Pulse Modulation

Eyob Mersha ([Link].)

E-mail:eyob.mersha12@[Link]
Phone:0909240909

Haramaya University

Haramaya Institute of Technology

School of Electrical and Computer Engineering

May 7, 2021

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 Pulse Modulation

Pulse Modulation

Introduction
In continuous-wave (CW) modulation some parameter of a sinusoidal car-
rier wave is varied continuously in accordance with the message signal.
In pulse modulation, some parameter of a pulse train is varied in accor-
dance with the message signal.
Depending on how the modulation is performed, we categorize pulse
modulation in to:
1 Analog pulse modulation and
2 Digital pulse modulation
.

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 Pulse Modulation

Pulse Modulation

In analog pulse modulation,

a periodic pulse train is used as the carrier wave
some characteristic feature of each pulse (e.g., amplitude, duration, or posi-
tion) is varied in a continuous manner in accordance with the corresponding
sample value of the message signal
information is transmitted basically in analog form, but the transmission
takes place at discrete times.
In digital pulse modulation, on the other hand, the message signal is rep-
resented in a form that is discrete in both time and amplitude, thereby
permitting its transmission in digital form as a sequence of coded pulses.
Simply put, digital pulse modulation has no CW counterpart.

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 Pulse Modulation

Pulse Modulation

The material presented in this chapter on pulse modulation teaches us the

following two lessons:
1 Lesson 1: Given a strictly band-limited message signal, the sampling theo-
rem embodies the conditions for a uniformly sampled version of the signal
to preserve its information content.
2 Lesson 2: Analog pulse-modulation systems rely on the sampling process
to maintain continuous amplitude representation of the message signal.
In contrast, digital pulse-modulation systems use not only the sampling
process but also the quantization process, which is non-reversible.
Quantization provides a representation of the message signal that is dis-
crete in both time and amplitude.
In so doing, digital pulse modulation makes it possible to exploit the full
power of digital signal-processing techniques.

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 Sampling Process

Sampling Process

Usually, but not exclusively, it is described in the time domain.

It is an basic operation to DSP and digital communications.
It is used to convert an analog signal into a corresponding sequence of
samples that are usually spaced uniformly in time.
Clearly, for such a procedure to have practical utility, it is necessary that
we choose the sampling rate properly, so that the sequence of samples
uniquely defines the original analog signal.

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 Sampling Process

Definition

Consider an arbitrary signal g(t) of finite energy, which is specified for all
time t.
A segment of the signal g(t) is shown in Figure 1(a).
Suppose that we sample the signal instantaneously and at a uniform rate,
once every Ts seconds.
Consequently, we obtain an infinite sequence of samples spaced Ts sec-
onds apart and denoted by g(nTs ), where n takes on all possible integer
values, both positive and negative.
àTs —sampling period or sampling interval
à fs = 1/Ts — sampling rate
This ideal form of sampling is called instantaneous sampling.

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 Sampling Process

Figure: Illustration of the sampling process. (a) Analog waveform g(t) (b)
Instantaneously sampled representation of g(t).

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 Sampling Process

Let gδ (t) denote the signal obtained by individually weighting the ele-
ments of a periodic sequence of Dirac delta functions spaced Ts seconds
apart by the sequence of numbers g(nTs ), as shown by Figure 1(b)
∞
X
gδ (t) = g(nTs )δ(t − nTs ) (1)
n=−∞

à gδ (t) is called the instantaneously (ideal) sampled signal.

The term δ(t − nTs ) represents a delta function positioned at time t = nTs .
From the definition of the delta function, it has unit area.
We may therefore view the multiplying factor g(nTs ) in Eq. (1) as a
”mass” assigned to the delta function δ(t − nTs ).
A delta function weighted in this manner is closely approximated by a
rectangular pulse of duration ∆t and amplitude g(nTs )/∆t; the smaller
we make the ∆t, better the approximation will be.

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 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 = 1/T0 .
Making a signal periodic in the time domain has the effect of sampling
the spectrum of the signal in the frequency domain.
Theorem
The duality property of the Fourier transform state that sampling a signal in
the time domain has the effect of making the spectrum of the signal periodic in
the frequency domain

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 Sampling Process

Time-frequency sampling-duality relationships

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 Sampling Process

The motivation for formulating the above Table is to lay the mathematical
groundwork for formulating the sampling theorem in the time-domain.
To this end, we reproduce the relationships listed at the bottom of the
right-hand side of the table in the form
∞
X ∞
X ∞
X
g(nTs )δ(t − nTs ) ↔ G(f − mfs ) = g(nTs )exp(−j2πTs f )
n=−∞ m=−∞ n=−∞
(2)
= Gδ (f )

where G(f) is the Fourier transform of the original signal g(t), and
fs = 1/Ts is the sampling rate.
In words, Eq. (2) states that the process of uniformly sampling a con-
tinuous time signal of finite energy results in a periodic spectrum with a
repetition frequency equal to the sampling rate.

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 Sampling Process

Sampling theorem

The relations of Equations (2) apply to any continuous-time signal g(t) of

finite energy.
Suppose, however, that the signal g(t) is strictly band-limited, with no
frequency components higher than W hertz.
That is, G(f) is zero for |f | > W, as illustrated in Fig.(a); the shape of the
spectrum shown in this figure is intended for the purpose of illustration
only.
1
Suppose also that we choose the sampling period Ts = 2W , which is the
maximum permissible value.
Then the corresponding spectrum Gδ (f ) of the sampled signal gδ (t) is as
shown in Fig.(b).

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 Sampling Process

Figure: (a) Spectrum of a strictly band-limited signal g(t). (b) Spectrum of

instantaneously sampled version of g(t) for a sampling period Ts = 1/2W. (c)
Frequency response of ideal low-pass filter aimed at recovering the original message
signal g(t) from its uniformly sampled version.
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 Sampling Process

Putting Ts = 1/(2W) in Eq. (2) and using Gδ (f ) to denote the Fourier

transform of Gδ (t), we may write
∞
X n jπnf
Gδ (f ) = g( )exp( ) (3)
n=−∞
2W W

Equation (3) defines the Fourier transform Gδ (f ) of the sequence g(n/2W)∞

−∞
, which is obtained by uniform sampling of a continuous-time signal g(t)
at the special rate (1/Ts ) = fs = 2W.
Eq. (3), is called the discrete-time Fourier transform of the sequence
g(n/2W)∞ −∞ .

