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Adama University

School of Engineering and Information Technologies

Department of Electrical Engineering
EEng-3210: Introduction to Communication systems
Chapter 3- Pulse Communication

3.1. Introduction

You have undoubtedly drawn graphs of continuous curves many times during your education.
To do that, you took data at some finite number of discrete points, plotted each point, and
then drew the curve. Drawing the curve may have resulted in a very accurate replica of the
desired function even though you did not look at every possible point. In effect, you took
samples and guessed where the curve went in between the samples. If the samples had
sufficiently close spacing, the result is adequately described. It is possible to apply this line
of thought to the transmission of an electrical signal, that is, to transmit only the samples and
let the receiver reconstruct the total signal with a high degree of accuracy. This is termed
pulse communication.

The Sample Frequency:

Using the concept described shortly, a signal can be fully reconstructed at the receiver to a
high degree of accuracy from its samples if it is sampled at twice the rate of its highest
significant frequency component. Stated inversely, a given bandwidth can carry pulse signals
of half its high frequency cut-off. This is known as the Nyquist rate and it is one of the most
critical specifications in a pulse modulated or digital modulated (PCM) system sampling
frequency selection. The Nyquist rate states that the sample frequency (fs) must be at least
twice the highest input frequency (fin).

f s  2 f in

If the sampling criterion is not met, the original analog signal frequency is lost and an alias
frequency is produced instead. The frequency of the alias signal is given by

f alias  f in  f s

This is shown in Fig. 3.4, where the sampling rate is 2/3 times the 1KHz signal rate. The
output signal in fig. 3.1 is at 1000 - 667 Hz , or 333 Hz, and will bear no resemblance to the
original signal.

Fig. 3.1: Generation of alias frequency.

In the case of voice transmission the standard sampling rate is 8 KHz, it being just slightly
more than twice the highest significant frequency component. This implies a pulse rate of
8KHz or 125microsecond period. Since a pulse duration of 1 microsecond may be adequate,
it is easy to see that a number of different message could be multiplexed (TDM) on the
channel (as shown in fig. 3.2), or alternatively it would allow a high peak transmitted power
with a much lower (1/125) average power. The high peak power can provide a very high
signal to noise ratio or a greater transmission range.

Fig. 3.2 illustrates an important feature of pulse communication - time division multiplexing
(TDM) system which enables joint utilization of a common transmission channel.

Fig. 3.2: Block diagram of TDM system.

At the transmitter, each input signal is first restricted by a low pass anti-aliasing filter to
remove the frequencies which are nonessential to an adequate signal representation. The low
pass filter output is then applied to a commutator (sampler) that takes a narrow sample of
each of the N input signals to be transmitted. The sampler illustrated is a rotating machine
making periodic brush contact with each signal. A similar rotating machine at the receiver is
used to distribute the N separate signals, and it must be synchronized to the transmitter. A
mechanical sampling system such as this may be suitable for low sampling rates such as
encountered in some telemetry systems but would not be adequate for the 8KHz rate required
for voice transmissions. In that case an electronic switching system would be incorporated.

Following the commutation process, the multiplexed signal is applied to a pulse modulator
(to be discussed latter) which transforms the multiplexed signal into a dorm suitable for
transmission over the channel. At the receiving end of the system, the received signal is
applied a pulse demodulator, which performs the reverse operation of the pulse modulator.
The narrow samples produced at the pulse demodulator output are distributed to the
appropriate low pass reconstruction filters by means of decommutator, which operates in
synchronism with the commutator in the transmitter.

It should be careful to realize that a price must be paid for system gains obtained by pulse
modulation schemes. More important than the greater equipment complexity is the
requirement for greater channel (bandwidth) size. If a maximum 3 KHz signal directly
amplitude modulates a carrier, a 6 KHz bandwidth is required. If a 1 microsecond pulse does
the modulating, just allowing its fundamental component of 1/1 micro second or 1MHz to do
the modulating means a 2MHz bandwidth is required in AM. In spite of the large bandwidth
required, TDM is still preferable (if not the only possible way) to using 100 different

transmitters, antennas or transmission lines, and receivers in cases where large numbers of
messages must be conveyed simultaneously.

