Bipolar
Figure 4.6: Line coding schemes

Unipolar
Unipolar encoding is very simple and very primitive. Although it is almost obsolete
today, its simplicity provides an easy introduction to the concepts developed with the
more complex encoding systems and allows us to examine the kinds of problems that
any digital transmission system must overcome.
Amplitude Amplitude
0 1
Time Time
(a) Logic 0 (b) Logic 1
Amplitude
0 1 1 1 0 0 1 0
Time
(c) Timing Diagram
Figure 4.7: Unipolar encoding

Figure 4.7 shows the idea of unipolar encoding. In this example, the 1s are encoded
as a positive value, and the 0s are encoded as a zero value. In addition to being
straightforward, unipolar encoding is inexpensive to implement.
However, unipolar encoding has at least two problems that make it undesirable: a dc
component and a lack of synchronization. The average amplitude of a unipolar
encoded signal is nonzero. This creates a dc component. Lack of synchronization is also
an issue in unipolar encoding. If the data contain a long sequence of 0 s or 1 s, there is no
change in the signal during this duration that can alert the receiver to potential
synchronization problems.
Unipolar encoding uses only one voltage level.
Polar
Polar encoding uses two voltage levels, one positive and another one negative. By using
two levels, in most polar encoding methods the average voltage level on the line is
reduced and the dc component problem seen in unipolar encoding is alleviated.
Of the many existing variations of polar encoding, we examine four of the most popular:
Nonreturn to Zero (NRZ), Return to Zero (RZ), Manchester, and Differential Manchester
(Figure. 4.8).
Polar
NRZ RZ Manchester Differential

Manchester
Figure 4.8: Types of polar encoding
Polar encoding uses two voltage levels (Positive and Negative).

Nonreturn to Zero (NRZ): In NRZ encoding, the value of the signal is always either
positive or negative. There are two popular forms of NRZ termed as NRZ-L (Nonreturn
to Zero-Level) and NRZ-I (Nonreturn to Zero-Invert).
In NRZ-L encoding, the level of the signal depends on the type of bit that it represents. A
positive voltage usually means the bit is a 0, while a negative voltage means the bit is a 1;
thus, the level of the signal is dependent upon the state of the bit. A problem can arise
when the data contain a long stream of 0s or 1s. The receiver receives a continuous
voltage and determines how many bits are sent by relying on its clock, which may or may
not be synchronized with the sender clock.
In NRZ-L, the level of the signal is dependent upon the state of the bit.
Amplitude Amplitude
0 1
Time Time
Amplitude
0 1 1 1 0 0 1 0
Time
(c) Timing Diagram
Figure 4.9: NRZ-L encoding
In NRZ-I encoding, an inversion of the voltage level represents a 1 bit. It is the transition
between a positive and a negative voltage, not the voltage itself, that represents a 1 bit. A
0 bit is represented by no change. NRZ-I is superior to NRZ-L due to the synchronization
provided by the signal change each time a 1 bit is encountered. The existence of 1s in the
data stream allows the receiver to synchronize its timer to the actual arrival of the
transmission. A string of 0s can still cause problems, but because 0s are not as likely, they
are less of a problem.
Logic 0 - No Transition from current state

Logic 1- Transition from current state to
different state
Amplitude
0 1 1 1 0 0 1 0
Time
Figure 4.10: NRZ-I encoding
Figure 4.9 and Figure 4.10 respectively shows the NRZ-L and NRZ-I representations of
the same series of bits. In the NRZ-L sequence, positive and negative voltages have
specific meanings: positive for 0 and negative for 1. In the NRZ-I sequence, the voltages
per sequences are meaningless. Instead, the receiver looks for changes from one level to
another as its basis for recognition of 1s.
In NRZ-I, the level of the signal is inverted if a 1 is encountered.
Return to Zero (RZ): As it can be seen, anytime the original data contain strings of
consecutive 1s or 0s, the receiver can lose its place. A solution is to somehow include
synchronization in the encoded signal, something like the solution provided by NRZ-I,
but one capable of handling strings of 0s as well as 1s.
To ensure synchronization, there must be a signal change for each bit. The receiver can
use these changes to build up, update, and synchronize its clock. As we saw above, NRZ-
I accomplishes this for sequences of 1s. But to change with every bit, we need more than
just two values. One solution is return to zero (RZ) encoding, which uses three values:
positive, negative, and zero. In RZ, the signal changes not between bits but during each
bit. Like NRZ-L, a positive voltage means 1 and a negative voltage means 0. But, unlike
NRZ-L, halfway through each bit interval, the signal returns to zero. A 1 bit is actually
represented by positive-to-zero and a 0 bit by negative-to-zero, rather than by positive
and negative alone. Figure 4.11 illustrates the concept.
This transition is used for

