UNIT-I

INTRODUCTION TO DIGITAL COMMUNICATIONS

ADVANTAGES OF DIGITAL COMMUNICATIONS:
1. D.C is more rugged than A.C because it can with stand channel noise and distortion much
better as long as the noise and distortion are within limits such is not the case with analog
messages any distortion or noise, no matter how small, will distort the received signal and
since the transmitted signal is digital in nature, a large amount of noise interference may be
tolerated.
2. The greatest advantage of DC over AC is usage of regenerative repeaters.
3. Digital H/W implementation is flexible and permits the use of microprocessors, digital
switching and large scale integrated circuits.
4. Digital signals can be coded to yield extremely low error rates and high fidelity as well as
privacy.
5. It is easier and more efficient to multiplex several digital signals
6. DC is inherently more efficient than analog in realizing the exchange of SNR for Bandwidth.
7. Digital signal storage is relatively easy and inexpensive.
8. In D.C, reproduction of digital messages is extremely reliable without deterioration.
9. Digital communication systems are simpler and cheaper because of the advanced made in the
IC technologies. i.e the cost of digital H/W continues to halve every 2 or 3 years, while
performance or capacity doubles over the same time period.
10. Since in DC, channel coding is used, therefore the errors may be detected and
corrected in Receivers.

DISADVANTAGES OF DIGITAL COMMUNICATIONS:

1. Due to analog to digital conversion, the data rate becomes high; more transmission
bandwidth is required for digital communication.
2. Digital Communication needs synchronization in case of synchronous modulation.
 Fig 1.1

Figure 1.1 shows the block diagram of a digital communication system. In this diagram,
three basic signal-processing operations are identified

i. source coding
ii. channel coding
iii. modulation.

It is assumed that the source of information is digital by nature or converted into it by design.

In source coding, the encoder maps the digital signal generated at the source output into
another signal in digital form. The mapping is one-to-one, and the objective is to eliminate or
reduce redundancy so as to provide an efficient representation of the source output. Since the
source encoder mapping is one-to-one, the source decoder simply performs the inverse mapping
and there by delivers to the user destination a reproduction of the original digital source output.
The primary benefit thus gained from the application of source coding is a reduced bandwidth
requirement.

In channel coding, the objective is for the encoder to map the incoming digital signal into a
channel input and for the decoder to map the channel output into an output digital signal in such a
way that the effect of channel noise is minimized. That is, the combined role of the channel
encoder and decoder is to provide for reliable communication over a noisy channel. This provision
is satisfied by introducing redundancy in a prescribed fashion in the channel encoder and exploring
it in the decoder to reconstruct the original encoder input as accurately as possible. Thus, in source
coding, we remove redundancy, whereas in channel coding, we introduce controlled redundancy.

Clearly, we may perform source coding alone, channel coding alone, or the two together. In
the latter case, naturally, the source encoding is performed first, followed by channel encoding in
the transmitter as illustrated in fig 1.1.

In the receiver, we proceed in the reverse order; channel decoding is performed first,
followed by source decoding. Whichever combination is used, the resulting improvement in
system performance is achieved at the cost of increased circuit complexity.

As for modulation, it is performed with the purpose of providing for the efficient
transmission of the signal over the channel. In particular, the modulator (constituting the last stage
of the transmitter in fig. 1.1) operates by keying shifts in the amplitude, frequency, or phase of a
sinusoidal carrier wave to the channel encoder output. The digital modulation technique for so
doing is referred to as amplitude-shift keying, frequency-shift keying, or phase-shift keying,
respectively. The detector performs demodulation (the inverse of modulation). Thereby producing
a signal that follows the time variations in the channel encoder output (except for the effects of
noise).

The combination of modulator, channel, and detector, enclosed inside the dashed rectangle
shown in fig. 1.1, is called a discrete channel. It is so called since both its input and output signals
are in discrete form.