3.2. Pulse Analog Modulation

Now we have understood the essence of sampling process, we are ready to define pulse
modulation. In its strictest sense, pulse modulation is not modulation but rather a message
processing technique. The message to be transmitted is sampled by the pulse, and the pulse is
subsequently used to either amplitude or frequency modulate the carrier. There are three
basic forms of pulse analog modulation as illustrated in Fig. 3.3. The three types we shall
consider here are usually termed pulse-amplitude modulation (PAM), pulse width modulation
(PWM), and pulse position modulation (PPM).

For the sake of clarity, the illustration of these modulation schemes has greatly exaggerated
the pulse widths. Since a major application of pulse modulation occurs when TDM is to be
used, shorter pulse durations, leaving room for more multiplexed signals, are obviously
desirable. As shown in Fig. 3.3, the pulse parameter that is varied in step with the analog
signal is varied in direct step with the signal's value at each sampling interval. Notice that the
pulse amplitude in PAM and pulse width in PWM are not zero when the signal is minimum.
This is done to allow a constant pulse rate and is important in maintaining synchronization in
TDM systems.

Fig. 3.3: Types of pulse analog modulation

3.2.1. Pulse Amplitude Modulation (PAM)

In pulse-amplitude modulation, the pulse amplitude is made proportional to the modulating

signal’s amplitude. This is the simplest pulse modulation to create in that a simple sampling
of the modulating signal at a periodic rate can be used to generate the pulses, which are
subsequently used to modulate a high-frequency carrier.

While PAM finds some use due to its simplicity, PWM and PPM use constant amplitude
pulses and provide superior noise performance. The PWM and PPM systems fall into a
general category termed pulse time modulation (PTM) since their timing, and not amplitude,
is the varied parameter.

There are two basic sampling techniques used to create a PAM signal. The first is called
natural sampling. Natural sampling is when the tops of the sampled waveform (the sampled
analog input signal) retain their natural shape. An example of natural sampling is shown in
Fig. 3.4(a). Notice that one side of the analog switch is connected to ground. When the gate
is asserted the JFET will short the signal to ground, but it will pass the unaltered signal to the
output when the gate is not asserted. Note too that there is no hold capacitor present in the

Fig. 3.4: (a) Natural sampling (b) flat-top sampling

Probably the most popular type of sampling used in PAM system is called flat-top sampling.
In flat-top sampling the sample signal voltage is held constant between samples. The method
of sampling creates a staircase that tracks the changing input signal. This method is popular
because it provides a constant voltage during a window of time for the binary conversion of
the input signal to be completed. An example of flat-top sampling is shown in Fig. 3.4(b).
With flat-top sampling the analog switch connects the input signal to the hold capacitor.

3.2.2. Pulse Width Modulation (PWM)

Pule-width modulation (PWM), a form of PTM, is also known as pulse-duration modulation

(PDM) and pulse-length modulation (PLM). In pulse width modulation, samples of the
message signal are used to vary the duration of the individual pulses.

A simple means of PWM generation is provided in Fig. 3.5. A simple comparator with a
sawtooth carrier can turn a sinusoidal command into a pulse-width modulated output. In
general, the larger the command signal, the wider the pulse. Command (modulating) signal
output stays high as long as the command is greater than the carrier. To obtain the modulating
signal from the PAM wave, integrator is used.

Fig. 3.5: PWM signal generation waveforms

3.2.3. Pulse-Position Modulation (PPM)

In pulse position modulation, the position of the pulse relative to its unmodulated time of
occurrence is varied in accordamce with the message signal. PWM and pulse position
modulation (PPM) are very similar, a fact that is underscored in Fig. 3.6, which shows PPM
being generated from PWM.

Fig. 3.6: PPM signal generation

Since PPM has superior noise characteristics, it turns out that the major use for PWM is to
generate PPM. By inverting the PWM pulses in fig. 3.6 and then differentiating them, the
positive and negative spikes shown are created. By applying them to a Schmitt trigger
sensitive to only positive levels, a constant amplitude and constant pulse width signal is
formed. However, the position of these pulses is variable and now proportional to the
original modulating signal, and the desired PPM signal has been generated. The information
content is not contained in either the pulse amplitude or width as in PAM and PWM, which
means the signal now has a greater resistance to any error caused by noise. In addition, when
PPM modulation is used to amplitude-modulate a carrier, a power savings results since the
pulse width can be made very small unlike the case for PWM, where long pulses expend
considerable power while bearing no additional information,

At the receiver, the detected PPM pulses are usually converted to PWM first and then
converted to the original analog signal by integrating as previously described. Conversion
from PPM to PWM can be accomplished by feeding the PPM signal into the base of one
transistor in a flip-flop. The other base is fed from synchronizing pulses at the original
(transmitter) sampling rate. The period of time that the PPM-fed transistor's collector is low
depends on the difference in the two inputs, and it is therefore the desired PWM signal.