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 Sampling Process

Returning to Eq. (2), suppose we isolate the term corresponding to m=0

in the summation term and thus write
∞
X
Gδ (f ) = fs G(f ) + fs G(f − mfs ) (4)
n=−∞

From this expression we find that, for a strictly band-limited signal, under
the two conditions
1 G(f ) = 0 for |f | ≥ W
2 fs = 2W
the summation term is constrained to be zero.
Then, solving the simplified expression for G(f), we obtain

1
G(f ) = Gδ (f ) −W <f <W (5)
2W

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 Sampling Process

Eliminating Gδ (f ) between Eq. (3) and Eq. (5) yields

∞
1 X n jπnf
G(f ) = g( )exp( ) −W <f <W (6)
2W n=−∞ 2W W

Therefore, if the sample values g(n/2W) of a signal g(t) are specified for
all time, then the Fourier transform G(f) of the signal g(t) is uniquely de-
termined, except for the scaling factor 1/2W, by the discrete-time Fourier
transform of Eq. (3) for the spectrum Gδ (f ) limited to the interval -
W ≤ f ≤ W.
Because g(t) is related to G(f) by the inverse Fourier transform, it fol-
lows that the signal g(t) is itself uniquely determined by the sample values
g(n/2W) for −∞ < n < ∞.
In other words, the sequence g(n/2W) has all the information contained in
g(t).

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 Sampling Process

Consider next the problem of reconstructing the signal g(t) from the se-
quence of sample values g(n/2W).
Substituting Eq. (5) into the formula for the inverse Fourier transform
defining g(t) in terms of G(f), we get
R∞
g(t) = −∞ G(f )exp(j2πft)dt = −W ∞
RW P n jπnf
n=−∞ g( 2W )exp(− W )exp(j2πft)d

We are permitted to interchange the order of summation and integration,

as they are both linear operations.
Accordingly, we may go on to redefine the desired signal g(t) as
∞ Z W
X n 1 n
g(t) = g( ) exp[j2πf (t − )]df (7)
n=−∞
2W 2W −W 2W

1
RW n sin(2πWt−nπ)
à But 2W −W exp[j2πf (t − 2W )]df = (2πWt−nπ) = sinc(2Wt − n)

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 Sampling Process

Using the above relation, the formula of Eq. (6) reduces to

∞
X n
g(t) = g( )sinc(2Wt − n), −∞<t <∞ (8)
n=−∞
2W

Equation (7) is the interpolation formula for reconstructing the original

signal g(t) from the sequence of sample values g(n/2W), with the sinc
function sinc(2Wt) playing the role of an interpolation function.
Each sample is multiplied by a delayed version of the interpolation func-
tion, and all the resulting waveforms are added to obtain g(t).
A synthesis filter or reconstruction filter aimed at recovering the origi-
nal strictly band-limited signal g(t) from its instantaneously sampled ver-
sion in accordance with Eq. (7) consists of an ideal low-pass whose fre-
quency response is limited exactly to the same band as the signal g(t)
itself-namely,−W ≤ f ≤ W.

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 Sampling Process

This reconstruction filter is non-causal and therefore non-physically real-

izable.
The discrete-time Fourier transform of Eq. (5) defines the message spec-
trum G(f) in terms of the uniformly spaced samples values g(n/2W) for
−∞ < n < ∞.
The interpolation formula of Eq. (7) defines the message signal g(t) in
terms of these same sample values.

Figure: Zero-order hold Figure: Linear

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 Sampling Process

On the basis of these two formulas, we may now state the sampling the-
orem for strictly band-limited signals of finite energy in two equivalent
parts:
1 Analysis: A band-limited signal of finite energy that has no frequency com-
ponents higher than W hertz is completely described by specifying the val-
ues of the signal at instants of time separated by 1/2W seconds.
2 Synthesis: A band-limited signal of finite energy that has no frequency
components higher than W hertz is completely recovered from knowledge
of its samples taken at the rate of 2W samples per second.

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 Sampling Process

The sampling rate of 2W samples per second for a signal bandwidth of W

hertz is called the Nyquist rate;
à 1/2W (measured in seconds) is called the Nyquist interval.
àNyquist rate is the minimum sampling rate permissible
The analysis part of the sampling theorem applies to the Tx .
The synthesis part of the theorem applies to the Rx .
A maximum of 2W independent elements of information per second can
be transmitted, error-free, over a noiseless channel of bandwidth W Hz.
The sampling theorem is important in signal analysis, digital signal pro-
cessing and transmission because it allows us to replace an analog signal
with a discrete sequence of numbers (i.e., a digital signal).

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 Sampling Process

Exercises 4.1: Specify the Nyquist rate and the Nyquist interval for each of
the following signals
a) g(t) = sinc(200t)
b) g(t) = sinc2 (200t)
c) g(t) = sinc(200t) + sinc2 (200t)
Exercises 4.2: Consider uniform sampling of the sinusoidal wave g(t) =
cos(πt). Determine the Fourier transform of the sampled waveform for each
of the following sampling periods:
a) Ts = 0.25s
b) Ts = 1s
c) Ts = 1.5s
Exercises 4.3: Consider a continuous-time signal defined by
g(t) = sin(πt)/πt. The signal g(t) is uniformly sampled to produce the
infinite sequence g(nTs )∞
−∞ . Determine the condition that the sampling
period Ts must satisfy so that the signal g(t) is uniquely recovered from the
sequence g(nT − s).
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 Sampling Process

Exercises 4.4: Find the minimum sampling rate and sampling interval for the
follwing signals
a) m(t) = 3cos(50πt) + 10sin(300πt) + 1.5cos(120πt)
b) m(t) = 8sin(600πt)cos(200πt)
Exercises 4.5: For a message signal x(t) = 5cos300πt
a) Find the minimum sampling rate
b) If fs =600 , what is the discrete time signal after sampling
c) If fs =200, what is the discrete time signal after sampling

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 Sampling Process

Aliasing phenomenon

Derivation of the sampling theorem is based on the assumption that the

signal g(t) is strictly band-limited.
In practice, however, no information-bearing signal of physical origin is
strictly band-limited, with the result that some degree of undersampling
is always encountered.
Consequently, aliasing is produced by the sampling process.
Definition
Aliasing refer 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

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 Sampling Process

The aliased spectrum shown by the solid curve in Fig. (b) pertains to an
”undersampled” version of the message signal represented by the spectrum
of Fig.(a).