Traditionally, coding and modulation are performed as separate operations, and the
introduction of redundant symbols by the channel encoder appears to imply increased transmission
bandwidth. In some applications, however, these two operations are performed as one function in
such a way that the transmission bandwidth need not be increased. In situations of this kind, we
define the joint function of the channel encoder and modulator as the imposition of distinct
patterns on the transmitted signal, which are discernible by the combined action of the channel
decoder and detector in the receiver.
Sampling Theory:
Sampling theorem - It is a bridge between continuous time signals and discrete time signals. It
provides a mechanism for representing continuous - time signals by discrete time signals.
Sampling converts continuous time signals to discrete time signals and convert back to continuous
signals from discrete time signals.
Statement of sampling theorem:
i) A band limited continuous of finite energy, which has no frequency component higher than fm
Hz is completely described by its sample values at uniform intervals less than or equal to sec a
part.
ii) A band limited continuous of finite energy, which has no frequency component higher than fm
Hz may be completely recovered from the knowledge of its samples taken at the rate of 2fm,
samples per sec. Combining the two-parts, the sampling theorem may be started as “A continuous -
time signal may be completely represented in its samples and recovered back if the sampling freq
is fS ≥ 2fm”.

Here fS - is the sampling frequency and

fm -is the maximum frequency present in the signal.

Proof of sampling theorem:

To prove the sampling theorem we shall show that a signal whole spectrum is band limited
to , can be reconstructed exactly without any error from its samples taken uniformly at a rate
Hz. The two statements of S.T can be proved by using the convolution theorem. Let us
consider a constant time signal x (t) whose spectrum is band limited to Hz. i.e. x (t) has no
frequency components beyond Hz.

∴x (ω) = 0 for | ω |> where ω = 2π

The spectrum (F.T) of can be obtained by using frequency convolution

theorem.

∴
 -

Fig 1.2 Sampling Process

This operation yields F (ω), repeating itself every

radians per sec.
The spectrum ( ) can also be achieved analytically.

The periodic function can be written as the sum of impulse located at =0, ± , ±2 ,
- - - - - - -± m .

=δ( )+δ( - )+______+δ( - )+δ( + ) + _ _ _ _ _ _ _+ δ (

+ )+_______

∴ (m=1, 2, 3_ _ _ _ _)

and
↔
∴ ( )= )*
By using sampling property of a delta function
( )= - )

The summation represents F ( ) repeating every radiating per sec.

It is obvious from FS (ω), that F ( ω) will repeat periodically without overlapping, provided

≥2

(or)

=> T≤ sec → (a)

Where T is the uniform sampling internal

A related term sampling rate (or) sampling freq f0 = should meet the following condition

samples / sec. → (b)

equations (a) and (b) proves the sampling theorem.

Sampling of Band - pass signals:
Let us consider a more general case of band pass signal with ωH as upper cutoff frequency and ωL
as lower cutoff frequency, centered at ωC, shown in fig.

low pass signals are the special case of this band pass signal when ωC = 0
|H (ω)|
= max freq component = ωC +ω m

- - - 0

Fig 1.3 Band - pass signals

∴Minimun sampling rate = 2(ωC +ω m) = 2ωH - expected value but actually it is much less than
this value.
The rate is specified as follows
Case 1:- If either ωH (or) ωL is a harmonic of sampling frequency ωS, the min sampling rate is

ωS = 2 (ωH - ωL ) = 2 × 2 ωm =4 ωm

Case 2:- If ωL (or) ωH is not harmonic of sampling frequency ωS, then a more general sampling
condition is

Where m is the largest integer not exceeding

Problem 1.
The spectral range of a function extends from 10.0 to 10.2 MHz Find the min sampling rate and
the max sampling rate
Sol:-
fS = 2(fH - fL) = 2(10.2 - 10) = 2 x 0.2 M = 0.4 MHz

Since fL is the (10MHz) is the 25th harmonic of fS (i.e 0.4 MHz) 0.4 x 25 = 10 MHz

Hence the necessary condition needed for case 1 is satisfied

Sampling rate of 0.4 MHz is the desired answer

Corresponding sampling time = 2.5μsec

SAMPLING:
A continuous time signal is first converted to discrete - time signal

Sampling theorem:
The statement of sampling theorem can be given in two parts as
i) A band limited signal of finite energy, which has no frequency - component higher than
fm Hz is completely described by its sample values at uniform intervals less than or
equal to sec apart.