3.3. Pulse Digital Modulation

3.3.1. Pulse-code Modulation:

Pulse-code modulation (PCM) is the most common technique used today in digital
communications for representing an analog signal by a digital word. It is used in many
applications, such as your telephone system, digital audio recording (DAT or digital audio
tape), CD laser disks, digitized video special effects, voice mail, and many other applications.
PCM techniques and applications are a primary building block for many of today's advanced
communications systems.

The basic operations performed in PCM system are:

 Sampling
 Quantizing
 Encoding

In sampling operation, the analog signal is input into a sample and hold circuit generating a
flat-top PAM signal resulting in a fixed voltage level. To ensure perfect reconstruction of the
message signal at the receiver, the sampling rate must be greater than twice the highest
frequency of the message signal. In practice, a low-pass pre-alias filter is used at the front end
of the sampler in order to exclude frequencies greater than the maximum frequency befor
sampling. The output of a sampler is still continuous in amplitude and can take on any value.
The number of possible values of the samples is infinite. To transmit as a digital signal we
must restrict the number of possible values. This is accomplished by quantizing.

Quantizing operation approximates the analog values by using a finite number of levels. It
makes the signal discrete in amplitude by rounding off to one of discrete levels as indicated in
fig. 3.7. This means that each of the quantized PAM signal is restricted to a number of
discrete magnitudes. We may choose to transmit these quantized sample values directly.
Alternatively, we may represent each quantized level by a code number rather than the
sample value itself. This is called encoding. Most of the time, the code number is converted
to its binary representation. If digits of the binary representation of the code number are
transmitted as pulses, the system of transmission is called pulse-code modulation (PCM).

Thus pulse-code modulated signal is obtained from the quantized PAM signal by encoding
each quantized sample value into a digital word. This whole process results in analog signal
conversion to digital signal and it is called analog-to-digital conversion (ADC). The block
diagram of ADC and the process involved are indicated in fig. 3.7. As we have seen in the
block diagram of the PCM circuit (Fig. 3.7), the analog to digital converter (ADC) is used to
convert the information signal to digital format.




Fig. 3.7: Block diagram of PCM process

A block diagram of a PCM system (transmitter and receiver) is shown in fig. 3.8. The ADC
is shown in the transmitting section and the DAC in the receiver section. DAC in the receiver
side consist of decoder and reconstruction filter. The decoder regroups the received pulses
into a code word and maps back into a quantized PAM signal. Reconstruction is the final
operation in the PCM receiver to recover the analog signal. This is done by passing the
decoder output through a low pass reconstruction filter.

Parallel to clock
Analog Anti-aliasing
input filter ADC
l bits
clock bits Communication
Serial to
clock DAC

bits clock
Fig. 3.8 PCM communication system
Quantization Noise:

The use of quantization introduces an error defined as the difference between the input signal
and the output signal. The error is called quantization noise or quantization error. The

quantization error depends on the step size. Hence if the steps are uniform in size, small
amplitude signals will have a poorer signal-to-quantization noise ratio than large-amplitude
signals. To correct this situation within the constraint of a fixed number of levels, it is
advantageous to taper the step size so that the steps are close together at low signal
amplitudes and further at large amplitudes – non-uniform step size. Such variation of step
size yields a signal-to-noise ratio improvement for small signal, although strong signals will
be impaired. Quantizer that uses non-uniform steps is referred to as non-uniform quantizer.
The use of non-uniform quantizer results in fewer steps than that would be needed if uniform
quantizer were used.