Synchronization
Amplitude Amplitude
0 1
Time Time
Amplitude
0 1 1 1 0 0 1 0
Time
(c) Timing Diagram
Figure 4.11: RZ encoding
The main disadvantage of RZ encoding is that it requires two signal changes to

encode 1 bit and therefore occupies more bandwidth. But of the three alternatives we
have examined so far, it is the most effective.
A good encoded digital signal must contain a provision for synchronization .
Manchester Encoding: In Manchester encoding uses an inversion at the middle of

each bit interval for both synchronization and bit representation. A negative-to-
positive transition represents binary 1, and a positive-to-negative transition represents
binary 0. By using a single transition for a dual purpose, Manchester encoding
achieves the same level of synchronization as RZ but with only two levels of
amplitude. Figure 4.12 shows Manchester encoding.
In Manchester encoding, the transition at the middle of the bit is used for both
synchronization and bit representation. A negative-to-positive transition represents
binary 1, and a positive-to-negative transition represents binary 0.
Amplitude Amplitude
0 1
Time Time
Amplitude
0 1 1 1 0 0 1 0
Time
(c) Timing Diagram
Figure 4.12: Manchester encoding
Differential Manchester Encoding: In differential Manchester encoding, the inversion

at the middle of the bit interval is used for synchronization, but the presence or absence
of an additional transition at the beginning of the interval is used to identify the bit. A
transition means binary 0, and no transition means binary 1. Differential Manchester
encoding requires two signal changes to represent binary 0 but only one to represent
binary 1. Figure 4.13 shows Differential Manchester encoding.
In Differential Manchester encoding, the transition at the middle of the bit is used
only for both synchronization. The bit representation is defined by the inversion or
noninversion at the beginning of the bit.
Logic 0 - Transition from current state to different state

Logic 1- Continuation of current state i.e. no transition
Amplitude
0 1 1 1 0 0 1 0
Time
Figure 4.13: Differential Manchester encoding

Bipolar
Bipolar encoding, like RZ, uses three voltage levels: positive, negative, and zero. Unlike
RZ, however, the zero level in bipolar encoding is used to represent binary 0. The 1s are
represented by alternating positive and negative voltages. If the first 1 bit is represented
by the positive amplitude, the second will be represented by the negative amplitude, the
third by the positive amplitude, and so on. This alternation occurs even when the 1 bits
are not consecutive.
Logic 0 – Level Zero

Logic 1- Levels are positive and negative alternately
Amplitude
0 1 1 1 0 0 1 0
Time
Figure 4.14 Bipolar AMI encoding

A common bipolar encoding scheme is called bipolar Alternate Mark Inversion (AMI). In
the term alternate mark inversion, the word mark comes from telegraphy and means
1. So AMI means alternate 1 inversion. A neutral zero voltage represents binary 0. Binary
1s are represented by alternating positive and negative voltages. Figure 4.14 gives an
example of Bipolar AMI encoding.
In bipolar encoding, we use three levels are used: Positive, Negative and Zero.
A modification of bipolar AMI has been developed to solve the problem of synchronizing
sequential Os, especially for long-distance transmission. It is called BnZS (bipolar n-zero
substitution). In this scheme, wherever n consecutive zeros occur in the sequence, some
of the bits in these n bits become positive or negative which helps synchronization. This
substitution violates the rules of AMI in a specified manner such that the receiver
knows that these bits are actually 0s and not 1s.
Some Other Schemes
There are some other line coding schemes created for special purposes in data commu -
nications. We discuss two interesting ones here-
 Two binary, one quaternary (2B1Q)
 Multiline transmission, three level (MLT-3).
Amplitude
00 01 10 11 01 00 11 10
-1 Time
-3
Figure 4.15: 2B1Q encoding

2BlQ: The 2B1Q (two binary, one quaternary) uses four voltage levels. Each pulse can
then represent 2 bits, making each pulse more efficient. Figure 4.15 shows an example
of a 2B1Q signal.
MLT-3: In Multiline transmission, three level (MLT-3) is very similar to NRZ-I

(nonreturn to zero, invert), but it uses three levels of signals (+1, 0, and -1). The signal
transitions from one level to the next at the beginning of a 1 bit; there is no transition at
the beginning of a 0 bit. Figure 4.16 shows a sample MLT-3 signal.
Amplitude
0 1 1 1 0 0 1 1 1 1 0 1 1 0 1
Time
Figure 4.16: MLT-3 Signal encoding
4.3.2 Block Coding

