Unit V

SAMPLING AND QUANTIZATION

INTRODUCTION
Computer plays a vital role in
communication which demands the need
to convert the analog information to
digital signal for processing and
transmission.
Pulse modulation is another important
classification of modulation, which is
fundamental to the digital transmission of
analog signal.
Pulse modulation is defined as the
 Analog Pulse Modulation: A periodic pulse
train is used as the carrier signal and some
characteristics of the pulse such as amplitude,
position or width is varied in accordancewith the
corresponding sampled message signal. The
information is transmitted in analog form but
the transmission takes place at discrete times
which requires only sampling. Pulse amplitude
modulation, pulse position modulation and
pulse width modulation are the examples of
analog pulse modulation.

 Digital Pulse Modulation: The message

signal is represented in discrete form both in
time and amplitude and transmitted in digital
form as a sequence of coded pulses. This type
of modulation involves the process of sampling,
LOW PASS SAMPLING
Sampling is the important operation in digital signal
processing and digital communication. It is defined as the
process of converting an analog signal into a sequence of
samples spaced uniformly in time.

The analog signal must be sampled in such a way that the

sampled sequences uniquely define the original analog signal.
So, it is important to choose the rate at which the sampling
must be performed.

This is the essence of sampling theorem. Low pass

sampling means sampling of low pass signal, whereas band
pass sampling means sampling of band pass signal.
Sampling theorem for band-limited signal

 A band-limited signal of finite energy, which

has no frequency components higher than W
Hz, may be completely described by specifying
the values of the signal at instants of time
separated by ½ W seconds.

 A band-limited signal of finite energy, which

has no frequency components higher than W
Hz, may be completely recovered from the
knowledge of its samples taken at the rate of
2W samples per second.

The minimum sampling rate of 2W samples

per second for a signal bandwidth of W
Hertz is called as Nyquist rate. The
reciprocal of Nyquist rate ½ W is called
Nyquist interval.
Aliasing & Its
Effect
Aliasing effect:
 The sampling theorem derivation is based on the
assumption that the signal is strictly band-limited.

 But in practice, an information bearing signal is not

strictly band-limited which results in under sampling.

 This results in signal distortion called as aliasing.

 The word “alias” means false or assumed identity.

 Aliasing or fold over refers to the phenomenon of a

high-frequency component in the spectrum of the
signal taking on the identity of a lower frequency in
the spectrum of its sampled version.
Effects of aliasing
1. Since high and low frequencies interfere
with each other, distortion is generated.
2. Data is lost and it cannot be recovered.

Different Ways to avoid aliasing

f s 2W

1. Using low pass anti aliasing filter (or)

pre-alias filter prior to sampling which
band limits the signal to W.
2. Sampling at a rate offs ≥ 2W
I)Sampling at a rate of fs ≥ 2W :
 When the sampling rate is made higher than 2W,
then the spectrums will not overlap and there will
be sufficient gap between the individual
spectrums.
Over Sampling
When the signal is sampled at a rate much
higher than Nyquist rate , it is called over
sampling. It is necessary to avoid aliasing error
in the signal. But it increases transmission
bandwidth.
II) : Using low pass anti aliasing filter (or) pre-
alias filter

y ' (t ) y (t ) y (t )
Band limiting
Sampler
LPF

• Hence a LPF is used before sampling the

signals. Thus the output of LPF is strictly
bandlimited and there are no frequency
components higher than “W” .Then there will
be no aliasing.
Nyquist rate & Nyquist interval.

Nyquist rate :
 When the sampling rate becomes exactly to 2W
samples per second for a signal bandwidth of W
Hertz is called as Nyquist rate.

Nyquist interval :

 Itis the time interval between any two adjacent

samples when sampling rate is Nyquist rate.

