PCS M4

BMS INSTITUTE OF TECHNOLOGY AND MANAGEMENT
Doddaballapur main road, Avalahalli, Yelahanka, Bangalore - 560064
Department of Electronics and Telecommunication Engineering
Principles of Communication Systems

Module - 4:
SAMPLING AND QUANTIZATION
Presented by
MALLIKARJUNA GOWDA C P
Associate Professor, Dept. of ETE
BMS Institute of Technology and Management
23-11-2020
1
12/17/2020 MG_PCS_18EC53_Dept_ETE@BMSIT&M
INTRODUCTION
 Amplitude and angle forms of CW modulation were
developed in the early 1900s with analog sources, voice, in
mind.
 Digital transmission from the 1930s to the 1960s, the
recognition that digital transmission has many advantages
over the transmission of analog information.
 Developments in solid state electronics, micro-electronics
and large-scale integration in the 1970s, the right capabilities
existed to take advantage of digital transmission in an
efficient and economical fashion.
MG_PCS_18EC53_Dept_ETE@BMSIT&M 12/17/2020 2
 Analog to digital transmission: conversion of common
information sources, such as voice and music, to digital
representation.
 Analog to digital conversion and the representation of the
analog information as a sequence of pulses.
 Analog to digital: An analog source is sampled at discrete
times. The resulting analog samples are then transmitted by
means of analog pulse modulation.
 Quantization: an analog source is not only sampled at discrete times
but the samples themselves are also quantized to discrete levels.
 Two methods of digitally representing an analog source:
pulse-code modulation and delta modulation.
 The conversion from an analog information source, such as voice or
video, to a digital representation and subsequent transmission, was
often implemented as a single step.
 For example, pulse-code modulation describes both a method of
digitally representing an analog source and a method for transmitting
that information over a baseband channel.
WHY DIGITIZE ANALOG SOURCES?
Advantages of the transmission of digital information:
 Digital systems are less sensitive to noise than analog. For long
transmission lengths, the signal may be regenerated effectively error-
free at different points along the path, and the original signal
transmitted over the remaining length.
 It is easier to integrate different services, for example, video and the
accompanying soundtrack, into the same transmission scheme.
 The transmission scheme can be relatively independent of the source.
For example, a digital transmission scheme that transmits voice at 10
kbps could also be used to transmit computer data at 10 kbps.
WHY DIGITIZE ANALOG SOURCES?
 Circuitry for handling digital signals is easier to repeat and

digital circuits are less sensitive to physical effects such as
vibration and temperature.
 Digital signals are simpler to characterize and typically do not
have the same amplitude range and variability as analog
signals. This makes the associated hardware easier to design.
All media (e.g., cable, radio waves, optical fiber) may be used for either
analog or digital transmission, digital techniques offer strategies for more
efficient use of those media:
 Various media sharing strategies, known as multiplexing techniques,
are more easily implemented with digital transmission strategies.
 There are techniques for removing redundancy from a digital
transmission, so as to minimize the amount of information that has to
be transmitted.
 There are techniques for adding controlled redundancy to a digital
transmission, such that errors that occur during transmission may be
corrected at the receiver without any additional information.
 Example, a forward error correcting technique that is relatively simple
by today’s standards can reduce an error rate of 7 percent at the
channel output to as little as 0.001 percent at the decoder output.
 Digital techniques make it easier to specify complex standards that
may be shared on a worldwide basis.
 This allows the development of communication components with
many different features (e.g., a cellular handset) and their
interoperation with a different component (e.g., a base station)
produced by a different manufacturer.
 Channel compensations techniques, such as equalization, especially
adaptive versions, are easier to implement with digital transmission
techniques.
The majority of these advantages for digital transmission rely on the
availability of low-cost microelectronics. This counterbalances the
original advantage for analog transmission of transporting a large
amount of information in a very simple fashion.
THE SAMPLING PROCESS
 The Fourier transform of a periodic signal with period 𝑻𝟎 consists of an