With a view to improve the performance of line coding, block coding was introduced.
We need some kind of redundancy to ensure synchronization. In addition, we need to
include other redundant bits to detect errors. Block coding can achieve, to some extent,
these two goals. Figure 4.17 shows the procedure.
Steps for Block coding

In this method, there are three steps: division, substitution, and line coding. These are
explained below-
Division
In this step, the sequence of bits is divided into groups of m bits. For example, in
4B/5B encoding, the original bit sequence is divided into 4-bit groups.
Substitution
The heart of block coding is the substitution step. In this step, an m-bit code for an n-
bit group. For example, in 4B/5B encoding a 5-bit code is substituted for a 4-bit
group. With a 4-bit block, we can have 16 (=2 4) different groups. With a 5-bit code,
we can have 32 (=2 5) possible codes. This means that we can map some of the 5-bit
groups to the 4-bit groups. Some of the 5-bit codes are not used. We can apply a
strategy or a policy to choose only the 5-bit codes that help us in synchronization and
error detection. Figure 4.18 shows how we can use just one-half of the 5-bit codes.
010111010101 ... 01011110
m Bits m Bits m Bits

m Bit
Division 01101 10101 011011
m Bits m Bits m Bits Line

Coding
0101 1101 1110
m Bit to n Bit
Substitution
Digital Signal
Figure 4.17: Block coding
To achieve synchronization, we can use the 5-bit codes in such a way that, for
example, we do not have more than three consecutive 0s or 1s. Block coding can
definitely help in error detection. Because only a subset of the 5-bit codes is used, if
one or more of the bits in the block is changed in such a way that one of the unused
codes is received, the receiver can easily detect the error.
4-bit Blocks 5-bit Blocks
0000 00000
0001 00011
0010 00010
1000 10000
1111 11011
11111
Figure 4.18: Substitution in block coding

Line Coding
After the substitution, we can use one of the line coding schemes to create a signal.
Normally a very simple line coding scheme is chosen because the block coding
procedure provides two desirable features of complex line coding schemes. Sometimes,
as we will see, the second step (substitution) and the third step (line coding) are
combined.
Some Common Block Codes

4B/5B:
As described above, in 4B/5B, every 4 bits of data is encoded into a 5-bit code. The
selection of the 5-bit code is such that each code contains no more than one leading 0
and no more than two trailing 0s. Therefore, when these 5-bit codes are sent in
sequence, no more than three consecutive 0s are. encountered. The 5-bit codes are
normally line coded using NRZ-I. Table 4.1 shows the 4B/5B encoding. The encoded
sequences for control characters (column 3) do not follow the 4B/SB rules of coding.
Data Sequence Encoded Sequence Data Sequence Encoded Sequence

0000 11110 Q (Quit) 00000
0001 01001 I (Idle) 11111
0010 10100 H (Halt) 00100
0011 10101 J ( start delimiter) 11000
0100 01010 K ( start delimiter) 10001
0101 01011 T (end delimiter) 01101
0110 01110 S(Set) 11001
0111 10010 R(Reset) 00111
1000 10011
1001 10011
1010 10110
1011 10111
1100 11010
1101 11011
1110 11100
1111 11101
Table 4.1: 4B/5B encoding
8B/10B: This is similar to 4B/5B encoding except that a group of 8 bits of data is now
substituted by a 10-bit code. It provides more error detection capability than 4B/5B.
The 8B/10B encoding table is very long.
4.4 Analog data, Digital signal

4.4.1 Sampling
In this section we examine the process of transforming analog data into digital signals. It
is a process of converting analog data into digital data; this process is known as
digitization. The device used for converting analog data into digital form for transmission
and subsequently recovering the original analog data from the digital, is known as a
codec (coder-decoder). In this section we examine the two principal techniques used in
codecs, pulse code modulation and delta modulation.
Line coding and block coding can be used to convert binary data to a digital signal.
Sometimes, however, our data are analog, such as audio. Voice and music, for exam ple,
are by nature analog, so when we record voice or video, we have created an analog
signal. If we want to store the recording in the computer or send it digitally, we need to
change it through a process called sampling. After the analog signal is sampled, we can
store the binary data in the computer or use line coding (or a combination of block coding
and line coding) to further change the signal to a digital one so that it can be transmitted
digitally.
Consider a continuous varying signal f(t) that is to be converted to digital form. We do this
simply by first sampling f(t) periodically at a rate of fc samples per second. Although in
practice this sampling process would presumably be carried out electronically by getting
the signal on and off at the desired rate, we show the sampling process conceptually in
Figure. 4.19 using a rotating mechanical switch.
Rotate fc times per sec
f (t)
Transmitting
fs (t)
System
(a)
Amplitude Amplitude
T

