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Chapter Five

Pulse Modulation

5.1 Introduction
In continuous –wave (CW) modulation, some parameter of a sinusoidal carrier wave

is varied continuously in accordance with the message signal .However, in pulse

modulation, some parameter of a pulse train is varied in accordance with the

message signal. We have two families of pulse modulation:

a) Analog pulse modulation:- a periodic pulse train is used as the carrier wave

and some characteristics features of each pulse(amplitude, duration or

position) is varied in a continuous manner in accordance with the

corresponding sample value of the message signal.

b) Digital Modulation: - the message signal is represented in a form that is

discrete in both time and amplitude, thereby permitting its transmission in

digital form as a sequence of coded pulses.

We begin the discussion by describing the sampling process, which is basic to all

pulse modulation systems, whether they are analog or digital.

5.2 Sampling Process

We may now state the Sampling theorem for strictly band-limited signals of finite

energy in two equivalent parts, which apply to the transmitter and receiver of a

pulse modulation system, respectively:

1. A band-limited signal of finite energy, which has no frequency components

higher than
 Hertz, is completely described by specifying the values of

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the signal at instants of time separated by 
seconds. In other word,



to completely described the signal m(t), it should satisfy condition:

 ≤ 12 (5.1)
i.e., the maximum allowable sampling time is  = 12 .

2. A band-limited signal of finite energy, which has no frequency components

higher than
 Hertz, may be completely recovered from knowledge of its

samples taken at the rate of 
 samples per second. In other word, to

completely recovered the signal m(t), it should satisfy condition:

 ≥2 (5.2)
i.e., the minimum value of the sampling rate is  =2 .
The sampling rate of 
 samples per second, for a signal bandwidth of
 Hertz, is
called the Nyquist rate; its reciprocal 
seconds is called the Nyquist interval.


Figure 5.1 (a) A signal m(t) which is to be sampled (b) The sampling function s(t) consists
of a train of very narrow unit amplitude pulses (c) The sampling operation is performed in a
multiplier (d) The samples of the signal m(t)

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The baseband signal m(t) which is to be sampled is shown in fig 5.1a. A periodic train of

pulses s(t) of unit amplitude and of period  is shown in fig5.1b. The pulses are arbitrarily

narrow, having a width dt. The two signals m(t) and s(t) are applied to a multiplier as shown

in fig5.1c, which then yields as an output the product s(t)m(t). This product is seen in fig

5.1d to be the signal m(t) sampled at the occurrence of each pulse. That is, when a pulse

occurs, the multiplier output has the same value as does m(t), and all other times the

multiplier output is zero.

The signal s(t) is periodic, with period  , and has the Fourier series(see example 1 of

chapter 1) with  =  and  =  .

∞
 2 2
() = +  cos (5.3)
  
!"

Then, the product signal ()$() is

∞
 2 2
()$() = $() +  m(t)cos (5.4)
  
!"

The requirement that  ≥2 applies to baseband or low-pass signals. However, when

bandpass signals are to be sampled, lower sampling rates can sometimes be used. We can

state bandpass sampling theorem as follows:

‘’A bandpass signal with highest frequency
( and bandwidth B, can be recovered from its

(
samples through bandpass filtering by sampling it with frequency
) = *, where k is

(
the largest integer not exceeding + ‘’.

Example 5.1 A signal $() = 2,-6000 + 4,-8000 + 6,-10000 is to be truthfully

represented by its samples. What is the minimum sampling rate from

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a. Low-pass sampling theorem consideration

b. Bandpass sampling theorem consideration

Solution: The signal given has two frequency components

Highest frequency, 1 =5000Hz

Lowest frequency, 3
=3000HZ

a)  =2 =2 1 = 245000 = 1000056
b) 7 = 1 − 9 = 5000 − 3000 = 200056 ,

k=floor( 5 /7)=floor(5000/2000)=floor(2.5)=2 where floor(x) gives the largest

integer that does not exceed x.