Figure: Spectrum of a signal. (b) Spectrum of an undersampled version of the signal,

exhibiting the aliasing phenomenon.

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 Sampling Process

Figure: Effects of under sampling

To combat the effects of aliasing in practice, we may use two corrective

measures:
1 Prior to sampling, a low-pass anti-alias filter is used to attenuate those high-
frequency components of a message signal that are not essential to the in-
formation 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 synthesis filter used to recover
the original signal from its sampled version.
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 Sampling Process

Consider the example of a message signal that has been anti-alias (low-
pass) filtered, resulting in the spectrum shown in Fig. (a).
The corresponding spectrum of the instantaneously sampled version of
the signal is shown in Fig.(b), assuming a sampling rate higher than the
Nyquist rate.
According to the picture depicted in Fig. (b), we now readily see that the
design of a physically realizable reconstruction filter aimed at recovering
the original signal from its uniformly sampled version may be achieved
as follows (see Fig.(c)):
The reconstruction filter is of a low-pass kind with a passband extending
from -W to W, which is itself determined by the anti-alias filter.
The filter has a non-zero transition band extending (for positive frequen-
cies) from W to fs - W, where fs is the sampling rate.

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 Sampling Process

Effects of aliasing in image signal

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 Sampling Process

The non-zero transition band of the filter assures physical realizability, it

is shown dashed to emphasize the arbitrary way of actually realizing it.

Figure: (a) Anti-alias filtered spectrum of an information-bearing signal. (b)

Spectrum of instantaneously sampled version of the signal, assuming the use of
a sampling rate greater than the Nyquist rate. (c) Idealized amplitude response
of the reconstruction filter.

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 Sampling Process

Exercises 4.5: For an information signal

xa (t) = 3cos2000πt + 5sin6000πt − 10cos12000πt
a) What is the Nyquist rate for this signal?
b) Using a sampling rate fs = 5000samples/sec . What is the discrete-time
signal obtained after sampling?
c) What is the analog signal, ya (t) we can reconstruct from the samples if we
use ideal interpolation?
Exercises 4.6: We want to digitize and store a signal on a CD, and then
reconstruct it at a later time. Let the signal x(t) be
x(t) = 2cos(500πt) − 3sin(1000πt) + cos(1500πt) and let the sampling
frequency be fs = 2000Hz.
1 Determine the continuous time signal y(t) after the reconstruction.
2 Notice that y(t) is not exactly equal x(t). How could we reconstruct the
signal x(t) exactly from its samples x[n]?

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 Analog pulse modulation

Analog pulse modulation

The continuous time signal g(t) is sampled, and sample values are used to
modify certain parameters of a periodic pulse train.
The amplitudes, widths, or positions of pulses in proportion of the sample
values of the signal g(t).
Accordingly, there are three types modulation systems. These are:
1 Pulse amplitude modulation (PAM),
2 Pulse width modulation (PWM), and
3 Pulse position modulation (PPM)

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 Analog pulse modulation

Analog pulse modulation

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 Analog pulse modulation

Analog pulse modulation

In all of these cases instead of transmitting g(t), the corresponding pulse

modulated signal is transmitted.
At the receiver, the information of the pulse modulated signal is read and
the analog signal g(t) is constructed from it.
One advantage of using pulse modulation is that it permits the simultane-
ous transmission of several signals on a time sharing basis and it is called
time division multiplexing (TDM).
In this manner several signals can be multiplexed on the same channel by
reducing the pulse widths.

Figure: Time division multiplexing of two signals

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 Analog pulse modulation

Pulse amplitude modulation (PAM)

PAM is the simplest and most basic form of analog pulse modulation tech-
niques.
In PAM, the amplitudes of regularly spaced pulses are varied in proportion
to the corresponding sample values of a continuous message signal;
Ý the pulses can be of a rectangular form or some other appropriate shape.
PAM as defined here is somewhat similar to natural sampling, where the
message signal is multiplied by a periodic train of rectangular pulses.
The waveform of a PAM signal is illustrated in [Link] .
The dashed curve in this figure depicts the waveform of the message signal
m(t), and the sequence of amplitude-modulated rectangular pulses shown
as solid lines represents the corresponding PAM signal s(t).

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 Analog pulse modulation

PAM

There are two operations involved in the generation of the PAM signal:
1 Instantaneous sampling of the message signal m(t) every Ts seconds, where
the sampling rate fs is chosen in accordance with the sampling theorem.
2 Lengthening the duration of each sample, so that it occupies some finite
value 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 band-
width is inversely proportional to pulse duration.
However, care has to be exercised in how long we make the sample dura-
tion T, as the following analysis reveals.

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 Analog pulse modulation

PAM

Figure: The waveform of a PAM signal

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 Analog pulse modulation

PAM

A PAM signal is generated by using a pulse train, called the sampling

signal(clock signal) to operate an electronic switch or ”chopper”.
This produce samples of the analog message signal as shown below
the switch is closed for the duration of the each pulse allowing the mes-
sage signal at the sampling time to become part of the output
The switch is open for the reminder of each sampling period making the
output zero.
à This type of sampling is called natural sampling
For Flat Top sampling, a sample-and-hold circuit is used in conjunction
with the chopper to hold the amplitude of each pulse at a constant level
during the sampling time.

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 Analog pulse modulation

PAM

Figure: Generation of PAM signal

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 Analog pulse modulation

PAM

The two different types of PAM signals

vFLAT TOP PAM:
The amplitude of each pulse is directly proportional to instantaneous
modulating signal amplitude at the time of pulse occurrence and then keeps
the amplitude of the pulse for the rest of the half cycle.

Figure

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 Analog pulse modulation

PAM

vNatural PAM:
The amplitude of each pulse is directly proportional to the instantaneous
modulating signal amplitude at the time of pulse occurrence and then follows
the amplitude of the modulating signal for the rest of the half cycle.

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 Analog pulse modulation

FLAT top and Natural PAM comparison

Flat top PAM is the best for transmission because we can easily remove
the noise and we can also easily recognize the noise.
When we compare the difference between the flat top PAM and natural
PAM, flat top PAM principle of sampling uses sample and hold circuit.
In natural principle of sampling, noise interference is minimum.
But in flat top PAM noise interference maximum.
Flat top PAM and natural PAM are practical and sampling rate satisfies
the sampling criteria.