Input Time Output Time

(b)
Figure 4.19: (a) Sampling of a Analog signal (b) Sampling process input and output
Assume that the switch remains on the f(t) line τ seconds, while rotating at the desired rate
of fc  1 times per second (τ <<T). The switch output, f s  t  is a sampled version of
T
f(t). A typical input function f(t), the sampling times, and the sampled output f s  t  are
shown in Figure. 4.19, here fc is called sampling rate and T is called the sampling
interval.
The question that immediately arises is: what should the sampling rate be? Are there any
limits to the rate at which we sample? One might intuitively feel that the process of
sampling has irretrievably distorted the original signal f(t). The process of sampling has
been introduced to convert the signal f(t) to a digital form for further processing and
transmission. A signal can be reconstructed from the sampled signals, if the signal is
sampled maintaining the sampling theorem, i.e. Nyquist Theorem.
Nyquist Theorem: If a signal is sampled in a regular interval of time and a rate higher
than the twice the highest signal frequency, then the samples contain all the information
of the original signal.
The theorem emphasizes that an analog signal has high time redundancy which can be
utilized for transmitting additional information. The telephone voice channel has the
highest frequency of 3400 Hz. If it is sampled at the rate of more than 6800 samples
per second, the speech signal can be entirely reconstructed from the samples. It is to
be ensured, before sampling is carried out, that the speech signal does not contain any
frequency component higher than 3400 Hz. Therefore, a low pass filter is always
provided just before the sampler. The standard sampling rate for the speech signal is
8000 samples/second to allow for gradual slope of the filter characteristics. At this
rate, the samples are separated by 125 μs (Figure. 4.19(b)).
4.4.2 Quantization
Quantization is the process of rounding off the values of the samples to certain
predetermined levels in order to make a finite and manageable number of levels available
to the A/D converter. Otherwise the sampling levels could take any value within the peak-
to-peak range of the analog signal, which in theory would result in an infinite, number
of levels.
In the quantization process, the total signal range is divided into a number of subranges as
shown in Figure. 4.20 Each subrange has its mid-value designated as the standard or code
level for that range. Comparators are used to determine which subrange a given pulse
amplitude is in, and the code for that subrange is generated.
To illustrate the process, a comparatively small number of levels, eight in all, are shown,
along with a possible binary code for each level. It will be noticed that the level
representing zero analog volts has two binary numbers, one for 0+ and one for 0-. In the
coding scheme shown, positive values of the analog signal are signed with a binary l and
negative values with a binary 0. Thus the leading-bit indicates the polarity of the ana log
signal. The remaining two bits then encode the segment that the sampled value lies in.
For example, the last sample point shown is negative, and therefore the leading bit is 0; it
lies in segment L-3 for which the binary code is 11; hence the binary code for this
quantized sample is 011.
Amplitude
L4
L3
L2
L1
L -1
L -2
L -3
L -4
Time
L 4L -2L 3L 1L -3L 2L -1L -4000001010011100101110111
Figure 4.20: Quantization level and corresponding code
4.4.3 Pulse Amplitude Modulation (PAM)

One analog-to-digital conversion method is called pulse amplitude modulation

(PAM). This technique takes an analog signal, samples it, and generates a series of
pulses based on the results of the sampling. The term sampling means measuring the
amplitude of the signal at equal intervals.
Amplitude
Time
(a) Analog signal

Amplitude
Time
(b) PAM signal
Figure 4.21: Pulse Amplitude Modulation
The method of sampling used in PAM is more useful to other areas of engineering
than it is to data communication. However, PAM is the foundation of an important
analog-to-digital conversion method called pulse code modulation (PCM).
In PAM, the original signal is sampled at equal intervals, as shown in Figure 4.21.
PAM uses a technique called sample and hold. At a given moment, the signal level is
read, then held briefly. The sampled value occurs only instantaneously in the actual
waveform, but is generalized over a short but measurable period in the PAM result.
PAM is not useful to data communications because even though it translates the
original waveform to a series of pulses, these pulses are still of any amplitude (still an
analog signal, not digital). To make them digital, we must modify them by using
pulse code modulation.
Pulse Amplitude Modulation is the first step in another very popular conversion
method called Pulse Code Modulation.
4.4.4 Pulse Code Modulation (PCM)