While it is possible to build a quantizer with tapered steps, it is more feasible to achieve an
equivalent effect by distorting the signal before applying to the transmitting quantizer. This
distortion involves compressing large amplitude signals. An inverse distortion (expanding
large amplitude signals) is introduced at the receiving end so that the overall transmission is
distortionless. The compression process is accomplished by a network called compressor and
the inverse operation is performed by an expander. The combination of compressor and
expander is called compander, which then performs the operation of companding.
Companding operation have been discussed in chapter one. There it is meant to avoid
nonlinear distortion.

Particular forms of compression laws that are used in practice are the so-called μ-law and A-
law. (Read about μ-law and A-law).

Advantages of PCM:

 PCM signals derived from all types of analog sources may be merged with data
signals and transmitted over a common high-speed digital communication system.
 In long-distance digital telephone systems requiring repeaters, a clean PCM waveform
can be regenerated at the output of each repeater, where the input consists of a noisy
PCM waveform.
 The probability of error for the system output can be reduced even further by the use
of appropriate coding techniques.
 Secure communication through the use of special modulation schemes or encryption.

These advantages are attained at the cost of increased system complexity and increased
channel bandwidth. However, the requisite device technology for the implementation of a
PCM system is available cost-efficiently as VLSI chips. The issue of bandwidth is not of a
real concern as it was used to be due to availability of wideband communication channels
from the deployment of communication satellites for broadcasting and the ever increasing use
of fiber optics for networking. Sophisticated data compression techniques can be used to
reduce redundancy inherently present in PCM signal thereby reducing the bit rate of
transmitted data without serious degradation in system performance. In effect, increased
processing complexity resulting in increased cost of implementation traded off for reduced bit
rate and therefore reduced bandwidth requirement.

3.3.2. Differential Pulse Code Modulation (DPCM)

In PCM, each sample of the baseband signal is encoded into a series of binary digits
independently of all others. However, voice or video signals sampled at the Nyquisit rate or
faster exhibit significant correlation between the successive samples. In other words, the
change in amplitude of successive samples is relatively small. When these highly correlated
samples are encoded using standard PCM system, the resulting encoded signal contains
redundant information and results in lower date rate from source to output. By removing this
redundancy before encoding we obtain a more efficient coded signal. A relatively simple

solution is to encode the difference between successive samples rather than the samples
themselves. Since the difference between the successive samples is expected to be smaller
than the actual sample amplitude, fewer bits are required to represent the differences. Further
if a sufficient part of a redundant signal is known, it is possible to make the most probable
estimate of the rest. Specifically if we know the past behaviour of a signal up to a certain
point of time, it is possible to make some inference about its future values based on the
previous values. Such a process is known as prediction. The possibility of prediction of future
values of a signal is the basis of differential pulse code modulation (DPCM) scheme which
involves encoding of the difference of successive sample values.

Consider a signal sampled at a rate to produce a sequence of correlated

samples . In the scheme in fig. 3.9, the input to the quantizer is a signal which
is the difference between the unquantized sample and a prediction of it denoted
by .


The predicted signal is produced by the prediction filter whose input consists of a quantized
version of the input sample. The difference sequence is called prediction
error since it corresponds to the amount of which the prediction filter fails to predict the input
exactly. The quantizer output is encoded to get a variation of standard PCM, known as
differential pulse code modulation (DPCM). The encoded signal is subsequently used for

. . Quantizer . Encoder wave


Fig. 3.9 DPCM system transmitter

The quantizer output can be represented mathematically as


where is the quantization error. According to the block diagram of the DPCM system
in fig. 3.9, the quantizer output is added to the predicted value to produce
prediction filter input which is given by


Substituting the value of from equation (3.2) into equation (3.3) and the value of
from equation (3.1) into equation (3.3),


Equation (3.4) represents the quantized version of the sample . It may therefore be
concluded that irrespective of the properties of the prediction filter, the quantized sample
at the prediction filter input differs from the sample of the original signal
by the quantization error .
A typical arrangement for the reconstruction of the quantized version of the input is shown in
fig. 3.10. It consists of a decoder which error signal. The quantized version of the original
input is reconstructed from the decoder output by using the same prediction filter used in the
transmitter of fig. 3.9. In the absence of channel noise, the encoded signal at the receiver
input is identical to the encoded output at the transmitter. Accordingly, the receiver output is
which differs from the original input only by quantization error
incurred as a result of quantizing the prediction error . It is therefore observed that in a
noise free environment, the prediction filters in the transmitter and receiver operate on the
same sequence of samples . It is with this purpose in mind that the feedback path is
added to the transmitter in the fig. 3.9.