 Thereciprocal of Nyquist rate 1/2W is called

Nyquist interval
Reconstruction Filter (Interpolation filter)

The above equation provides an interpolation

formula for recontructing the original signal y(t) from
the sample values y(n/2W) with the sinc function
playing the role of an interpolation function.
 Each sample is multiplied by a delayed version of the
interpolation function and all the resulting waveforms
are added to obtain y(t) as shown in figure
The above equation also represents the response of an ideal low pass filter of
bandwidth W. So, the original signal y(t) can be recovered exactly from the
sequence of samples y(n/2W) by passing through an ideal low pass filter of
bandwidth W termed as reconstruction filter whose response is shown in
figure .
The passband of physically realizable filter
extends from –W to +W Hertz and has a non-
zero transition band from W to fs-W Hertz.The
region separating the passband and stopband is
called as guard band with the width of fs-2W
Hertz.
 Nonuniform Quantization
 Quantization using nonuniform or unequal step size is called as non-uniform
quantization or robust quantization as shown in figure 5.13.

Figure 5.13 Nonuniform Quantization

Need for nonuniform quantization:
 The uniform quantization provides better SNR for uniformly distributed signal.
But signals like speech have nonuniform distribution, where the large amplitude
levels are relatively rare compared to small amplitude levels. Therefore, with
uniform quantization, SNR becomes worse for low amplitude levels than for
high amplitude levels.
 Nonuniform quantization can provide fine quantization for frequently
occurring low amplitude levels and coarse quantization for rarely occurring
high amplitude levels. It improves the overall SNR by reducing the noise.
Companding:
 Non uniform quantization can be achieved by using a technique called
companding, which is a combination of compression at transmitter and
expander at receiver as shown in figure
 Compander = Compressor + Expander
Logarithmic Companding

The two commonly used logarithmic compression laws are,

1. µ-law companding:
The compressor characteristics is continuous.

In North America, the standard practical value for µ is 255. It is

used for PCM telephone systems in US, Canada and Japan.
Logarithmic Companding

2. A-law companding:
The compressor characteristics is piecewise.

The standard practical value for A is 87.56. It is used for PCM telephone
system in Europe.
 Analog Pulse
Modulation
 A periodic pulse train is used as the carrier signal and
some characteristics of the pulse such as amplitude, position
or width is varied in accordance with the corresponding
sampled message signal.

 The information is transmitted in analog form but the

transmission takes place at discrete times which requires only
sampling.

 Pulse amplitude modulation(PAM), pulse position

modulation(PPM) and pulse width modulation (PWM) are
the examples of analog pulse modulation.

 PULSE WIDTH MODULATION(PWM)
 In pulse width modulation, the samples of the message signal
are used to vary the width of the individual pulses keeping
amplitude constant.

 Thisform of modulation is also referred to as Pulse

Duration Modulation or Pulse Length Modulation.

 The modulating signal and periodic pulse carrier are shown

in figure 5.21 (a) and (b) respectively.

 Figure 5.21 (c) illustrates the PWM signal, where the trailing
edge of each pulse is varied in accordance with the message
signal.
Figure 5.21 (a) Modulating wave (b) Pulse carrier (c) PWM signal
 Generation of PWM signal:
 The block diagram of PWM signal generation using
comparator is shown in figure 5.22.
 The input message signal is fed to the non-inverting terminal
of the comparator and sawtooth signal is fed to the inverting
terminal of the comparator.
 The sawtooth signal acts as carrier signal.

The comparator generates PWM signal, if the maximum

of the input signal is less than sawtooth signal as shown in
figure 5.23.
 When the sawtooth signal rises with a fixed slope and crosses
input signal value, the inverting input of comparator is at
higher potential and the comparator output will be negative.

 The duration for which the comparator stays high depends on

the input signal amplitude and this decides the width of the
pulse.

 Thus,the pulse width generated is directly proportional to

the amplitude of the message signal at that instant.
Generation of PWM
Detection of PWM signal:
 The block diagram of PWM detection is shown in figure 5.24.
 The PWM signal received at the input of the detection circuit is
applied to synchronous pulse generator circuit and ramp generator.
 The synchronous pulse generator produces a train of constant
amplitude and constant width pulses.
 The ramp generator generates a constant slope ramp for the
duration of the pulse.
 The height of the ramp is proportional to the width of the PWM pulses.
 At the end of the pulse, a sample and hold amplifier retains the final
ramp voltage until it is reset at the end of the pulse.
PULSE POSITION
MODULATION (PPM)
In pulse position modulation, the
position of each pulse is varied
in accordance with the
amplitudes of the sampled
values of the modulating signal,
keeping the amplitude and
width of the pulses constant.
Figure 5.26 (a) Modulating signal (b) Pulse signal (c) PPM signal
 Generation of PPM signal:
 The PPM signal can be generated from the PWM.
 The PWM signal is fed to an inverter which reverses
polarity of the pulses.