Pulse Code Modulation (PCM) modifies the pulses created by PAM to create a complete
digital signal. To do so, PCM first quantizes the PAM pulses. Quantization is a method of
assigning integral values in a specific range to sampled instances. The result of
quantization is presented in Figure 4.22.
Amplitude
127 120 120

120
100 80 80
80 56
60 42 22 22
20 18
40 18 12
18 08 14
20
0 Time
-20
-06 -04
-40
-22 -22
-60 -42 -42
-80 -56
-62
-100
-120
-127
Figure 4.22: Quantized PAM signal Amplitude

Figure 4.23 shows a simple method of assigning sign and magnitude to quantized
samples. Each value is translated into its 7-bit binary equivalent. The eighth bit indicates
the sign.
Sign Bit: (+) is Zero and (-) is One
Quantized
ValueCodeQuantized
ValueCodeQuantized
ValueCode2000010100-4210101010120011110001800010010-6210111110-
56101110004600101110-0610000110-
221001011040001010001400001110220001011008000010008001010000-
0410000100
Figure 4.23: Quantizing by using sign and magnitude
The binary digits are then transformed to a digital signal by using one of the line coding
techniques. Figure 4.24 shows the result of the pulse code modulation of the original
signal encoded finally into a unipolar signal. Only the first three sampled values are
shown.
Amplitude
00010100 00010010 00101110 00101000
Time
Figure 4.24: Encoded Sample

PCM is actually made up of four separate processes: PAM, quantization, binary encoding,
and line coding. Figure 4.25 shows the entire process in graphical form. PCM is the
sampling method used to digitize voice in T-line transmission in telecommunication
system.
000110000100110 …
Quantization
Binary Data
Analog Data
118
32
Pulse
Amplitude Line
Modulation Coding
-42
Quantized Data
Binary
Encoding
Sampled Analog Data Digital Signal
Figure 4.25: Conversion of analog signal to PCM digital code
4.4.5 Delta Modulation (DM)

A variety of techniques have been used to improve the performance of PCM or to reduce
its complexity. One of the most popular alternatives to PCM is delta modulation.
With delta modulation, an analog input is approximated by a staircase function that
moves up or down by one quantization level (δ) at each sampling interval (TS). An
example is shown in Figure 4.26, where the staircase function is overlaid on the original
analog waveform. The important characteristics of this staircase function is that its
behavior is binary, that is at each sampling time, the function moves up or down a
constant amount δ. Thus, the output of the delta modulation process can be represented as
a single binary digit for each sample. In essence, a bit stream is produced by
approximating the derivative of an analog signal rather than its amplitude. A 1 is
generated if the staircase function is to go up during the next interval; a 0 is generated
otherwise.
Figure 4.26: An example of delta modulation
The transition (up or down) that occurs at each sampling interval is chosen so that the
staircase function tracks the original analog waveform as closely as possible. Figure 4.27
illustrates the logic of the process, which is essentially a feedback mechanism. For
transmission, at each sampling time, the analog input is compared to the most recent
value of the approximating staircase function. If the value of the sampled waveform
exceeds that of the staircase function, a 1 is generated; otherwise, a 0 is generated. Thus,
the staircase is always changed in the direction of the input signal. The output of the delta
modulation process is therefore a binary sequence that can be used at the receiver to
reconstruct the staircase function. The staircase function can then be smoothed by some
type of integration process or by passing it through a lowpass filter to produce an analog
approximation of the analog input signal.
Figure 4.27: Block diagram of delta modulation

There are two important parameters in a delta modulation scheme-

 The size of the step assigned to each binary digit δ
 The sampling rate.
As Figure 4.27 illustrates, δ must be chosen to produce a balance between two types of
errors or noise. When the analog waveform is changing very slowly, there will be
quantizing noise. This noise increases as δ is increased. On the other hand, when the
analog waveform is changing more rapidly than the staircase can follow, there is slope
overload noise. This noise increases as δ is decreased. It should be clear that the accuracy
of the scheme can be improved by increasing the sampling rate. However, this increases
the data rate of the output signal. The principal advantage of delta modulation over PCM
is the simplicity of its implementation. In general, PCM exhibits better SNR
characteristics at the same data rate.
4.5 Spread Spectrum

4.5.1 Basic Principle
In spread spectrum system, input is fed into a channel encoder that produces an analog
signal with a relatively narrow bandwidth around some center frequency. This signal is
further modulated using a sequence of digits known as a spreading code or spreading
sequence. Typically, but not always, the spreading code is generated by a pseudonoise, or
pseudorandom number, generator. The effect of this modulation is to increase
significantly the bandwidth of the signal to be transmitted. On the receiving end, the same
digit sequence is used to demodulator the spread spectrum signal. Finally, the signal is
fed into a channel decoder to recover the data. Figure 4.28 highlights the key
characteristics of any spread spectrum system.
Figure 4.28: General model of spread spectrum digital communication system