ii) A band limited signal of finite energy, which has no frequency - component higher than
fm Hz may be completely recovered from the knowledge of its samples taken at the rate of

2fm samples/sec combining the two parts, the sampling theorem may be stated as under: “A
continuous - time signal may be completely represented in its samples and recovered back if the
sampling freq is fS ≥ 2 fm. Here fS is the sampling freq and fm is max freq present in the signal”

Nyquist Rate: fS = 2fm - minimum sampling rate

Nyquist Interval: max sampling interval

Sampling process:-

Fig 1.4 basic sampler

Reconstruction Filter:- Low pass filter

A LPF is used to recover original signal from its samples. This is also known as interpolation filter.
Ideal LPF Practical LPF

Fig 1.5 Ideal and practical filters

Types of sampling techniques:

There are basically 3 types of sampling techniques
1) Ideal sampling / Instantaneous sampling / Impulse sampling
2) Natural sampling
3) Flat - top sampling
Ideal sampling:

G(f)

Fig 1.6 Ideal sampling process

g (t) = x (t) ×
 (i) G (f)
, (Nyquist
G (f) Sampling)

(iii) (Over Sampling)

G (f)

(i) (Nyquist Sampling)

(iii) (Under Sampling)

Fig 1.7 (i) Nyquist sampling (ii)Over sampling (iii)Under sampling

Hence the signal is under sampled in this case fS < 2 fm and some amount of aliasing is produced
in this under - sampling process.
 In fact, aliasing is the phenomenon in which a high freq component in the freq - spectrum of
the signal takes identity of a lower freq component in the spectrum of the sampled signal.
From fig (iii), it is obvious that because of the overlap due to aliasing phenomenon, it is not
possible to recover original signal x (t) from sampled signal by low - pass filtering since the
spectral components in the overlap regions add and hence the signal is distorted since any
information signal contains a large no. of freq so to decide a sampling freq is always a problem.
Therefore a signal is first passed through a LPF. This LPF blocks all the freq which are above fm
Hz. This process is known as band - limiting of the original signal. This LPF is pre-alias filter
because it is used to prevent aliasing effect. After band limiting it becomes easy to decide sampling
freq since the maximum freq is fixed at fm Hz.

In short, to avoid aliasing:

i) Pre-alias filter must be used to limit band of frequencies of the signal to fm Hz.

ii) Sampling frequency fS must be selected such that fS > 2 fm.

Natural sampling:
The ideal sampling is possible on in theory since it is impossible to have a pulse whose width
approaches zero. So natural sampling is practical method. In this, the pulses have a finite width
equal to τ.

The spectrum of a naturally sampled

signal is weighted by a SinC function.
G (f)

↔
τ ↔

Fig 1.8 Natural sampling process

 Flat top sampling (or) Rectangular sampling:
It is also practically possible method. But natural sampling is little complex where as it is quite
easy to get flat - top samples. In this flat top sampling, the top of the samples remains constant and
is equal to the instantaneous value of the baseband signal x (t) at the start of sampling.

P (t), constant width pulse

function with width τ and
Height 1

Flat-top sampled signal

g (t) = S (t) * P (t)
x (t)
g (t)

↔ 0
τ
 Fig 1.9 Flatop sampling process

The spectrum of g (t) is shown in above figure, which is obtained by multiplying S (f) with p (f).
As the p (f) value is different at different frequencies, the shape of C(f) is not similar to S(f) which
shows that a distortion will be introduced if the signal is recovered by an ideal low pass filter of a
cut - off freq fm Hz.

Aperture effect:-

i.e from above figure, it may be observed that by using flat top samples an amplitude
distortion is introduced in the reconstructed signal x(t) from g(t). In fact the high frequency roll -
off of H(f) acts like a LPF and this attenuates the upper portion of message signal spectrum. These
high frequencies of x(t) are affected. This type of effect is known as Aperture effect.
Now, as the duration „‟ of the pulse increases the aperture effect is more prominent.
Hence during reconstruction an equalizer is needed to compensate for this effect.