2 5< =>?@@@
Therefore,  = ; = = = 500056

5.2.1 Natural sampling

A technique by which we may take the advantage of sampling principle for the purpose of

time-division multiplexing is illustrated in the idealized representation of Fig 5.2. At the

transmitting end on the left, a number of bandlimited signals are connected to the contact

point of a rotary switch. We assume that signals are similarly bandlimted. For example,

they may all be voice signals, limited to 3.3kHz. As the rotary arm of the switch swings

around, it samples each signal sequentially. The rotary switch at the receiving end is in

synchronism with the switch at the sending end. The two switches make contact

simultaneously at similarly numbered contacts with each revolution of the switch; one

sample is taken of each input signal and presented to the correspondingly numbered

contact of the receiving end switch. The train of samples at, say, terminal 1 in the

receiver, pass through low-pass filter 1, and, at the filter output, the original signal $" ()

appears reconstructed. Of course, if
 is the highest-frequency spectral component

present in any of the input signals, the switches must make at least 
 revolutions

per second.

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Figure 5.2 illustrating how the sampling principle may be used to transmit a number

of bandlimited signals over a single-communication channel

When the signals to be multiplexed vary slowly with time, so that the sampling rate

is correspondingly slow, mechanical switches, indicated in Fig 5.2, may be employed.

When the switching speed required is outside the range of mechanical switches,

electronic switching systems may be employed. In either event, the switching

mechanism, corresponding to the switch at the lift in Fig 5.2, which samples the

signals, is called commutator. The switching mechanism which performs the functions
of the switch at the right in Fig 5.2 is called the decommutator. The commutator samples

and combines samples, while the decommutator separates samples belonging to

individual signals so that these signals may be reconstructed.

A much more reasonable manner of sampling, referred to as natural sampling, is

shown in Fig 5.3. Here the sampling waveform s(t) consists of a train of pulses

having duration A and separated by the sampling time  . The baseband signal is

m(t), and the sampled signal s(t)m(t) is shown in Fig 5.3c.observe that the sampled

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signal consists of a sequence of pulses of varying amplitude whose tops are not

flat but follow the waveform of the signal m(t).

Figure 5.3 (a) A baseband signal m(t) (b) A sampling signal s(t) with pulses of finite

duration (c) The naturally sampled signal s(t)m(t)

With natural sampling, as with instantaneous sampling (commutation and

decommutation), a signal sampled at the Nyquist rate may be reconstructed

exactly by passing the samples through an ideal LPF with cutoff at the

frequency B, where B is the highest frequency spectral component of the signal.

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5.2.2 Flat-Top sampling

Pulses of the type shown in Fig5.3c, with tops contoured to follow the wave-form of the

signal are actually not frequently employed. Instead flat-topped pulses are customarily

used, as shown in Fig 5.4

Figure 5.4 Flat-topped sampling

A flat-topped pulse has constant amplitude established by the sample value of the

signal at some point within the pulse interval. In Fig 5.4a we have arbitrarily

sampled the signal at the beginning of the pulse. In sampling of this type the

baseband signal m(t) cannot be recovered exactly by simply passing the samples

through an ideal LPF. However, the distortion need not be large. Flat-top

sampling has the merit that it simplifies the design of the electronic circuitry

used to perform the sampling operation.

5.3 Analog Pulse Modulation

5.3.1 Pulse Amplitude Modulation(PAM)

Now that we understand the essence of the sampling process, we are ready to

formally define pulse-amplitude modulation, which is the simplest and most basic

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form of analog pulse modulation. In pulse-amplitude modulation (PAM), the

amplitudes of regularly spaced pulses are varied in proportion to the

corresponding sample values of a continuous message signal; the pulses can be of

a rectangular form or some other appropriate shape. Pulse-amplitude modulation as

defined here is somewhat similar to natural sampling, where the message signal is

multiplied by a periodic train of rectangular pulses. However, in natural sampling

the top of each modulated rectangular pulses varies with the message signal,

whereas in PAM it is maintained flat.

The waveform of a PAM signal is illustrated in figure 5.5. The dashed curve in this

figure depicts the waveform of a message signal m(t), and the sequence of

amplitude-modulated rectangular pulses shown as solid lines modulated represents

the corresponding PAM signal s(t).