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 Analog pulse modulation

PAM

Sample-and-hold filter: analysis

Let s(t) denote the sequence of flat-top pulses generated in the manner
described in figure above.
Hence, PAM signal is expressed as
∞
X
s(t) = m(nTs )h(t − nTs ) (9)
n=−∞

where Ts is the sampling period and m(nTs ) is the sample value of m(t)
obtained at time t = nTs .
The h(t) is a standard rectangular pulse of unit amplitude and duration T,
defined as follows (see Fig. 4.8(a)):
t − T/2 1, 0<t<T 1
, t=0, t=T
h(t) = rect( ) = {0, otherwise
2
(10)
T
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 Analog pulse modulation

PAM

By definition, the instantaneously sampled version of m(t) is given by:

∞
X
mδ (t) = m(nTs )δ(t − nTs ) (11)
n=−∞

where δ(t − nTs ) is a time-shifted delta function.

To modify mδ (t) so as to assume the same form as the PAM signal s(t),
we convolve mδ (t)) with
R ∞the pulse h(t), obtaining
mδ (t) ? h(t) = n=−∞ mδ (τ )δ(t − nTs )
R∞
= n=−∞ ∞
P
n=−∞ m(nTs )δ(τ − nTs )h(t − τ )dτ
∞
X Z ∞
mδ (t) ? h(t) = m(nTs ) δ(τ − nTs )h(t − τ )dτ (12)
n=−∞ n=−∞

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 Analog pulse modulation

PAM

Using the shifting

R ∞ property of the delta function - namely,
−∞ δ(τ − nTs )h(t − τ )dτ = h(t − nTs )
we find that Eq. (12) reduces to
∞
X
mδ (t) ? h(t) = m(nTs )h(t − nTs ) (13)
n=−∞

The summation terms in Eqs. (9) and (13) are identical.

It follows therefore that the PAM signal s(t) is mathematically equivalent
to the convolution of mδ (t) , the instantaneously sampled version of m(t),
and the pulse h(t), as shown by

s(t) = mδ (t) ? h(t) (14)

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 Analog pulse modulation

PAM

Figure: (a) Rectangular pulse h(t). (b) Spectrum H(f) defined in terms of its
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 Analog pulse modulation

PAM

Taking the Fourier transform of both sides of Eq. (14) and recognizing
that the convolution of two time functions is transformed into the multi-
plication of their respective Fourier transforms, we get
S(f ) = Mδ (f )H(f ) (15)
where S(f) = F[s(t)], Mδ (f ) = F[mδ (t)], and H(f) = F[h(t)].
From Eq. (2) we find that the Fourier transform Mδ (f ) is related to the
original message spectrum m(t) as follows:
∞
X
Mδ (f ) = fs M(f − mfs ) (16)
n=−∞

Therefore, substitution of Eq. (16) into (15) yields

∞
X
S(f ) = fs M(f − kfs )H(f ) (17)
n=−∞

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 Analog pulse modulation

PAM

But the Fourier transform of the rectangular pulse h(t) is given by

H(f ) = Tsinc(fT)exp(−jπft) (18)

To recover the original message signal m(t) from a PAM signal s(t), pass
s(t) through a low-pass filter whose frequency response is defined in Fig.
4.2(c); here it is assumed that the message signal m(t) is limited to band-
width W and the sampling rate fs is larger than the Nyquist rate 2W.
Then from Eq. (17) the spectrum of the resulting filter output is equal to
M(f)H(f).
This output is equivalent to passing the message signal m(t) through an-
other low-pass filter of transfer function H(f).
The next step in recovering the message signal m(t) requires the use of
equalization.

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 Analog pulse modulation

PAM

Aperture effect and its equalization

Figure 4.8(b) shows plots of the magnitude and phase of the Fourier trans-
form H(f) versus frequency f.
From this figure we see that by using flat-top samples to generate a PAM
signal, we have introduced amplitude distortion as well as a delay of T/2.
This effect is rather similar to the variation in transmission with frequency
that is caused by the finite size of the scanning aperture in television.
For this reason, the distortion caused by the use of PAM (based on flat-top
sampling) to transmit an analog information-bearing signal is referred to
as the aperture effect.
This distortion may be corrected by connecting an equalizer in cascade
with the low-pass reconstruction filter, as shown in figure below.

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 Analog pulse modulation

PAM

The equalizer has the effect of decreasing the in-band loss of the recon-
struction filter as the frequency increases in such a manner as to compen-
sate for the aperture effect.
Ideally, the amplitude response of the equalizer is given by
1 1 πf
= = (19)
|H(f )| Tsinc(fT) sin(πfT)

Reconstruction Message
PAM signal s(t) Equalizer
filter signal m(t)

Figure: Recovering the message signal m(t) from the PAM signal s(t).

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 Analog pulse modulation

PAM

The amount of equalization needed in practice is usually small.

Indeed, for a duty cycle (T/Ts ) 6 0.1, the amplitude distortion is less
than 0.5 percent, in which case the need for equalization may be omitted
altogether.
The noise performance of a PAM system can never be better than direct
transmission of the message signal.
Accordingly, we find that for transmission over long distances, PAM would
be used only as a means of message processing for time-division multi-
plexing.

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 Analog pulse modulation

vAdvantages of Pulse Amplitude Modulation (PAM):

It is the base for all digital modulation techniques and it is simple process
for both modulation and demodulation technique.
No complex circuitry is required for both transmission and reception.
à Tx and Rx circuitry is simple and easy to construct.
PAM can generate other pulse modulation signals and can carry the mes-
sage or information at same time.
vDisadvantages of Pulse Amplitude Modulation (PAM):
Bandwidth should be large for transmitting the PAM signal.
àDue to Nyquist criteria also high bandwidth is required.
Pulse amplitude signal varies, so power required for transmission will be
more, peak power is also, even at receiving more power is required to
receive the pulse amplitude signal.

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 Analog pulse modulation

The frequency varies according to the message signal.

Due to these variations in the signal frequency, interferences will be there.
So noise will be great.
For PAM, noise immunity is less when compared to other modulation tech-
[Link] is almost equal to AM.
vApplications of Pulse Amplitude Modulation (PAM):
It is mainly used in Ethernet which is type of computer network communi-
cation, we know that we can use Ethernet for connecting two systems and
transfer data between the systems.
It is also used for photo biology which is a study of photosynthesis.
Used as electronic driver for LED lighting.
Used in many micro controllers for generating the control signals etc.