Decoder . Output


Fig. 3.9 DPCM system receiver

3.3.3. Delta Modulation (DM)

In delta modulation analog input is sampled at a rate much higher than the Nyquisite rate i.e.;
oversampled to purposely increase the correlation between adjacent samples of the signal.
This is done to permit the use of a simple quantizing strategy for constructing encoded signal.
In its basic form DM approximates the input message signal by staircase function that moves
up or down by one quantization level (step size) ( ) at each sampling interval as illustrated in
fig. 3.11. Step size and sampling rate are two important parameters that must be chosen in
such a way that the staircase signal is a close approximation of the actual analog waveform.
The staircase approximation which follows the variation of the input signal is
illustrated in fig. 3.11. The corresponding binary sequence at the delta modulator output is
also indicated.
The difference between the input and staircase approximation is quantized into only two
levels, . If the staircase function is less than the analog signal, function goes up by one
step and 1 is generated. If the staircase function is greater than the analog signal, the function
goes down by one step and 0 is generated. Thus DM can be viewed as a simplified form of
DPCM in which two-level (1-bit) quantizer is used. Note that the quantised signal must
change at each sampling point.
Once the quantisation operation is performed, transmission of the signal can be achieved by
sending a one for a positive transition, and a zero for a negative transition, i.e.; one bit
encoding. For the case in the fig. 3.11, the transmitted bit train would be 001010111101000.

Fig. 3.11 Example of delta modulation

Let denote the input signal, and denote its staircase approximation.
Mathematically, delta modulation can be represented by the following discrete time



Where is an error signal representing the difference between the present sample value
of the input signal and the latest approximation to it, that is .
is the quantized version of at the output of the quantizer and it is the desired
delta modulation wave. sgn(.) is the signum function that gives the sign of its argument.
Delta modulated wave can be generated by applying the sampled version of the incoming
message signal to a modulator that involves a summer, a quantizer, an accumulator and an
encoder interconnected in a manner shown in fig. 3.12. The interconnection of the modulator
directly follows from mathematical representation of delta modulation (equations 3.5 to 3.8).
The accumulator increments the approximation by a step of in positive or negative
direction, depending on the sign of the error signal at each sampling instant . That
is if the input signal is greater than the recent approximation ,
a increment is applied to the approximation. On the other hand , if the input is smaller
than the approximation, a – is applied. Thus the accumulator can track the input samples by
one step ( ) at a time.

One-bit Encoder DM
. + . Quantize . wave
_ r


Fig. 3.12 Transmitter for Delta Modulation

The block diagram arrangement of the receiver of delta modulation system is shown in fig.
3.13. In the receiver unit the staircase approximation can be reconstructed by passing
the sequence of bits produced at the decoder output through an accumulator in a manner used
at the transmitter. The out-of-band quantizing noise in the high frequency staircase waveform
is rejected by passing it through a low pass filter having a bandwidth equal to that of the
message signal.
DM Accumulator
Decoder Output
wave . LPF


Fig. 3.13 Receiver for Delta Modulation

Quantization errors in Delta Modulation

As shown in fig. 3.14, if the step size is too small then the staircase approximation cannot
follow the fast variations in the analog input waveform. This causes the error in the staircase
approximation called slope overload noise. On the other hand, if the step size is too large
then the staircase approximation oscillates around the relatively flat segments of input
waveform resulting in error in the approximation called granular noise which can be viewed
to be analogous to quantizing noise in PCM system. We thus note that should be large so as
accommodate a wide dynamic range and it should be small for accurate representation of a
relatively low level signal.

Fig. 3.14 Slope overload distortion and granular noise in delta modulation

One can play with the two important parameters (step size and sampling rate) of delta
modulation to reduce slope overload and granular noise. Increasing sampling rate improves
accuracy. However, this increases the data rate. By adjusting the step size ( ) based on
window of past samples, the staircase approximation can be made to track the analog input
waveform more accurately. This type of delta modulation is called adaptive delta modulation.