• The inverted signal is then fed to a differentiator which

produces positive spikes, when the original PWM signal
transition is from HIGH to LOW and negative spikes,
when the transition is from LOW to HIGH as shown in
figure 5.28.
• These spikes are fed to a positive edge
triggered fixed width pulse generator
which generates pulses of fixed width
when a positive spike appears,
coinciding with the falling edge of
original PWM signal.

• The occurrences of these falling edges

depends on the message signal and
hence the delay in occurrence of these
fixed width pulses are proportional to the
amplitude of the input message at that
instant.
Figure 5.28 Principle of PPM signal generation
Detection of PPM signal:
The reference pulse is generated
by reference pulse generator of the
receiver with the synchronization
signal from the transmitter.
The SR edge triggered flip-flop is
set or turned ON by the positive
edge of the clock.
It remains set so that the output Q
is high, till a positive edge from
PPM signal resets it.
 PULSE CODE
MODULATION(PCM)
Pulse code modulation (PCM) is the most
basic form of digital pulse modulation.
In pulse code modulation, a message signal is
represented by a sequence of coded pulses
which is accomplished by representing the
signal in discrete form in both amplitude and
time.
The essential operations in the transmitter of a
PCM system are sampling, quantization and
encoding which are performed in the same
circuit called as analog to digital converter.
Regenerated PCM
Wave
I) Sampling:
 The incoming message signal is sampled with a
train of narrow rectangular pulses so as to
closely approximate the instantaneous sampling
process.

 The sampling rate must be greater than twice the

highest frequency component W of the message
signal in order to ensure the perfect reconstruction
of the message at receiver.

 In practice a low-pass pre-alias filter or anti-

aliasing filter is used before sampler to band-
limit the signal.
II) Quantization:
 The process of converting a discrete signal into a digital form is called
quantization.
 The quantization provides the staircase approximation to the input signal.
 If the step size is uniform, then it is called as uniform quantization. The
uniform quantization is further classified as midtread and midriser based
on the origin of the quantization in the middle of a tread or riser
respectively as shown in figure 5.32.
 Quantization noise or error is defined as the difference between the input
and quantized value.
 The SNR of the uniform quantization depends on the number of bits used
to represent the sample.
 In case of speech signal, uniform quantization cannot be
used as it reduces the SNR for the frequently occurring low
amplitude signals.
 This demands the need for the signal dependent
quantization.
 Non-uniform quantization provides fine quantization for low
amplitude levels and coarse quantization of high amplitude
levels.
 Quantization with non-uniform or unequal step size is called
as non-uniform quantization.

Figure : Nonuniform Quantization

 Nonuniform quantization is implemented using companding technique
which is the combination of compression at the transmitter and expansion
at receiver.
 Compander = Compressor + Expander
 The model of non-uniform quantizer is shown in figure 5.34.
 At the transmitter, the original signal is first distorted by using the
logarithmic compression characteristics and then using a uniform
quantizer.
 The compressor has a much steeper slope for low amplitude levels than
for high amplitude levels.
 Thus, the compressor provides higher gain to the low amplitude signals
and lower gain to the high amplitude signals.

 At the receiver, expander is used to restore the original signal which has
inverse characteristic of compressor.

 Generally compander uses a logarithmic compression, where the levels

are crowded near the origin and spaced farther apart near the peak values.
The two commonly used logarithmic compression laws are,
1. µ-law companding:
 The compressor characteristics is continuous, approximating a
linear dependence for low input levels and logarithmic for high
input levels. The µ-law compressor characteristics is
mathematically expressed as,

where,

 where µ is a positive constant, x and y represent input and output

voltages and xmax and ymax are the maximum positive excursions of
the input and output voltages respectively.
 The compression characteristics for several values of μ is shown in
figure 5.36(a). Uniform quantization (linear amplification)
corresponds to the parameter μ = 0. In North America, the standard
practical value for µ is 255. It is used for PCM telephone systems
in US, Canada and Japan.
2. A-law companding:

 The compressor characteristics is piecewise made up of linear

segment for low level inputs and logarithmic segment for high
level inputs.