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 Analog pulse modulation

pulse time modulation

In pulse time modulation, amplitude of pulse is held constant, whereas

position of pulse or width of pulse is made proportional to the amplitude
of signal at the sampling instant.
There are two types of pulse time modulation.
1 Pulse Width Modulation
2 Pulse Position Modulation
Pulse width modulation is also called
Pulse-duration modulation (PDM)
Pulse-length modulation (PLM)
In PAM, pulse amplitude is the variable parameter.
In PDM, 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.
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 Analog pulse modulation

PDM is wasteful of power, in that 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 transi-
tions are essentially 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 occur-
rence is varied in accordance with the message signal.
Let Ts denote the sample duration.
Using the sample m(nTs ) of a message signal m(t) to modulate the posi-
tion of the nth pulse, we obtain the PPM signal
∞
X
s(t) = g(t − nTs − kp m(nTs )) (20)
n=−∞

where kp is the sensitivity factor of the pulse-position modulator (in sec-

onds per volt) and g(t) denotes a standard pulse of interest.
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 Analog pulse modulation

Clearly, the different pulses constituting the PPM signal must be strictly
non-overlapping; a sufficient condition for this requirement to be satisfied
is to have
Ts
g(t) = 0, |t| > ( ) − kp |m(t)|max (21)
2
which in turn requires that
Ts
( ) > kp |m(t)|max (22)
2
The closer kp |m(t)|max is to Ts/2, the narrower must the standard pulse be
to ensure that the individual pulses of the PPM signal s(t) do not interfere
with each other, and the wider will the bandwidth occupied by the PPM
signal be.
Assuming that Eq. (19) is satisfied and that there is no interference be-
tween adjacent pulses of the PPM signal s(t), then the signal samples
m(nTs ) can be recovered perfectly.
Furthermore, if the message signal m(t) is strictly band-limited, it follows
from the sampling theorem that the original message signal m(t) can be
recovered from the PPM signal s(t) without distortion.
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 Quantization

Quantization

A continuous signal, such as voice, has a continuous range of amplitudes

and therefore its samples have a continuous amplitude range.
In other words, within the finite amplitude range of the signal, we find an
infinite number of amplitude levels.
In actual fact, however, it is not necessary to transmit the exact amplitudes
of the samples.
We say so because any human sense (the ear or the eye) as the ultimate
receiver can detect only finite intensity differences.
This means that the original continuous signal may be approximated by
a signal constructed of discrete amplitudes selected on a minimum-error
basis from an available set.

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 Quantization

The existence of a finite number of discrete amplitude levels is a basic

condition of digital pulse modulation.
Clearly, if we assign the discrete amplitude levels with sufficiently close
spacing, we can make the approximated signal indistinguishable from the
original continuous signal for all practical purposes.
Quantization is non-reversible.
Definition
Quantization is representing the sampled values of the amplitude by a finite
set of levels, which means converting a continuous-amplitude sample into a
discrete-time signal.

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 Quantization

The following figure shows how an analog signal gets quantized.

The blue line represents analog signal while the brown one represents the
quantized signal.

Figure: Quantization

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 Quantization

Definition
Amplitude quantization is defined as the process of transforming the sample
amplitude m(nTs) of a baseband signal m(t) at time t = nTs into a discrete
amplitude v(nTs ) taken from a finite set of possible levels.

Quantization process that is memoryless and instantaneous

ài.e the transformation at time t = nTs is not affected by earlier or later
samples of the message signal.
This form of quantization, though not optimal, is commonly used in prac-
tice because of its simplicity.
Quantization is a non-linear transformation which maps elements from
continuous to a finite set.
It is also the second step in A/D conversion.

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 Quantization

There are two types of distortion associated with a quantizer:

1 Overload or clipping distortion
2 Quantization distortion
Overload distortion occurs when the input signal exceeds the quantizer’s
input range.
Once the quantizer range has been exceed the output will remain at its
maximum or minimum value until the input falls within the quantizer in-
put range.
To avoid clipping, the quantizer is matched to the input signal.
The quantization error signal produces quantization noise or distortion in
the reconstructed message signal.
The sampling rate will also affect the quantization noise since the quanti-
zation error will become larger as the sampling rate decreases.

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 Quantization

Uniform quantization and error

The difference between an input value and its quantized value is called a
Quantization Error.
A Quantizer is a logarithmic function that performs Quantization (round-
ing off the value).
An analog-todigital converter (ADC) works as a quantizer.
Average error power:
1
RV 2 1
R ∆2 2 ∆2 (V.2−n+1 )2 V 2 −2n
2V −V e (s) = ∆ −∆ x dx = 12 = 12 = 3 2
2
Suppose the input signal is a triangular wave between -V and V.
V2
Then the average power is 3

s
( )out = 22n (23)
N

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 Quantization

Uniform quantization and error

Figure

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 Quantization

Definition
Dynamic range of an input signal is the ratio of the largest to the smallest
power levels which the input signal can take on and be reproduced with the
acceptable signal distortion.

The dynamic range of the quantizer input in the PCM system is 6n dB.
Nonuniform Quantizer Used to reduce quantization error and increase the
dynamic range when input signa not uniformly distributed over its allowed
range of values.

Figure: Non-uniform Quantization

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 Quantization

Types of Quantization

There are two types of Quantization

Uniform Quantization and
Non-uniform Quantization
The type of quantization in which the quantization levels are uniformly
spaced is termed as a Uniform Quantization.
The type of quantization in which the quantization levels are unequal and
mostly the relation between them is logarithmic, is termed as a Non-
uniform Quantization.
There are two types of uniform quantization. They are
Mid-Rise type and
Mid-Tread type.

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 Quantization

At the quantizer output, the index k is transformed into an amplitude vk

that represents all amplitudes that lie inside the interval Ik.
The amplitudes vk, k = 1, 2, ...., L, are called representation levels or re-
construction levels, and the spacing between two adjacent representation
levels is called a quantum or step-size.
Thus, the quantizer output v equals vk if the input signal sample m belongs
to the interval Ik . The mapping

v = g(m) (24)
is the quantizer characteristic.
This characteristic is described by a staircase function.

Figure: Description of a memoryless quantizer.

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 Quantization

Quantizers can be of a uniform or nonuniform type.