Several things can be gained from this apparent waste of spectrum.
We can gain immunity from various kinds of noise and multipath distortion. The earliest
applications of spread spectrum were military, where it was used for its immunity to
jamming. It can also be used for hiding and encrypting signals. Only a receipient who
knows the spreading code can recover the encoded information.
Several users can independently use the same higher bandwidth with very little
interference. This property is used in cellular telephony applications, with a technique
know as code division multiplexing (CDM) or code division multiple access (CDMA).
Wide Band Systems

The entire bandwidth is made available to each user; and is many times larger than the
bandwidth required to transmit information. Such systems are called "Spread Spectrum"
systems. These include:
 Direct Sequence Spread Spectrum (DSSS)
 Frequency Hopping Spread Spectrum (FHSS)
 Time Hopping Spread Spectrum (THSS)
4.5.2 Direct Sequence Spread Spectrum (DSSS)

DSSS is an averaging type system where the reduction of interference occurs because the
interference is averaged over a large time interval. Frequency hopping spread spectrum
(FHSS) and time hopping spread spectrum (THSS) systems are avoidance systems, where
reduction in interference occurs because the signal is made to avoid the interference a
large fraction of time by changing the frequency or time slot. Figure 4.29 illustrates
Direct Sequence Spread Spectrum.
In DSSS system the transmission bandwidth exceeds the coherence bandwidth. The
received signal, after despreading, resolves into multiple signals with different time
delays. A rake receiver is used to recover the multiple time delayed signals and combine
them into one signal, providing an inherent time diversity receiver with lower frequency
of deep fades. The DSSS systems provide an inherent robustness against mobile-channel
degradation. Also, DSSS systems have greater resistance to interference effects in a
frequency reuse situation.
Figure 4.29: Direct Sequence Spread Spectrum
4.5.3 Frequency Hopping Spread Spectrum (FHSS)

FH is the periodic changing of frequency or frequency set associated with transmission. If
the modulation is multiple Frequency Shift Keying (FSK), two or more frequencies are in
the set that change at each hop. For other modulation, a single center or carrier frequency
is changed at each hop. Figure 4.30 illustrates Frequency Hopping Spread Spectrum.
Frequency
fn
fn-1
f3
f2
f1
0 Tc 2Tc Time
Figure 4.30: Frequency Hopping Spread Spectrum

4.5.4 Time Hopping Spread Spectrum (THSS)
Transmission time is divided into intervals called as frames. Each frame is divided into
time slots. During each frame one and only one slot is modulated with a message. All of
the message bits accumulated in previous frames are transmitted in a burst during the
selected time slot. Figure 4.31 illustrates Time Hopping Spread Spectrum.
M = Time Slot in Each frame

t = Tf / M
Transmitted Time slot
One Frame
0 τ Tf 2Tf 3Tf Time
Figure 4.31: Time Hopping Spread Spectrum