X (f).P (f)
TF=
X (f)
G (f) PAM
Message
Signal
Signal
S (f).P (f) g (t)
x (t)
Should be slightly higher
than the maximum It compensates for
frequency of message signal. Fig 1.20 Recovering x (t) at Receiver the aperture effect

P (f) = =
Comparison of 3 sampling techniques
Parameter Ideal/Instantaneous Natural Sampling Flat top Sampling
Sampling
1) Sampling It uses multiplication It was chopping It uses sample and
Principle Principle hold circuit
Sampling switch
C (t)
2) Generation
circuit x (t) g
x (t) C
g (t) x (t) g (t) Discharged
switch

x (t)
x (t) x (t) x (n )
3) Wave forms g (t) x (n )

4) Feasibility This in not a This method is used This method is also

practically possible practically used practically
method

5) Sampling It tends to infinity Sampling rate satisfies Sampling rate

Rate Nyquist criteria Satisfies Nyquist criteria
criteria
Problems:-
1. A signal x1(t) is band limited to 2KHz while x2(t) is band limited to 3KHz. Find the
Nyquist sampling rate for
a) x1 (2t) b) x2 (t-3) c) x1 (t) + x2(t) d) x1(t) . x2 (t) e) x1 (t) * x2 (t)

Sol:
a) The spectrum of x1 (2t) (time compression) stretches to 4 KHz. Thus the Nyquist rate is 8
KHz.
b) The spectrum of x2 (t-3) (time shift changes only phase) extends to 3 KHz. thus Nyquist
rate is 6 KHz.
c) The spectrum of x1 (t) + x2 (t) (sum of sprectra) extends to 3KHz thus the Nyquist rate is
6 KHz.
d) The spectrum of x1 (t) . x2 (t) (convolution in freq domain) extends to 5 KHz. Thus the
Nyquist rate is 10 KHz.
e) The spectrum of x1 (t) * x2 (t) (Product in freq domain) extends to 2 KHz. Thus the
Nyquist rate is 4 KHz.

2. m (t) = J Cos (1000t) Cos(4000t)

= 2.5 [Cos(3000t) + Cos (5000t)]

2π =3000π
2π =5000π

=2500

3. Given the signal m(t) = 10 Cos 2000πt Cos 8000πt

a) What is the minimum sampling rate based on the low pass uniform sampling
theorem?
b) Repeat (a) based on the band pass sampling theorem
Sol:-
a) m (t) = 10 Cos 2000πt Cos 8000πt
= 5 Cos (6000πt) + 5 Cos 10000πt
fmax = 5000 Hz = 5 KHz

fS = 2 fmax = 10 KHz

b) =5KHz

Based on the band pass sampling theorem = 5KHz

4. Specify the Nyquist rate and Nyquist interval for each of the following signals.
1) g(t) = Sin C (200t)
2) g(t) = Sin C2 (200t)
3) g(t) = Sin C (200t) + Sin C2 (200t)

Sol:- 1) g(t) = Sin C (200t) =

ω =200π => 2πf = 200π

Nyquist interval =

2) g (t) = sinc2(200t) =

= =

2πf = 400π => Hz

3) g(t) = Sin C (200t) + Sin C2 (200t)

The B.W of g(t) is determined by the highest freq component of Sin C (200t) or
Sin C2 (200t) max freq component
2πf = 400π => Hz
5. A signal m (t) = 4 Cos (60πt) + 2Cos (160πt) + Cos (280πt) is sampled at
i) 150 Hz ii) 75Hz iii) 300Hz.
Find the frequency components of the signal that appear at the o/p of an ideal LPF with
cutoff at 290Hz in each case. What is the Nyquist rate of sampling and Nyquist interval
for m(t)?

6. For the modulating signal m(t) = 2Cos (100t) + 18 Cos (2000πt). Determine the
allowable sampling rates and sampling intervals.
m (t) = 2 Cos (100t) + 18 Cos (2000πt)

2π =100π
2π

∴Sampling interval = = 0.5msec

7. A TV signal has a B.W of 4.5 MHz. Determine the sampling rate and sampling
intervals for
i) Minimum sampling
ii) 10% under sampling
iii) 20% over sampling