In a uniform quantizer, the representation levels are uniformly spaced;
otherwise, the quantizer is nonuniform.
The quantizers considered in this section are of the uniform variety.

Figure: Two types of quantization: (a) midtread and (b) midrise

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 Quantization

The quantizer characteristic can also be of a midtread or midrise type.

Figure 4.11(a) shows the input-output characteristic of a uniform quan-
tizer of the midtread type, which is so called because the origin lies in the
middle of a tread of the staircase like graph.
Figure 4.11(b) shows the corresponding input-output characteristic of a
uniform quantizer of the midrise type, in which the origin lies in the mid-
dle of a rising part of the staircase like graph.
Note that both the midtread and midrise types of uniform quantizers, il-
lustrated in Fig. 4.11, are symmetric about the origin.
In going from continuous-wave modulation to analog pulse modulation,
we have moved ourselves into discrete-time signal processing.
Why not go one step further and also incorporate amplitude discretiza-
tion? Indeed, this is precisely what is done in digital pulse modulation.

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 Quantization

In so doing, we have a new family of modulation techniques for the trans-

mission of analog signals over communication channels.
The advantages offered by digital pulse modulation techniques include
the following:
1 Performance: In an analog communication system, using either continuous-
wave modulation or analog pulse modulation technique, the effects of sig-
nal distortion and channel noise (incurred along the transmission path) are
cumulative.
In sharp contrast, digital pulse modulation permits the use of regenerative
repeaters, which, when placed along the transmission path at short enough
distances, can practically eliminate the degrading effects of channel noise
and signal distortion.
2 Ruggedness: Unlike an analog communication system, a digital communi-
cation system can be designed to withstand the effects of channel noise and
signal distortion, provided the noise and distortion are kept under certain
limits.

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 Quantization

Reliability: Digital communication systems can be made highly reliable

by exploiting powerful error-control coding techniques in such a way that
the estimate of a message signal delivered to a user is almost indistin-
guishable from the message signal delivered by a source of information at
the other end of the system.
Security: Digital communication systems can be made highly secure by
exploiting powerful encryption algorithms that rely on digital processing
for their implementation.
Efficiency: Digital communication systems are inherently more efficient
than analog communication systems in the tradeoff between transmission
bandwidth and signal-to-noise ratio.
System integration: The use of digital communications makes it possible
to integratebdigitized analog signals (i.e., voice and video signals) with
digital computer data, which is not possible with analog communications.

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 Quantization

The benefits of using digital pulse modulation, however, are attained at

the expense of increased system complexity.
Nevertheless, by exploiting the computing power of digital signal proces-
sors in hardware and/or software form and the flexibility these processors
offer, digital communication systems can be designed in a cost-effective
manner, thanks to the continuing improvements in very-large-scale inte-
grated (VLSI) silicon chips.
There are three family members of digital pulse modulation technique -
namely,
Pulse-code modulation
Delta modulation
Differential pulse-code modulation.
,

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 Quantization

Pulse code modulation(PCM)

In PCM, a message signal is represented by a sequence of coded pulses,

which is accomplished by representing the signal in discrete form in both
time and amplitude.
The basic operations performed in the transmitter of a PCM system are
sampling, quantization, and encoding, as shown in Fig. 4.12(a); the low-
pass filter prior to sampling is included merely to prevent aliasing of the
message signal.
The quantizing and encoding operations are usually performed in the
same circuit, which is called an analog-to-digital converter.
The basic operations in the receiver are regeneration of impaired signals,
decoding, and reconstruction of the train of quantized samples, as shown
in Fig.(c) below.

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 Quantization

Regeneration also occurs at intermediate points along the transmission

path as necessary, as indicated in Fig(b) above.
When TDM is used, it becomes necessary to synchronize the receiver to
the transmitter for the overall system to operate satisfactorily.

Figure: The basic elements of a PCM system: (a) Transmitter, (b) transmission path,
connecting the transmitter to the receiver, and (c) receiver.
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 Quantization

Operations in the transmitter

(i) Sampling
The incoming message (baseband) signal is sampled with a train of rect-
angular pulses, narrow enough to closely approximate the instantaneous
sampling process.
To ensure perfect reconstruction of the message signal at the receiver, the
sampling rate must be greater than twice the highest frequency component
W of the message signal in accordance with the sampling theorem.
In practice, an anti-alias (low-pass) filter is used at the front end of the
sampler in order to exclude frequencies greater than W before sampling,
as shown in Fig. 4.12(a).
Thus the application of sampling permits the reduction of the continu-
ously varying message signal (of some finite duration) to a limited number
of discrete values per second.

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 Quantization

Operations in the transmitter

v (ii) Nonuniform Quantization

The sampled version of the message signal is then quantized, thereby pro-
viding a new representation of the signal that is discrete in both time and
amplitude.
The quantization process may follow a uniform law
In certain applications, however, it is preferable to use a variable separa-
tion between the representation levels.
For example, the range of voltages covered by voice signals, from the
peaks of loud talk to the weak passages of weak talk, is on the order of
1000 to 1.
By using a nonuniform quantizer with the feature that the step size in-
creases as the separation from the origin of the input-output amplitude
characteristic is increased, the large end-step of the quantizer can take
care of possible excursions of the voice signal into the large amplitude
ranges that occur relatively infrequently.
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 Quantization

Operations in the transmitter

In other words, the weak passages that need more protection are favored
at the expense of the loud passages.
In this way, a nearly uniform percentage precision is achieved throughout
the greater part of the amplitude range of the input signal, with the result
that fewer steps are needed than would be the case if a uniform quantizer
were used.
(iii) Encoding
In combining the processes of sampling and quantization, the specifica-
tion of a continuous message (baseband) signal becomes limited to a dis-
crete set of values, but not in the form best suited to transmission over a
wire line or radio path.
To exploit the advantages of sampling and quantization for the purpose of
making the transmitted signal more robust to noise, interference and other
channel degradations, we require the use of an encoding process.
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 Quantization

Operations in the transmitter

Any plan for representing this discrete set of values as a particular ar-
rangement of discrete events is called a code.
One of the discrete events in a code is called a code element or symbol.
Example, the presence or absence of a pulse is a symbol.
A particular arrangement of symbols used in a code to represent a single
value of the discrete set is called a 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 customarily denoted as 0 and 1.
In practice, a binary code is preferred over other codes (e.g., ternary code)
for two reasons:
1 The maximum advantage over the effects of noise in a transmission medium
is obtained by using a binary code, because a binary symbol withstands a
relatively high level of noise.
2 The binary code is easy to generate and regenerate.
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 Quantization

Operations in the transmitter

Suppose that, in a binary code, each code word consists of R bits: the bit
is an acronym for binary digit.
Then R denotes the number of bits per sample.
Hence, by using such a code, we represent a total of 2R distinct numbers.
For example, a sample quantized into one of 256 levels may be repre-
sented by an 8-bit code word.
There are several ways of establishing a one-to-one correspondence be-
tween representation levels and code words.
A convenient method is to express the ordinal number of the representa-
tion level as a binary number.
In the binary number system, each digit has a place- value that is a power
of 2, as illustrated in Table 4.2 for the case of four bits per sample (i.e.,
R=4).