Figure 5.5 Flat-top samples, representing an analog signal

There are two operations involved in the generation of the PAM signal:

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1. Instantaneous sampling of the message signal m(t) every  seconds, where

the sampling rate  = 1⁄  is chosen in accordance with the sampling

theorem.

2. Lengthening the duration of each sample so obtained to some constant vale

T.

In digital circuit technology, these two operations are jointly referred to as

‘’sample and hold’’. One important reason for intentionally lengthening the

duration of each sample is to avoid the use of an excessive channel

bandwidth, since bandwidth is inversely proportional to pulse duration.

5.3.2 Pulse Duration Modulation (PDM)

In pulse modulation system we may use the increased bandwidth consumed by the

pulses to improve the noise performance of the system. This can be achieved by

representing the sample values of the message signals by some property of the

pulse other than amplitude.

Pulse-duration modulation (PDM), also referred to as pulse-width modulation

(PWM), where samples of the message signal are used to vary the duration of the

individual pulses in the carrier. Figure 5.6c represents the PDM wave.

5.3.3 Pulse Position Modulation (PPM)

Pulse-position modulation (PPM), the position of a pulse relative to its unmodulated

time of occurrence is varied in accordance with the message signal. Figure 5.6d

represents PPM wave.

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Figure 5.6 illustrating two different forms of pulse-time modulation for the case
of a sinusoidal modulating wave (a) modulating wave (b) pulse carrier (c) PDM wave
(d) PPM wave

5.4 Quantization of Signals

Quantization is defined as the process of transforming the sample amplitude of

m(nTF ) of a message signal m(t) at time t = nTF into a discrete amplitude v(nTF )

taken from a finite set of possible amplitudes. When quantizing a signal m(t), we

create a new signal $G () which is an approximation to m(t). The operation of

quantization is represented in Fig 5.7.

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Figure 5.7 The operation of Quantization.

If you are given an analog signal m(t) and the number of bits N, we can determine
the quantization parameters:

1. The quantization levels , H = 2I

2. The step size, s
($BJ> − $BK )
=
H
3. The index corresponding to the binary code is

M-N($ − $BK )
L=

4. The values of the quantization levels

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$K = $BK + L ∗ 
Where i=0, 1, 2….M

The difference $() − $G () or error due to the quantization process is called

quantization error. This quantization error can be regard as a noise and is called

quantization noise.

5.5 Digital Pulse Modulation

5.5.1 Pulse-Code Modulation (PCM)

A signal which is to be quantized prior to transmission is usually sampled as well.

The quantization is used to reduce the effects of noise, and the sampling allows

us to time division multiplex a number of messages. The combined operations of

sampling and quantizing generate a quantized PAM waveform, that is, a train of

pulses whose amplitudes are restricted to a number of discrete magnitudes.

We may transmit these quantized sample values directly. Alternatively we may

represent each quantized level by a code number and transmit the code number

rather than the sample value itself. Most frequently the code number is

converted, before transmission, into its representation in binary arithmetic, i.e.,

base-2 arithmetic. The digits of the binary representation of the code number

are transmitted as pulses. Hence the system of transmission is called binary

pulse-code modulation (PCM).

The essential features of binary PCM are shown in Fig 5.8. We assume that the

analog message signal m(t) is limited in its excursions to the range from -4 to +4

volts. We have set the step size between quantization levels at 1volt.Eight

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quantization levels are employed, and these are located at -3.5,-2.5,…,+3.5 volts.

We assign the code number 0 to the level at -3.5 volts, the code number 1 to the

level at -2.5 volts, etc., until the level at +3.5 volts, which is assigned the code

number 7.Each code number has its representation in binary arithmetic ranging

from 000 for code number 0 to 111 for code number 7.

Figure 5.8 A message signal is regularly sampled, quantization levels are indicated.
For each sample the quantized value is given and its binary representation is
indicated

In Fig 5.8, in correspondence with each sample, we specify the sample value, the
nearest quantization, and the code number and its binary representation. If we
were transmitting the analog signal, we would transmit the sample values 1.3, 3.6,
2.3 etc. If we were transmitting the quantized signal, we would transmit the

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quantized sample values 1.5, 3.5, 2.5 etc. In binary PCM we transmit the binary
representations 101,111,110, etc.