Sol:- Given band width fm = 4.5 MHz

i) Min sampling = fS = 2fm = 2 x 4.5 MHz = 9MHz

Sampling interval =

ii) 10% under sampling

100% − ?
=> so 9-0.9=8.1MHz
10% − ?
∴

Sampling interval =
 iii) 20% over sampling

100% − ?
=> =1.8
20% − ?
∴

8. A signal m(t) = 2 Cos (100πt) Cos (500πt) is ideally sampled at 700 Hz, and is sent
through an ideal LPF with cutoff at 650 Hz. Determine the frequency components in the
filter o/p. What changes will be there if the sampling is done at Nyquist rate?
Sol: a) m (t) = 2 Cos (100πt) Cos (500πt)
= 2 Cos (2π x 50t) Cos [2π (250)t]

The different frequencies present in the sampled signal are

Cut off freq of low pass filter is 650 Hz

The freq which can be present in the o/p are

b)

Nyquist rate =

∴ Then the different frequencies present in the sampled signal are

 =750Hz
=950Hz
fC of LPF = 650Hz

∴The freq components which can be present at the o/p of LPF are

∴ The only change is that instead of 650Hz we get 550Hz when the sampling is at Nyquist rate.

PULSE MODULATIONS
There are three types of pulse modulation systems

i) PAM
ii) PWM/PDM
iii) PPM
PAM(Pulse Amplitude Modulation): The amplitude of a carrier (a periodic train of rectangular
pulses) is varied in proportion to sample values of a message signal.
PDM (Pulse Duration Modulation): The pulse duration is varied in proportion to sample values
of the message signal
PPM (Pulse Position Modulation): The pulse position is varied in proportion to sample values of
the message signal.
Fig 1.21 Base band signal f(t), Carrier pulse train C(t), PAM signal, PWM signal, PPM signal
Comparison of various pulse analog modulation methods:

PAM PWM/PDM PPM

1) Waveform Waveform Waveform

2) Amplitude of the pulse Width of the pulse The relative position of

is proportional to amplitude is proportional to amplitude the pulse is proportional
of modulating signal of modulating signal to amplitude of modulating
signal
3) Width constant position to Amplitude constant position to Amplitude constant width
amplitude of modulating amplitude of modulating to amplitude of
signal signal modulating signal

4) The B.W of the transmission B.W of transmission channel B.W of

transmission
Channel depends on depends on rise time channel depends on rising
width of the pulse of the pulse time of the pulse

5) The instantaneous power The instantaneous The instantaneous power

of transmitter varies power of the transmitter of the transmitter remains
varies constant.

6) Noise interference is high Noise interference is Noise interference is

minimum minimum

7) System is complex similar to Simple to implement

Simple to implement
Amplitude modulation similar to frequency similar to phase.
Modulation modulation
PAM modulator circuit:

A PAM modulator circuit is shown in Fig.7.1.5. This circuit is a simple emitter follower. In
the absence of the clock signal, the output follows the input. The modulating signal is applied as
the input signal. Another input to the base of the transistor is the clock signal. The frequency of the
clock signal is made equal to the desired carrier pulse train frequency. The amplitude of the clock
signal is so chosen that the high level is at some negative voltage which is sufficient to bring the
transistor in the cut-off region. Thus, when the clock signal is high, the circuit behaves as an
emitter follower, and the output follows the input modulating signal. When the clock signal is low,
the transistor is cut-off and the output is zero. Thus the output wave form, shown in Fig.7.1.5 is the
desires pulse amplitude modulated signal.

Fig 1.22 PAM Modulator

Demodulation of PAM Signals:

Demodulation of natural sampled signal can be done with the help of an ideal low pass
filter with a cut-off frequency . But, for this, the pulse –top shape is to be maintained after
transmission. This is very difficult due to the transmitter and receiver noise.

Therefore, normally, flat-top sampling is preferred over natural sampling.

There are two demodulation methods for the flat-top sampled signal.

1).Using an Equalizer:-
If the flat-top sampled signal is passed through an ideal low pass filter, the spectrum of the
output will be F (ω) P (ω). The time function of the output is somewhat distorted due to the
multiplying factor P(ω). If the low pass filter output is passed through a filter having a
 transfer function 1/ P(ω) over the range 0 - , the spectrum at the output of this filter will
be F(ω)P(ω).
= F (ω), and hence, the original time function f (t) will be recovered. The filter with a

transfer function is known as equalizer.