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 Quantization

Table 4.2 Binary Number System for R = 4

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 Quantization

Operations in the receiver

(i)Decoding and Expanding

The first operation in the receiver is to regenerate (i.e., reshape and clean
up) the received pulses one last time.
These clean pulses are then regrouped into code words and decoded (i.e.,
mapped back) into a quantized PAM signal.
The decoding process involves generating a pulse whose amplitude is the
linear sum of all the pulses in the code word; each pulse is weighted by
its place value (20 , 21 , 22 2R−1 ) in the code, where R is the number of bits
per sample.
The sequence of decoded samples represents an estimate of the sequence
of compressed samples produced by the quantizer in the transmitter.
We use the term ”estimate” here to emphasize the fact that there is no way
for the receiver to compensate for the approximation introduced into the
transmitted signal by the quantizer.
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 Quantization

Moreover, other sources of noise include bit errors and jitter produced
along the transmission path.
In order to restore the sequence of decoded samples to their correct rel-
ative level, we must, of course, use a subsystem in the receiver with a
characteristic complementary to the compressor, used in the transmitter.
Such a subsystem is called an expander.
Ideally, the compression and expansion laws are exactly inverse so that,
except for the effect of quantization, the expander output is equal to the
compressor input if these two devices were connected directly.
The combination of a compressor and an expander is referred to as a com-
pander.

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 Quantization

(ii) Reconstruction
The final operation in the receiver is to recover the message signal.
This operation is achieved by passing the expander output through a low-
pass reconstruction filter whose cutoff frequency is equal to the message
bandwidth.
Recovery of the message signal is intended to signify estimation rather
than exact reconstruction.
The term ”modulation” in pulse-code modulation is a misnomer.
In reality, pulse-code modulation is a source-encoding strategy, by means
of which an analog signal emitted by a source is converted into digital
form.

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 Quantization

Important concepts in TDM

N signals which are TDM and band limited to the highest frequency of
the multiplexed signals then the frame size is
Ts = NnTb
Rb
Sampling rate fs = Nn
Bit rate Rb = Nnfs
Rb
Bandwidth 2
To ensure speed synchronization, small amount of additional bits will be
transmitted at the end of each frame and these bits are called synchroniza-
tion [Link],
Ts = (Nn + a)Tb
Rb
Sampling rate fs = Nn+a

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 Quantization

Transmission Bandwidth and Output SNR

For Binary PCM, we assign a distinct group of n binary digits (bits) to

each of the L quantization levels.
Since a sequence of n binary digits can be arranged in 2n patterns,
L = 2n or n = logL2
Each quantized levels is thus encoded in to n bits.
Because a signal m(t) is band-limited to B Hz we require a minimum of
2B samples per second
Thus sampling rate is fs = 2nB bits per second
Since a unit bandwidth (1 Hz) can transmit a maximum of two pieces of
information(bits), the minimum transmission bandwidth (BT ) is
BT = f2s = 2nB
2 = nB Hz
BT is a theoretical minimum transmission bandwidth required to transmit
a PCM signal.

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 Quantization

Example A signal m(t) band limited to 3 kHz is sampled at a rate of 33 13 %

higher than the Nyquist rate. The maximum acceptable error in the sample
amplitude (the maximum quantization error) is 0.5% of the peak amplitude
mp . The quantized samples are binary coded. Find the minimum bandwidth of
a channel required to transmit the encoded binary signal. If 24 such signals
are time division multiplexed, determine the minimum transmission
bandwidth required to transmit the multiplexed signal.
Solution
The Nyquist sampling rate fs = 2x3000 Hz= 6000 Hz(samples per second)
However, the actual given sampling rate is fs = 3000 × 1 13 =8000 Hz
The quantization step is ∆v, and [Qe ]max = ±∆v/2
Therefore,
∆v mp 0.5
2 = L = 100 mp =⇒ L=200
For binary coding, L must be a power of 2. Hence, the next higher value of L
that is a power of 2 is L=256.
We need n=log2562 = 8 bits per sample

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 Quantization

We require to transmit 8000 samples/sec= 8000 samples/sec×(8

bits/sample)= 64,000 bits/sec
Because we can transmit up to 2 bits/s per Hz of bandwidth, we require a
minimum transmission bandwidth
BT = nfs /2 = (64, 0000 bits/s)/(2 bits) = 32 KHz
The multiplexed signal has a total of fs = 24× 64,000 bits/s= 1.536 Mbits/sec
The minimum transmission bandwidth to transmit these multiplexed signals is
BT = (1.536 Mbits/sec)/2 bits = 0.786 MHz

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 Quantization

PCM system parameters

PCM data rates (bps)=2nfm

PCM Bandwidth (Hz) = (1/2) PCM Data Rate =(1/2)2nfm =nfm
Dynamic Range (dB) = 20 log (2n − 1)
Coding Efficiency (%) = [(minimum bits)/(actual bits)] x100
Where n is number of PCM encoding bits and
fm frequency component of information signal

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 Quantization

Line codes

In reality, PCM, DM, and DPCM represent different strategies for source
encoding, whereby an analog signal is converted into digital form.
However, all three of them share a common feature: once a binary se-
quence of 1s and 0s is produced, a line code is needed for electrical rep-
resentation of that binary sequence.
There are several line codes that can be used for this representation, as
summarized here:
On-off signaling:
In which symbol 1 is represented by transmitting a pulse of constant ampli-
tude for the duration of the symbol, and symbol 0 is represented by switch-
ing off the pulse, as in Fig. 4.13(a).
Nonreturn-to-zero (NRZ) signaling:
In which symbols 1 and 0 are represented by pulses of equal positive and
negative amplitudes, as illustrated in Fig. 4.13(b)