 Where A is a positive constant, x and y represent input and

output voltages, and xmax and ymax are the maximum positive
excursions of the input and output voltages respectively.
 The compression characteristics for several values of A is shown
in figure 5.36(b). Uniform quantization (linear amplification)
corresponds to the parameter A=1.
 The standard practical value for A is 87.56. It is used for PCM
telephone system in Europe.
III) Encoding:

Encoding is defined as the process of
translating the discrete set of sample
values to a more appropriate form suited
for transmission through communication
channel.
The member of each discrete set of values
are represented as a particular
arrangement of discrete events called
code.
One of the discrete events in a code is
called as code element or symbol.
A particular arrangement of
symbols used in a code to represent
a single value of the discrete set is
called as code word or character.

In a binary code, each symbol

may be either of two distinct
values, such as a negative pulse or
positive pulse. The two symbols of
the binary code are denoted as 0
and 1.
In a binary code, each code word
consists of R bits, which denotes
the number of bits per sample.
Hence, by using such a code, we
represent a total of 2R distinct
values.
For example, a sample quantized
into one of 256 levels may be
represented by an 8-bit code word.
The electrical representation of the binary sequence is called as
line codes. The important line codes are listed as follows:

 Unipolar nonreturn-to-zero or On–off signaling: In this

signaling format, symbol 1 is represented by transmitting a pulse
of constant amplitude for the duration of the symbol, and symbol
0 is represented by switching off the pulse, as in figure 5.37(a).

 Polar nonreturn-to-zero signaling: In this format, symbols 1

and 0 are represented by pulses of equal positive and negative
amplitudes, as illustrated in figure 5.37(b). This format is
relatively easy to generate but its disadvantage is that the power
spectrum is large near zero frequency.

 Unipolar return-to-zero signaling: In this signaling format,

symbol 1 is represented by a positive rectangular pulse of half-
symbol width and symbol 0 is represented by transmitting no
pulse, as illustrated in figure 5.37(c).
 Bipolar return-to-zero signaling: This format uses three amplitude
levels as indicated in figure 5.37(d). Specifically, positive and
negative pulses of equal amplitude are used alternately for symbol
1, and no pulse is always used for symbol 0. A useful property of
this signaling is that the power spectrum of the transmitted signal
has no dc component and relatively insignificant low-frequency
components for the case when symbols 1 and 0 occur with equal
probability.
 Split-phase (Manchester code): In this format, symbol 1 is
represented by a positive pulse followed by a negative pulse, with
both pulses being of equal amplitude and half-symbol width. For
symbol 0, the polarities of these two pulses are reversed. The
Manchester code suppresses the dc component and has relatively
insignificant low-frequency components, regardless of the signal
statistics. Manchester code is illustrated in figure 5.37(e).

 Differential encoding:
 In this format, the information is encoded in terms of signal
transitions, as illustrated in figure 5.37(f).

 Here, a transition is used to designate symbol 0, whereas no transition

is used to designate symbol 1.

 It is apparent that a differentially encoded signal may be inverted

without affecting its interpretation.

 The original binary information is recovered by comparing the

polarity of adjacent symbols to establish whether or not a transition
has occurred.

 Differential encoding requires the use of a reference bit, as indicated

in figure 5.37(f).
Figure 5.37 Line codes (a) Unipolar nonreturn-to-zero (b) Polar nonreturn-to-zero(c) Unipolar
return-to-zero (d) Bipolar return-to-zero (e) Manchester code
(f) Differential encoding
IV)Regeneration:
The important feature of PCM lies in
the ability to control the effects of
distortion and noise produced by
transmitting a PCM signal through a
channel.
This is achieved by reconstructing the
PCM signal by means of a chain of
regenerative repeaters located at
sufficiently close spacing along the
transmission path.
The important blocks of regenerative repeater is shown in
figure 5.38. The three basic functions performed by the
regenerative repeaters are,

 Equalization – Shapes the received pulse so as to

compensate for the effects of amplitude and phase
distortions produced by imperfections of the channel.