A PCM communication system is represented in Fig 5.9. The analog signal m(t) is
sampled, and these samples are subjected to the operation of quantization. The
quantized samples are applied to an encoder. The encoder responds to each such
sample by the generation of a unique and identifiable binary pulse (or binary level)
pattern.
Communication channel

Analog-to- digital converter

Analog
Sampler Quantizer Encoder Quantizer Decoder
Signal m(t)

Quantized PAM Digitally encoded signal

Figure 5.9 A PCM communication system.

The combination of the quantizer and encoder in the dashed box of Fig 5.9 is

called an analog-to-digital converter, usually abbreviated A/D converter. In

commercially available A/D converters there is normally no sharp distinction

between that portions of the electronic circuitry used to do the quantizing and

that portion used to accomplish the encoding.

In summery the A/D converter accepts an analog signal and replaces it with a

succession of code symbols, each symbol consisting of a train of pulses in which

each pulse may be interpreted as the representation of a digit in an arithmetic

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system. Thus the signal transmitted over the communication channel in a PCM

system is referred to as a digitally encoded signal.

The decoder, also called a digital-to-analog (D/A) converter, performs the inverse

operation of the encoder. The decoder output is sequence of quantized multilevel

sample pulses. The quantized PAM signal is now reconstructed.

Bandwidth of PCM: Suppose that in binary PCM, M quantizing levels are used,

satisfying H = 2I ,P = Q-R= , where N is an integer. For this case, P = Q-R= pulses

must be transmitted for each sample of the message signal. If the message

bandwidth is and the sampling rate is  =2 , then S
) binary pulses must be

transmitted per second.

Assuming the PCM signal is low-pass signal of bandwidth 7TU , the required

minimum sampling is 27TU . Thus,

27TU = P 

Or
P
7TU =  =P
2

This equation shows that the minimum required bandwidth for PCM is proportional

to the message signal bandwidth and the number of bits per symbol. Note that the

actual bandwidth required for a PCM system depends on the PCM representation.

5.5.2 Delta Modulation (DM)

A method for converting analog signals to a string of binary digits that requires

much simpler circuitry than PCM is called delta modulation. In DM, an incoming

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message signal is oversampled (i.e., at a rate much higher than the Nyquist rate)

to purposely increase the correlation b/n adjacent samples of the signal. This is

done to permit the use of a simple quantizing strategy for constructing the

encoded signal.

In its basic form, DM provides a staircase approximation to be the oversampled

version of the message signal, as illustrated in Fig 5.10a.The difference b/n the

Figure 5.10 Illustration of Delta Modulation

Input and the approximation is quantized into only two levels, namely,±∆,

corresponding to +ve and –ve differences. Fig 5.10a illustrates the way in which the

staircase approximation $G ()follows variations in the input signal m(t) and Fig

5.10b displays the corresponding binary sequence at the delta modulator output.It

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is apparent that in a delta modulation system the rate of information transmission

is simply equal to the sampling rate  = 1 .



Figure 5.11 DM system

The principal virtue of delta modulation is its simplicity. It may be generated by

applying the sampled version of the incoming message signal to a modulator that

involves a comparator, quantizer, and accumulator represents a unit delay, i.e., a

delay equal to one sampling period. The comparator computes the difference b/n

its two inputs. The quantizer consists of a hard limiter with an input-output

relation that is a scaled version of the signum function. The quantizer output is

then applied to an accumulator.

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5.5.3 Phase Shift Keying (BPSK)

When it becomes necessary, for the purpose of communication, to superimpose a

binary waveform on a carrier, amplitude modulation (AM), phase modulation (PM),

or frequency modulation (FM) may be used.

a. Binary Phase Shift Keying (BPSK)

In BPSK the transmitted signal is sinusoid of fixed amplitude. It has one fixed

phase when the data is at one level and when the data is at the other level the
"
phase is different by 180X . If the signal is of amplitude A it has a power Y = = Z= so

that Z = [2Y . Thus the transmitted signal is either

\]T^_ () = [2Y cos (` )

Or

\]T^_ () = [2Y cos(`  + )

= −[2Y cos(` )