infinite sequence of delta functions occurring at integer multiples of the
fundamental frequency 𝒇𝟎 = 𝟏ൗ𝑻𝟎 .
 A signal periodic in the time domain has the effect of sampling the
spectrum of the signal in the frequency domain.
 Sampling a signal in the time domain has the effect of making the
spectrum of the signal periodic in the frequency domain.
 The sampling process is described in the time domain.
 It is an operation that is basic to digital signal processing and digital
communications.
 Through use of the sampling process, an analog signal is converted into a
corresponding sequence of samples that are usually spaced uniformly in
time.
choose the sampling rate properly, so that the
sequence of samples uniquely defines the original
analog signal.
The derivation of the sampling theorem, as described herein,
is based on the assumption that the signal g(t) is strictly
band limited.
An information-bearing signal is not strictly band
limited, with the result that some degree of undersampling
is encountered.
 Aliasing is produced by the sampling process.
 Aliasing refers to the phenomenon of a high frequency
component in the spectrum of the signal seemingly taking
on the identity of a lower frequency in the spectrum of its
sampled version, as illustrated in Figure 7.3.
The aliased spectrum shown by the solid curve in Figure
7.3b pertains to an “undersampled” version of the message
signal represented by the spectrum of Figure 7.3a.
To combat the effects of aliasing in practice, we may use two corrective
measures, as described here:
1. Prior to sampling, a low-pass pre-alias filter is used to attenuate
those high-frequency components of the signal that are not
essential to the information being conveyed by the signal.
2. The filtered signal is sampled at a rate slightly higher than the
Nyquist rate.
 The use of a sampling rate higher than the Nyquist rate also has the
beneficial effect of easing the design of the reconstruction filter used to
recover the original signal from its sampled version.
 Consider the example of a message signal that has been pre-alias
(lowpass) filtered, resulting in the spectrum shown in Figure 7.4a.
 The corresponding spectrum of the instantaneously sampled version of the
signal is shown in Figure 7.4b, assuming a sampling rate higher than the
Nyquist rate.
According to Figure 7.4b, the design of the reconstruction
filter may be specified as follows (see Figure 7.4c):
 The reconstruction filter is low-pass with a passband
extending from -W to W, which is itself determined by the
pre-alias filter.
 The filter has a transition band extending (for positive
frequencies) from W to fs - W, where fs is the sampling rate.
The fact that the reconstruction filter has a well-defined
transition band means that it is physically realizable.
This is to be compared to the implementation of the ideal
reconstruction filter corresponding to sinc(2Wt) that would
be necessary if the signal was not oversampled.
PULSE-AMPLITUDE MODULATION(PAM)
 Pulse-amplitude modulation, is the simplest and most
basic form of analog pulse modulation.
 In 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: The message signal is
multiplied by a periodic train of rectangular pulses.
 In natural sampling the top of each modulated
rectangular pulse varies with the message signal,
whereas in PAM it is maintained flat;
 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.
 However, care has to be exercised in how long we
make the sample duration T, as the following analysis
reveals.
Some final remarks are in order:
 The transmission of a PAM signal imposes rather stringent
requirements on the amplitude and phase responses of the
channel, because of the relatively short duration of the
transmitted pulses.
 Furthermore, the noise performance of a PAM system can
never be better than baseband-signal transmission.
Accordingly, we find that for transmission over long distances,
PAM would be used only as a means of message processing for
time-division multiplexing, from which conversion to some
other form of pulse modulation is subsequently made.
TIME-DIVISION MULTIPLEXING(TDM)
 The sampling theorem provides the basis for transmitting the
information contained in a band-limited message signal m(t)
as a sequence of samples of m(t) taken uniformly at a rate that
is usually slightly higher than the Nyquist rate.
 An important feature of the sampling process is a
conservation of time.
 The transmission of the message samples engages the
communication channel for only a fraction of the sampling
interval on a periodic basis, and in this way some of the time
interval between adjacent samples is cleared for use by other
independent message sources on a time-shared basis.
 A time-division multiplex (TDM) system, enables the joint
utilization of a common communication channel by a
plurality of independent message sources without mutual
interference among them.
 Each input message signal is first restricted in
bandwidth by a low-pass pre-alias filter to remove the
frequencies that are nonessential to an adequate signal
representation.
The low-pass filter outputs are then applied to a
commutator, which is usually implemented using
electronic switching circuitry.
The function of the commutator is twofold:
(1) To take a narrow sample of each of the N input
messages at a rate 𝒇𝒔 that is slightly higher than 2W, where
W is the cutoff frequency of the pre-alias filter.
(2) To sequentially interleave these N samples inside the
sampling interval 𝑻𝒔 .
Indeed, this latter function is the essence of the time-
division multiplexing operation.
 Following the commutation process, the multiplexed
signal is applied to a pulse modulator, the purpose of
which is to transform the multiplexed signal into a
form suitable for transmission over the common
channel.
 It is clear that the use of time-division multiplexing
introduces a bandwidth expansion factor N, because
the scheme must squeeze N samples derived from N
independent message sources into a time slot equal
to one sampling interval.
 At the receiving end of the system, the received signal
is applied to 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 a decommutator, which operates in
synchronism with the commutator in the
transmitter.
 This synchronization is essential for a satisfactory
operation of the system.
 The way this synchronization is implemented
depends naturally on the method of pulse
modulation used to transmit the multiplexed
sequence of samples.
PULSE-POSITION MODULATION(PPM)
 In a pulse modulation system, we may use the increased bandwidth
consumed by pulses to obtain an improvement in noise performance
by representing the sample values of the message signal by some
property of the pulse other than amplitude.
 In pulse-duration modulation (PDM ),or pulse-width modulation