4.6 Key points

 Line coding is the process of converting binary data to a digital signal.
 The number of different values allowed in a signal is the signal level. The
number of symbols that represent data is the data level.
 Bit rate is a function of the pulse rate and data level.
 Line coding methods must eliminate the dc component and provide a means of
synchronization between the sender and the receiver.
 Line coding methods can be classified as unipolar, polar, or bipolar.
 NRZ, RZ, Manchester, and differential Manchester encoding are the most
popular polar encoding methods.
 AMI is a popular bipolar encoding method.
 Block coding can improve the performance of line coding through redundancy
and error correction.
 Block coding involves grouping the bits; substitution, and line coding. 4B/5B,
8B110B, and 8B/6T are common block coding methods. Analog-to-digital
conversion relies on PCM (pulse code modulation). PCM involves sampling,
quantizing, and line coding.
 The Nyquist theorem says that the sampling rate must be at least twice the
highest frequency component in the original signal.
 Delta modulation improves the performance of PCM.
 DSSS is an averaging type system where the reduction of interference occurs.
 The DSSS systems provide an inherent robustness against mobile-channel
degradation.
 DSSS systems have greater resistance to interference effects in a frequency reuse
situation.
 FH is the periodic changing of frequency or frequency set associated with
transmission.
 In THSS transmission time is divided into intervals called as frames.
4.7 Exercises
Review Questions
1. What do you mean by digital communication?
2. What is Line coding?
3. What is the relation between bit rate and pulse rate?
4. Give the signal level for each line coding method discussed (NRZ, RZ, etc.).
5. What is the dc component?
6. Can the bit rate be less than the pulse rate? Why or why not?
7. Why is synchronization a problem in data communications?
8. How does NRZ-L differ from NRZ-I?
9. What is the major disadvantage in using NRZ encoding? How does RZ encoding
attempt to solve the problem?
10. What are the three major steps in block coding?
11. Repeat Exercise 13 for a data stream of ten 1s.
12. Repeat Exercise 13 for a data stream of 10 alternating 0s and 1s.
13. Repeat Exercise 13 for a data stream of three 0s followed by two 1s followed by
two 0s and another three 1s.
14. Using the Nyquist theorem, calculate the sampling rate for the following analog
signals. a) An analog signal with bandwidth of 2000 Hz. b) An analog signal
with frequencies from 2000 to 6000 Hz. c) A signal with a horizontal line in the
time-domain representation. d) A signal with a vertical line in the time-domain
representation.
15. Suppose a signal is sampled 8000 times per second, what is the interval between
each sample?
16. How can block coding aid in synchronization?
17. Figure 4.32 is the unipolar encoding of a data stream. What is the data stream?
Figure 4.32
18. Figure 4.33 is the NRZ-L encoding of a data stream. What is the data stream?
Figure 4.33
19. What is the sampling rate for PCM if the frequency ranges from 1000 to 4000
Hz?
20. What is the Nyquist theorem?
21. Assume a data stream is made of ten 0s. Encode this stream, using the following
encoding schemes. How many changes (vertical line) can you find for each
scheme? a) Unipolar b) NRZ-L c) NRZ-I d) RZ e) Manchester f) Differential
Manchester g. AMI.
22. If the interval between two samples in a digitized signal is 125 μs what is the
sampling rate?
23. A signal is sampled. Each sample represents one of four levels. How many bits
are needed to represent each sample? If the sampling rate is 8000 samples per
second, what is the bit rate?
24. What is “Spread Spectrum” system? What are the basic types?
25. What do you mean by DSSS? Why it is used?
26. Illustrate Direct Sequence Spread Spectrum.
27. What do you mean by FHSS? Show the timing diagram.
28. What do you mean by THSS?
29. Compare DSSS, FHSS and THSS.
Multiple Choice Question

1. ASK, PSK, FSK, and QAM are examples of_________modulation.
a. Digital-to-digital
b. Digital-to-analog
c. Analog-to-analog
d. Analog-to-digital
2. AM and FM are examples of _________ modulation.
a. Digital-to-digital
b. Digital-to-analog
c. Analog-to-analog
d. Analog-to-digital
3. In QAM, both phase and _________ of a carrier frequency are varied.
a. Amplitude
b. Frequency
c. Bit rate
d. Baud rate
4. Which of the following is most affected by noise?
a. PSK
b. ASK
c. FSK
d. QAM
5. If the baud rate is 400 for a 4-PSK signal, the bit rate is _________ bps.
a. 100
b. 400
c. 800
d. 1600
6. If the bit rate for an ASK signal is 1200 bps, the baud rate is
a. 300
b. 400
c. 600
d. 1200
7. If the bit rate for an FSK signal is 1200 bps, the baud rate is
a. 300
b. 400
c. 600
d. 1200
8. If the bit rate for a QAM signal is 3000 bps and a signal unit is represented by a
tribit, what is the baud rate?
a. 300
b. 400
c. 1000
d. 1200
9. If the baud rate for a QAM signal is 3000 and a signal unit is represented by a •
tribit, what is the bit rate?
a. 300
b. 400
c. 1000
d. 9000
10. If the baud rate for a QAM signal is 1800 and the bit rate is 9000, how many bits
are there per signal unit?
a. 3
b. 4
c. 5
d. 6
11. In 16-QAM, there are 16 _________
a. Combinations of phase and amplitude
b. Amplitudes
c. Phases
d. bps
12. Which modulation technique involves tribits, eight different phase shifts, and
one amplitude?
a. FSK
b. 8-PSK
c. ASK
d. 4-PSK
13. Given an AM radio signal with a bandwidth of 10 KHz and the highest-
frequency component at 705 KHz, what is the frequency of the carrier signal?
a. 700 KHz
b. 705 KHz
c. 710 KHz
d. Cannot be determined from given information
14. A modulated signal is formed by _________
a. Changing the modulating signal by the carrier wave
b. Changing the carrier wave by the modulating signal
c. Quantization of the source data
d. Sampling at the Nyquist frequency
15. When an ASK signal is decomposed, the result is _________
a. Always one sine wave
b. Always two sine waves
c. An infinite number of sine waves
d. None of the above
16. The bandwidth of an FM signal requires 10 times the bandwidth of the