In BPSK the data b(t) is a stream of binary digits with voltage levels which, as a

matter of convenience, we take to be at +1V. When b(t)=1V we say it is at logic

level 1 and when b(t)=-1V we say it is at logic level 0. Hence \]T^_ () can be written,

with no loss of generality, as

\]T^_ () = a()[2Y cos (` )

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b. Differential Phase Shift Keying (DPSK)

DPSK is a modification of BPSK which has the merit that it eliminate the ambiguity

about whether the demodulated data is or is not inverted. In addition, DPSK avoids

the need to provide the synchronous carrier required at the demodulator for

detecting a BPSK signal.

A means of generating a DPSK signal is shown in Fig 5.12. The data stream to be

transmitted, d(t), is applied to one input of an exclusive-OR logic gate. To the

other gate input is applied the output of the excusive OR gate b(t) delayed by the

time b allocated to one bit. This second input is then a( − b ).

Figure 5.12 Means of generating a DPSK signal

In Fig 5.13 we have drawn logic waveforms to illustrate the response b(t) to an

input d(t). The upper level of the waveforms corresponds to logic 1,the lower level

to logic 0.The truth table for XOR gate is given in Fig 5.12and with this table we

can easily verify that the waveform for d(t), a( − b ), and b(t) are consistent with

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one another. We observe that, as required, a( − b ) is indeed b(t) delayed by one

bit time and that in any bit interval the bit b(t) is given by a() = ()cde a( − b ).

Figure 5.13 Logic waveforms to illustrate the response b(t) to an input d(t)

Because of the feedback involved in the system of Fig 5.13there is a difficulty in

determining the logic levels in the interval in which we start to draw the waveform

(interval 1 in Fig 5.13). we cannot determine b(t) in this first interval of our

waveform unless we know b(0). But we cannot determine b(0) unless we know both

d(0) and b(-1), etc. Thus, to justify any set of logic levels in an initial bit interval

we need to know the logic levels in the preceding interval. But such a determination

requires information about the interval two bit times earlier and so on. In the

waveform of Fig 5.13 we have circumvented the problem by arbitrarily assuming

that in the first interval b(0)=0.

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Example: How does the phase of carrier vary for message

f$()} = f1, 0, 1, 1, 0, 1, . . } in

a. BPSK

b. DPSK

Solution

a. Using BPSK equations , the phase of the carrier will be { 0, , 0,0, , 0, …}

b. In DPSK b(t)=d(t) XOR a( − b ), Now, the sampled version of d(t) is

input m(n). Hence the sampled version of b(t) can be written as

b(n)=m(n)XORb(n-1).

Considering, initial value of storage element=0, fa()} = f1, 1, 0, 1, 1, 0, . . }

Then the phase of the carrier will be { 0, 0, , 0,0, , …} and the carrier

amplitude {c (n)} = {+1, +1, -1, +1, +1,-1}

5.5.4 Binary Frequency Shift keying (BFSK)

In BFSK the binary data waveform d(t) generates a binary signal

\]i^_ () = [2Y cos (`  + ()Ω)

Here d(t)=+1 or -1 corresponding to the logic levels 1 and 0 of the data waveform.

The transmitted signal is of amplitude [2Y and is either

\]i^_ () = 1 () = [2Y cos (` + Ω)

or

\]i^_ () = 3 () = [2Y cos (` − Ω)

And thus has an angular frequency ` + Ω or ` − Ω with Ωa constant offset from

the nominal carrier frequency ` . We shall call the higher frequency `1 (= ` + Ω )

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and the lower frequency `3 (= ` − Ω ). We may conceive that the BFSK signal is

generated in the manner indicated in Fig 5.14.Two balanced modulators are used,

one with carrier `1 and one with carrier `3 .The voltage values of Y1 () and of

Y3 () are related to the voltage values of d(t) in the following manner.

Thus when d(t) changes from +1 to -1 Y1 changes from 1 to 0 and Y3 from 0 to 1. At

any time either Y1 or Y3 is 1 but not both so that the generated signal is either at

angular frequency `1 or `3 .

Figure 5.14 A representation of a manner in which a BFSK signal can be generated

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