The combination of an ideal LPF and equalizer is known as composite filter. The transfer
function H (ω) of this composite filter is shown in Fig.1.23(b). (It may be noted that the
transfer function of the equalizer outside can be chosen according to the convenience of
design).
H (ω) is given by
H (ω) = , <

= 0, otherwise

(a) (b)

Fig 1.23 Demodulation of flat top sampled PAM signal using equalizer

(a) Composite filter with transfer function H(ω) (b) The desired characteristics of H(ω)

(i) Using Holding Circuit:-

In this method, the received signal is passed through a holding circuit and a LPF, shown in
Fig 1.24 (a).
Fig 1.24 (b) shows a simple holding circuit. The switch S closes after the arrival of the pulse,
and it opens at the end of the pulse. The capacitor C gets charged to the pulse amplitude
value, and it holds this value during the interval between the two pulses. Thus, the sampled
values are held as shown in fig.1.24(c). The holding circuit output is smoothed in LPF as
shown in fig. 1.24(d). it can be seen that some distortion is introduced because of the holding
circuit. The circuit of fig 1.24(b) is known as zero order holding circuit, which considers
only the previous sample to decide the value between the two pulses. the first order holding
circuit considers the previous two samples; the second order holding circuit considers the
previous three samples; and so on. As the order of the holding circuit increases, the distortion
decreases at the cost of the circuit complexity. The amount of permissible distortion decides
the order of the holding circuit.
 Fig.1.24 Demodulation of flat top sampled PAM signal

7.2 PULSE TIME MODULATION

The two types of PTM systems, namely PWM and PPM, are shown in fig. 7.2.1. Figure
7.2.1(a) is the base band signal f(t) whereas fig. 7.2.1(b) is the carrier pulse train t. figure 7.2.1(c)
is the PWM signal where the width of each pulse depends on the instantaneous value of the base
band signal at the sampling instant. Figure 7.2.1(d) is the PPM signal where the shift in the
position of each pulse depends on the instantaneous value of the base band signal at the sampling
instant. It can be seen that in PWM, the information about the base band signal lies in the trailing
edge of the pulse, whereas, in PPM, it lies in both the edges of the pulse. (Although, basically it
lies in the leading edge, but since the width of the pulse is same always, the trailing edge also
carries the same information.)
 a)

b)

c)

d)

Fig 1.25 (a)Base band signal f(t), (b)Carrier pulse train fc(t), (c)PWM signal, (d)PPM signal

Generation of PTM signals

Indirect method:

The scheme of generation of the PTM signals is shown in Fig.1.26. Firstly, the flat-topped
PAM signals are generated as explained earlier. The synchronized ramp waveform shown in
fig.1.26 (b) is generated during each pulse interval. These two signals are added as shown in fig
1.26(c), and the sum is applied to a comparator circuit whose reference level is shown by a broken
line in Fig.1.26(c).

The second crossing of the comparator reference level by the wave form of fig.1.26(c) is
used to generate the pulse of constant amplitude and width as shown in Fig.1.26 (d) giving the
desired PPM waveform. The leading edge of the synchronized ramp of fig.1.26(b) is used to start a
pulse, and the trailing edge of the PPM waveform of Fig.1.26(d) is used to terminate the pulse, as
shown in Fig.1.26(e), giving the desired PWM waveform.

The ramp amplitude is so adjusted that it is slightly greater than the maximum variation in
amplitude of the PAM signals. The comparator reference level is such that it always intersects the
sloping of the waveform of fig.1.26(c).
 (e) PWM

Fig.1.26 Indirect method of generation of PTM signals

(a) PAM signal, (b) synchronized ramp, (c) PAM signal+ synchronized ramp, (d)PWM signal,
(e) PPM signal

Direct method:

In the direct method, the PTM waveforms are generated without generating the PAM waveform.
Here, the base band signal f(t) of Fig.1.27(a) and a ramp signal of Fig. 1.27(b), occurring at the
sampling instants, are added to give the waveform of Fig 1.27(c). This is compared in a
comparator whose reference level is shown in Fig. 1.27(c) by a broken line.
The PPM (Fig. 1.27 d) and PWM (Fig 1.27 e) waveforms are then obtained in the same manner as
explained in the indirect method of generation of PTM signals.