_________ signal.
a. Carrier
b. Modulating
c. Bipolar
d. Sampling
17. Modulation of an analog signal can be accomplished through changing the
_________ of the carrier signal.
a. Amplitude
b. Frequency
c. Phase
d. Any of the above
18. For a telephone line, the bandwidth for voice is usually _________ the
bandwidth for data.
a. Equivalent to
b. Less than
c. Greater than
d. Twice
19. For a given bit rate, the minimum bandwidth for ASK is _________ the
minimum bandwidth for FSK.
a. Equivalent to
b. Less than
c. Greater than
d. Twice
20. As the bit rate of an FSK signal increases, the bandwidth _________
a. Decreases
b. Increases
c. Remains the same
d. Doubles
21. For FSK, as the difference between the two carrier frequencies increases, the
bandwidth
a. Decreases
b. Increases
c. Remains the same
d. Halves
22. The bit rate always equals the baud rate in which type of signal?
a. FSK
b. QAM
c. 4-PS K
d. All the above
23. A modulator converts a(n) _________ signal to a(n)_________signal.
a. Digital; analog
b. Analog; digital
c. PSK; FSK
d. FSK; PSK
Chapter 4
Digital Modulation
4.1 Introduction 84
4.2 Digital Communication System 84
4.3 Digital Data, Digital Signal 85

4.3.1 Line coding 85
4.3.2 Block Coding 97
4.4 Analog data, Digital signal 100

4.4.1 Sampling 100
4.4.2 Quantization 102
4.4.3 Pulse Amplitude Modulation (PAM) 103
4.4.4 Pulse Code Modulation 105
4.4.5 Delta Modulation 107
4.5 Spread Spectrum 110

4.5.1 Basic Principle 110
4.5.2 Direct Sequence Spread Spectrum (DSSS) 111
4.5.3 Frequency Hopping Spread Spectrum (FHSS) 112
4.5.4 Time Hopping Spread Spectrum (THSS) 113
4.6 Key points 113
4.7 Exercises 114

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Chapter 4: Digital Modulation 83

4.2 Digital Communication System

Analog A/D Source Channel Modulator Power

Analog D/A Source Channel De- Low Noise

Figure 4.1: Block diagram of a typical digital communication system.

4.3 Digital Data, Digital Signal

4.3.1 Line coding

Binary Data Digital Signal

Figure 4.2: Line coding.

Signal level and Data Level

(b) Three Signal level and two data Level

Figure 4.3: Signal level versus data level Amplitude

Bit Rate and Pulse rate

(a) Signal with DC Components Time

(b) Signal without DC Components

Figure 4.4: Signal with and without DC component

Figure 4.5: Lack of synchronization

A self-synchronizing digital signal includes timing information in the data being

Line coding schemes

Unipolar Polar Bipolar

Figure 4.6: Line coding schemes

Figure 4.7: Unipolar encoding

Unipolar encoding uses only one voltage level.

NRZ RZ Manchester Differential

Figure 4.8: Types of polar encoding

Polar encoding uses two voltage levels (Positive and Negative).

(c) Timing Diagram

Figure 4.9: NRZ-L encoding

Logic 0 - No Transition from current state

Figure 4.10: NRZ-I encoding

In NRZ-I, the level of the signal is inverted if a 1 is encountered.

This transition is used for

(c) Timing Diagram

Figure 4.11: RZ encoding

The main disadvantage of RZ encoding is that it requires two signal changes to

A good encoded digital signal must contain a provision for synchronization .

Manchester Encoding: In Manchester encoding uses an inversion at the middle of

(a) Logic 0 (b) Logic 1

(c) Timing Diagram

Figure 4.12: Manchester encoding

Differential Manchester Encoding: In differential Manchester encoding, the inversion

Logic 0 - Transition from current state to different state

Figure 4.13: Differential Manchester encoding

Logic 0 – Level Zero

Figure 4.14 Bipolar AMI encoding

Figure 4.15: 2B1Q encoding

MLT-3: In Multiline transmission, three level (MLT-3) is very similar to NRZ-I

Figure 4.16: MLT-3 Signal encoding

4.3.2 Block Coding

Steps for Block coding

010111010101 ... 01011110

m Bits m Bits m Bits

m Bits m Bits m Bits Line

Figure 4.17: Block coding

4-bit Blocks 5-bit Blocks

Figure 4.18: Substitution in block coding

Some Common Block Codes