 Timing – Provides a periodic pulse train derived from the

received pulses, for sampling the equalized pulses at the
instants of time where the SNR is maximum.

 Decision making – Enabled when at the sampling time

determined by the timing circuit, amplitude of the
equalized pulse plus noise exceeds a predetermined voltage
level.
Figure 5.38 Components of regenerative repeater
V) Decoding:
• The first operation in the receiver is to regenerate
the received pulses.

• These clean pulses are then regrouped into code

words and decoded into a quantized PAM signal.

VI) Reconstruction:
• The final operation is to recover the analog signal.

• This is done by passing the decoder output through a

low pass reconstruction filter whose cutoff
frequency is equal to the message bandwidth.
Synchronization:
For PCM system with TDM to operate
satisfactorily it is necessary that the timing
operations at the receiver, except for the time lost in
transmission and regenerative repeating follow
closely the corresponding operations at the
transmitter.
This needs a local clock at the receiver to keep
the same time as a distant standard clock at the
transmitter, except that the local clock is slower
by an amount corresponding to the time required to
transport the message signals from the transmitter
to the receiver.
Frame synchronization

Set aside a code element or pulse at

the end of a frame and to transmit
this pulse every other frame only.

A receiver search for the pattern of

1’s and 0’s alternating at half the
frame rate.
 Bandwidth requirement of PCM:
• The bit rate (R) is = n * fs = 2nW bits/sec.

• The minimum bandwidth requirement for binary

transmission of R bits/sec is R/2.

• Therefore, the minimum bandwidth requirement

of a PCM system is nfs /2

• At Nyquist rate, Minimum BW required is nW

• Thus, a PCM system expands the bandwidth of the

original signal by a factor of n.
Advantages of PCM:
PCM permits regeneration along the transmission
path which reduces noise and interference.
Multiplexing of multiple data streams is easy
possible.
Security can be easily incorporated using encryption
and decryption.

Disadvantages of PCM:
Cost and complexity is high compared to other pulse
modulation techniques.
Bandwidth requirement is high.
 MULTIPLEXING

Transmitting two or more signals simultaneously

can be accomplished by setting up one
transmitter-receiver pair for each channel, but
this is an expensive approach.

Multiplexing technique reduces the cost by

using the channel effectively.

It is defined as the technique of combining

separate message signals into a composite
signal for transmission over a common
channel.
The two commonly used methods for signal
multiplexing are

Frequency Division Multiplexing

(FDM)
Time Division Multiplexing (TDM)

FDM may be used with either analog or digital

signal transmission, whereas TDM is usually
used to transmit digital information.
 Frequency Division Multiplexing(FDM)

Voice frequencies transmitted over telephone

systems, for example, range from 300 to 3400
Hz.

To transmit a number of these signals over

the same channel (e.g. cable), the signals
must be kept apart so that they do not
interfere with each other and thus they can
be separated at the receiving end.
Frequency division multiplexing (FDM)
is defined as the technique of dividing the
total bandwidth available in a
communication medium into a series of
non-overlapping frequency bands and
allocating each frequency to carry a
separate message signal.
In FDM all users use the same common channel
at full time.
But each of the users is allocated with different
frequencies for transmission for avoiding the
signal interference.
Sometimes there is a possibility of cross talk
because all the users use the transmission
medium at the same time.
Block diagram of FDM Transmitter system
 In FDM signals are generated by sending devices and there are multiple
input lines. From the block diagram (Figure 6), channel 1 to channel n
are taken as the input channels.

 These signals reach at the input of the corresponding modulator where it

receives another signal from a crystal oscillator known as carrier signal,
which is a high-frequency high-amplitude signal.

 The carrier signal is modulated with the input signal. Different

modulators use different carrier signals for modulation.It should be
noted that the frequency band of one modulator will not make any
interference to the frequency band of other modulators.