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 Quantization

Return-to-zero (RZ) signaling:

In which 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 Fig. 4.13(c).
Bipolar return-to-zero (BRZ) signaling:
Which uses three amplitude levels as indicated in Fig. 4.13(d). Specifi-
cally, 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 BRZ signaling is that the power spectrum of the trans-
mitted signal has no dc component and relatively insignificant low-frequency
components for the case when symbols 1 and 0 occur with equal probabil-
ity.
Split-phase (Manchester code)
which is illustrated in Fig. 4.13(e).
In this method of signaling, 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 in-
significant low-frequency components, regardless of the signal statistics.
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 Quantization

Figure: Line codes. (a) OnâĂŞoff signaling. (b) Nonreturn-to-zero signaling. (c)
Return-to-zero signaling. (d) Bipolar return-to-zero signaling. (e) Split-phase or
Manchester encoding. (f) Differential encoding.
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 Quantization

Differential encoding, in which the information is encoded in terms of

signal transitions, as illustrated in Fig. 4.13(f).
In the example of the binary PCM signal shown in the figure, 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.
Note that differential encoding requires the use of a reference bit, as indi-
cated in Fig. 4.13(f).
The waveforms shown in parts (a) to (f) of Fig. 4.13 are drawn for the
binary data stream 01101001.
It is important, to note that rectangular pulse-shaping is used to draw these
waveforms, largely to simplify the electrical representation.

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 Quantization

Exercise 4.7: Twenty four voice signals are sampled uniformly and then time
division multiplexed. The highest frequency component of each voice signals
are 3.4 KHz.
1 If the signals are pulse amplitude modulated using Nyquist sampling rate,
what is the minimum channel bandwidth required?
2 If the signals are pulse code modulated with an 8 bit encoder, what is the
sampling rate? The signaling rate of the system is 1.5 × 106 bits/sec.

Solution:
i) If N channels are TDM, then the minimum transmission bandwidth is
BT = NW
where W is the maximum frequency in the signals
BT = NW = 24 × 3.4KHz = 81.6KHz
ii) The signaling rate of the system is given as
r = 1.5 × 106 bits/sec
Since there are 24 signals in the system the bit rate of an individual channel is,
6
r(onechannel) = 1.5×10 24 bit/ssec = 62500 bits/sec
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Since each sample is encoded using 8 bits, the samples per second will be,
Samples/sec = r(onechannel) bits/sec
bits/sample
Samples per second is sampling frequency.
i.e fs = 62500 bits/sec
8 bits/sample
=7812.5 Hz or samples/ second

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Exercise 4.8: Twenty four voice signals are sampled uniformly and then time
division multiplexed. The sampling operation uses flat top samples with
1µsec duration. The multiplexing operation provides for synchronization by
adding an extra pulse of 1µsec duration. Assuming sampling rate of 8 KHz,
calculate spacing between successive pulses of multiplexed signal and setup a
scheme for accomplishing a multiplexing requirement.
Solution: There are 24 voice signal pulses plus 1 synchronization pulse.
Thus, there are 25 pulses in total. fs =8 KHz
Hence, duration of one frame pulse is
Ts = f1s = 8000
1
= 125µsec

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Thus, in 125 µsec time there are 25 pulses at a uniform distance.

As shown above each pulse are separated with a distance 125µsec
25 = 5µsec
Width of the pulse is 1µsec.
Spacing between pulses =5µsec − 1µsec = 4µsec

Figure: Multiplexing scheme

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 Quantization

Exercise 4.9: 5 messages signals, each having 2 KHz are multiplexed using
TDM. Number of quantization levels used are 256. Determine the
transmission bandwidth of the systems.
Solution
Given that N=5, W= fm = 2KHz and L = 256 = 28 then n=8
We know that fs = 2fm =4 KHz
Hence, Bandwidth of transmission is
BT = R2b = nNf2
s

8×5×4
= 2 =80 KHz
Exercise 4.10: 10 signals, each band limited to 2 KHz are multiplexed using
TDM. Time taken by the commutator to make one complete rotation is
125µsec. Determine the bit rate of the transmitter if 5 bit encoder is used.
Solution
Given that, the number of signals N=10,
Sampling period of the multiplexer Ts = 125µsec
n=5 bits/sample

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 Quantization

Then, 1
fs = T1s = 125µsec = 8KHz
Hence, the bit rate of the system is
Rb = nNfs = 5 × 10 × 8KHz=400 kbps
Exercise 4.11: 10 sinusoidal message signals, each having frequency of 10
KHz are multiplexed using TDM. Sampling rate is 25% higher than the NR.
Maximum quantization error can be at most of 1% of peak to peak amplitude
of message signal. 5 number of synchronization bits are transmitted at the end
of each frame. Find Bit rate of the transmitter?
Solutions
Given that N = 10, fm = 10 KHz, , a = 5
[Qe ]max = 1% of peak to peak amplitude
= 0.01× peak to peak amplitude
fs = 1.25NR = 1.25(2fm )
= 1.25(20KHz) = 25KHz

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 Quantization

Now [Qe ]max ≤ 0.01 × 2Am

∆
2 ≤ 0.02Am
1 2Am
2 [ 2n ] ≤ 0.02Am
2n ≥ 50
n=6
Thus the bit rate is
Rb = (Nn + a)fs
= [(6)(10) + 5]25KHz
= 1625 Kbps

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 Quantization

Exercise 4.12: The bandwidth of signal input to the PCM is restricted to 4

KHz. The input varies from -3.8 V to +3.8 V and has an average power of 30
mW. The required signal to noise ratio is 20 dB. The modulator produces a
binary coded output. Assume uniform quantization
1 Calculate the number of bits required er sample
2 Outputs of 30 such PCM coders ate time multiplexed. What is the mini-
mum required transmission bandwidth for the multiplexed signals.
Solution
The given value of signal to noise ratio is 20 dB
( S )W
Hence, ( NS )dB = 10 log10N = 20dB
S 2
N = 10 = 100
a)The signal to quantization ratio is given as,
S 3P.22v
N = x2 max
S
Herem, xmax = 3.8V, P=30mW and N = 100
3(30×10−3 ).22v
100 = (3.8)2

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