(PWM) or pulse-length modulation: The samples of the message
signal are used to vary the duration of the individual pulses.
 The modulating signal may vary the time of occurrence of the leading
edge, the trailing edge, or both edges of the pulse.
 In PDM, long pulses expend considerable power
during the pulse while bearing no additional
information.
 If this unused power is subtracted from PDM, so
that only time transitions are preserved, we obtain a
more efficient type of pulse modulation known as
pulse-position modulation (PPM).
In PPM, the position of a pulse relative to its
unmodulated time of occurrence is varied in
accordance with the message signal.
GENERATION OF PULSE-POSITION MODULATION(PPM) WAVES:
Sample and Hold circuit
 The message signal m(t) is first converted into a
PAM signal by means of a sample-and-hold circuit,
generating a staircase waveform u(t).
 The pulse duration T of the sample-and-hold
circuit is the same as the sampling duration Ts.
 The signal u(t) is added to a sawtooth wave, yielding
the combined signal v(t).
 The combined signal v(t) is applied to a threshold
detector that produces a very narrow pulse
(approximating an impulse) each time v(t) crosses
zero in the negative-going direction.
 The resulting sequence of “impulses” i(t) is shown.
 The PPM signal s(t) is generated by using this
sequence of impulses to excite a filter whose
impulse response is defined by the standard pulse
g(t).
PPM Waveforms
DETECTION OF PPM WAVES:
 Convert the received PPM wave into a PDM wave

with the same modulation.
 Integrate this PDM wave using a device with a finite
integration time, thereby computing the area under
each pulse of the PDM wave.
 Sample the output of the integrator at a uniform
rate to produce a PAM wave, whose pulse
amplitudes are proportional to the signal samples
m(nTs) of the original PPM wave s(t).
 Demodulate the PAM wave to recover the message
signal m(t).
 A practical PPM receiver includes a nonlinear
device called a slicer at its input end.
 The input-output characteristic of an ideal slicer
is shown in Figure 7.13, where the slicing level is
normally set at approximately half the peak pulse
amplitude of the received PPM wave.

The function of the slicer is to preserve the positions
of the edges of the received pulses (as modified by
noise) and remove everything else.
 It does so by producing almost “rectangular” pulses
with fairly sharp leading and trailing edges at the
same instants as the corresponding edges of the
received pulses.
 The slicer acts as a “noise cleaning device” in that the
final noise level at the output of the receiver is greatly
reduced by eliminating all the noise in the received
PPM wave except in the neighborhood of the leading
and trailing edges.
• The output of the slicer is differentiated and then
half-wave rectified, yielding a very short pulse
(approximating an impulse) each time the
amplitude of a pulse in the received PPM wave
passes through the slicing level.
• Having converted the received (noisy) PPM wave
into a PDM wave with the same modulation, the
receiver then proceeds to reconstruct the original
baseband signal m(t) in the manner described above.
NOISE IN PULSE-POSITION MODULATION
• In a PPM system, the transmitted information is contained in the
relative positions of the modulated pulses.
• The presence of additive noise affects the performance of such a
system by falsifying the time at which the modulated pulses are
judged to occur.
• Immunity to noise can be established by making the pulse build
up so rapidly that the time interval during which noise can exert
any perturbation is very short.
• Indeed, additive noise would have no effect on the pulse
positions if the received pulses were perfectly rectangular,
because the presence of noise introduces only vertical
perturbations.
• However, the reception of perfectly rectangular pulses implies
an infinite channel bandwidth, which is of course impractical.
• Thus, with a finite channel bandwidth in practice, we
find that the received pulses have a finite rise time,
and so the performance of the PPM receiver is
affected by noise.
• As with a CW modulation system, the noise
performance of a PPM system may be described in
terms of the output signal-to-noise ratio.
• Also, to find the noise improvement produced by PPM
over baseband transmission of a message signal, we
may use the figure of merit defined as the output
signal-to-noise ratio of the PPM system divided by
the channel signal-to-noise ratio.

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BMS INSTITUTE OF TECHNOLOGY AND MANAGEMENT

Doddaballapur main road, Avalahalli, Yelahanka, Bangalore - 560064

Department of Electronics and Telecommunication Engineering

Principles of Communication Systems

Associate Professor, Dept. of ETE

BMS Institute of Technology and Management

Advantages of the transmission of digital information:

 Circuitry for handling digital signals is easier to repeat and

 The Fourier transform of a periodic signal with period 𝑻𝟎 consists of an

by representing the sample values of the message signal by some

property of the pulse other than amplitude.

 In pulse-duration modulation (PDM ),or pulse-width modulation

signal are used to vary the duration of the individual pulses.

 Convert the received PPM wave into a PDM wave

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