 Each of the modulator produces the corresponding modulated signal at

their output. All the output of the modulators will be given to an adder or
mixer circuit; from there it is given to another modulator for further shift
of total bandwidth.

 Finally, this higher-frequency signal will be transmitted over the

Block diagram of FDM Receiver system
 The following block diagram (Figure 7) shows the concept of
demodulation of FDM signal at the receiving side.

 The antenna receives the multiplexed modulated signal from the

transmitter. This signal will be weak at the receiver. Therefore it is
necessary to amplify the signal. This is done at the initial stage of the
receiver.

 The amplified signal is then forwarded to the demodulator. The output

of the demodulator will be given to the band-pass filters which are
well designed with the central frequencies of the carriers as used
individually at the transmitting side.

 Thus the output of each BPF will be the same as that of the originally
modulated output of the corresponding modulator.

 Then we use the corresponding individual demodulators to recover the

original signal.
Advantage of FDM
 A large number of signals (channels) can be transmitted simultaneously.
 Demodulation of FDM is easy.
 FDM does not need synchronization between its transmitter and receiver
for proper operation.
 Due to slow narrowband fading, only a single channel gets affected.

Disadvantages of FDM
 The communication channel must have a very large bandwidth.
 Inter-modulation distortion takes place.
 A large number of modulators and filters are required.
 FDM suffers from the problem of cross talk.
 All the FDM channels get affected due to wideband fading.

Applications of FDM
 FDM is used for FM and AM radio broadcasting.
 FDM is used in television broadcasting.
 First-generation cellular telephone also uses FDM.
 Time Division
Multiplexing(TDM)
 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,
thus some of the time interval between adjacent samples is free for use
by other independent message sources on a time-shared basis.

 This enables the joint utilization of a common communication channel

by several independent message sources without mutual interference
among them.

 Time division multiplexing (TDM) is defined as the technique of

combining multiple data streams from independent sources by
assigning each stream a different time slot.

 The block diagram of time division multiplexing system is shown in

figure
The transmitter section of TDM consists of the following components,

Prealias filter:
 Each input message signal is first restricted in bandwidth by a low pass
prealias filter to remove the frequencies that are not used for an adequate
signal representation.

Commutator:
 The outputs of prealias filter are then applied to commutator, which is
usually implemented using electronic switching circuitry. It performs
two functions,
1.Sampling - To take a narrow sample of each of the N input message at a
rate fs that is slightly higher than 2W, where W is the cutoff frequency of
the pre-alias filter.

2.Interleaving - To sequentially interleave these N samples inside the

sampling interval Ts = 1/fs
Pulse amplitude modulator:
 After the commutation process, the multiplexed signal is applied to a
PAM, which transform the multiplexed signal into a suitable form for
transmission over the communication channel.
 The sampling rate for each of N message signal is determined in
accordance with sampling theorem. Let Ts denote the sampling period
for each message signal. Let Tx denote the time spacing between the
adjacent samples in the time-multiplexed signal as shown in figure 5.42.
It is clear that, Tx = Ts / N
 Thus, the use of TDM introduces a bandwidth expansion factor N

Figure 5.42 Waveform illustrating TDM for two message signal s(t) and m(t)
The receiver section of TDM consists of

Pulse amplitude demodulator:

 It performs the reverse operation of the pulse amplitude
demodulator.

Decommutator:
 It performs the reverse operation of the commutator present in
the transmitter. It separates the individual message signal
from the multiplexed signal. It must operate in synchronism
with the commutator.

Reconstruction filter:
 It performs the low pass filtering for the individual message
signals.
Advantages of TDM:
 Simple circuit compared to FDM.
 Immune to amplitude nonlinearities which is the source of
crosstalk.
 Easy to add signals

Disadvantages of TDM:
 Highly sensitive to dispersion in the channel which induces
ISI.
 Requires equalization at the receiver.
 Possibility of Synchronization and timing jitter problems at
high bit rates.
Applications of TDM:
 Digital telephony
 Satellite access
 Cellular radio
Multiple Access
Schemes(MAS)