If we want to modulate the leading edge pulse in the PWM waveform, the ramp waveform shown
in fig.1.27 (a) should be used. For modulating both the edges of the pulses in the PWM waveform,
the ramp waveform shown in fig.1.27(b) needs to be used.

Fig. 1.27 direct method of generation of PTM signals

PWM Modulator circuit:

A PWM modulator circuit is shown in Fig. 1.28. The clock signal of the desired frequency
is applies as shown, from which the negative trigger pulses are derived with the help of a diode and
an combination which works as a differentiator. These negative trigger pulses are applied
to the pin no.2 of the 555 timer which is working in the monostable mode. They decide the starting
time of the PWM pulses. The end of the pulses depends on an combination, and on the
signal at pin no.5, to which the modulating signal is applied. Therefore, the width of the pulses
depends upon the value of the modulating signal, and thus the output at pin no.3 is the desired
pulse width modulated signal.
 Fig. 1.28 PWM Modulator

PPM Modulator circuit:

A PPM modulator circuit is shown in Fig.1.29. The PWM signal (which is obtained as
shown in Fig.1.29) is applied to pin no.2 through the diode and - combination. Thus the
input to pin.no.2 is the negative trigger pulses which correspond to the trailing edges of the PWM
waveform. The 555 timer is working in a monostable mode and the width of the pulse is constant
(governed by an combination).

The negative trigger pulses and, thus, the output at pin no.3 is the desired pulse position modulated
signal.

Fig. 1.29 PPM Modulator

Demodulation of PTM signals:

The PWM waveform of Fig. 1.30(a) is used to generate a ramp waveform as shown in Fig.
1.30(b). The leading edges of the PWM pulses start the ramp of same slope, and the trailing edges
of the PWM pulses terminate the ramp. The height attained by the ramp is, therefore, proportional
to the width of the PWM pulses. The height attained by the ramp is sustained for some time, thus
creating a porch, after which the voltage returns to its initial level.

A similar type of synchronized ramp is shown in Fig. 1.30(b). This is generated with the help of
the PPM pulses shown in Fig.1.30(a). Here, the ramp is initiated at the beginning of the time slot,
and it is terminated by the leading edge of the PPM pulse. Thus, the height attained by the ramp is
proportional to the displacement of the leading edge of the PPM pulses from the beginning of the
time slot. Here, too, the height attained by the ramp is sustained for some time, thus creating a
porch, and then it is returned to the initial level.

Fig. 1.30 Demodulation of PWM waveform

(a) PWM waveform,(b)Ramp waveform with Porch (c) Ramp waveform with locally
generated Pulse on Porch (d) PAM waveform
The remaining procedure for both PWM and PPM is same. A sequence of locally generated pulses
of a fixed amplitude are added to the synchronized ramp on the porch as shown in Figs.1.30(c) and

1.31(c).The lower portions in these two waveforms are clipped by a clipping circuit, with the
clipping level adjusted in such a way that it never crosses the ramp.

Fig. 1.31 Demodulation of PPM waveform

(a) PPM waveform,(b)Ramp waveform with Porch (c) Ramp waveform with locally generated
Pulse on Porch (d) PAM waveform

The output of the clipper is a PAM waveform (Fig.1.30(d) and Fig. 1.31(d)), from which the base
band signal can be recovered.
PWM Demodulator circuit:-

A PWM demodulator circuit is shown in Fig. 1.32. The transistor works as an inverter. Hence,
during the time interval A-B, when the PWM signal is high, the input to the transistor is low.
Therefore, during this time interval, the transistor is cut-off and the capacitor C gets charged
through an R-C combination. During the time interval B-C when the PWM signal is low, the input
to the transistor is high, and it gets saturated. The capacitor C then discharges very rapidly
through . The collector voltage of during the interval B-C is then low. Thus the waveform at
the collector of is more or less a saw-tooth waveform whose envelop is the modulating signal.
When this is passed through a second-order OP-AMP low pass filter, we get the desired
demodulated output.

Fig. 1.32 PWM Demodulator

PPM Demodulator circuit:-

A PPM demodulator circuit is shown in Fig.1.33. This utilizes the fact that the gaps between the
pulses of a PPM signal contain the information regarding the modulating signal. During the gap A
– B between the pulses, the transmitter is cut-off, and the Capacitor C gets charged through the R –
C combination. During the pulse duration B – C, the capacitor discharges through the transistor,
and the collector voltage becomes low. Thus, the waveform at the collector approximately a saw-
tooth waveform whose envelop is the modulating signal. When this is passed through a second
order OP-AMP low pass filter, we get the desired demodulated output.
 Fig. 1.33 PPM Demodulator

Bandwidth of PTM signals:

The bandwidth can be estimated by observing the spectrum of the PTM signals. The
spectral analysis for such waves is complicated and hence will not be treated in this text. We will
discuss only the qualitative features of the spectrum. The one-sided spectrum of a PWM signal is
shown in Fig.1.34.Assume that the modulating signal is a single-tone sinusoid of the frequency ,
and the sampling frequency is .

The PWM spectrum has the following frequency components.

(i) A d.c component at ω = 0, which represents the average value of the pulses.
(ii) The modulating frequency .
(iii) The harmonics of the sampling frequency .
(iv) Sidebands spaced .
 Fig. 1.34 One-sided Spectrum of PWM signal

The presence of harmonics of is due to the contribution of the unmodulated pulse train
which may be taken as the carrier of the PDM wave. Each harmonic of is associated with the
sidebands of an FM type. The sidebands of each extend to infinity outward, but with a decaying
magnitude. However, the useful message band is available in a band 0 - and hence a low pass
filter can be used to recover the message from PWM. But the output of LPF is distorted due to the
presence of cross – modulation terms that lie in the baseband. The lower sidebands of may
extend to lie in the message baseband to cause distortion. This can be prevented by restricting the
maximum excursion of the trailing edge of the PWM pulse.

The spectrum of a naturally sampled PPM wave for a single tone modulating signal has
form similar to that of a PDM wave, with the only difference that it contains a component
proportional to the derivative of the modulating signal in place of modulating component itself.
Therefore, the PPM direction can be achieved by an LPF followed by an integrator. An alternative
direction method is to convert PPM into PWM, and then pass it through an LPF. This provides
greater signal amplitude with less distortion in the receiver.
Multiplexing
Multiplexing may be defined as a technique which allows many users to share a common
communication channel simultaneously. There are two major types of multiplexing techniques.
They are

1. Frequency division multiplexing (FDM)

2. Time division multiplexing (TDM)

Frequency division multiplexing (FDM): -

This technique permits a fixed frequency band to every user in the complete channel
bandwidth. Such frequency slot is allotted continuously to that user. As an example considers that
the channel bandwidth is 1MHz. Let there be ten users, each requiring up to 100KHz bandwidth.
Then the complete channel bandwidth of 1MHz can be divided into ten frequency bands, i.e. each
of 100 KHz and every user can be allotted one independent frequency band. This technique is
known as Frequency Division Multiplexing (FDM). It is mainly used for modulated signal. This
is due to the fact that a modulated signal can be placed in any frequency band by just changing the
carrier frequency. However, at the receiver, these frequency multiplexed signals can be separated
by the use of tuned circuits (i.e. band pass filters) of their respective frequency band. And for every
band, there are independent tuned circuits and demodulators.

Time Division Multiplexing(TDM): -

As discussed earlier, in PAM, PPM and PDM the pulse is present for a short duration
and for most of the time between the two pulses, no signal is present. This free space between the
pulses can be occupied by pulses from other channels. This is known as Time Division
Multiplexing (TDM). Thus, time division multiplexing (TDM) makes maximum utilization of the
transmission channel.

Hence, we can say that in FDM, all the signals are transmitted simultaneously over the
same communication medium, and the signals occupy frequency slots. However, in TDM, the
signals to be multiplexed are transmitted sequentially one after the other. Each signal occupies a
short time slot as shown in figure. Thus, the signals are isolated from each other in the time
domain, but all of them occupy the same slot in the frequency spectrum. Therefore, in TDM, the
complete bandwidth of the communication channel is available to each signal being transmitted.
 Fig.1.35

At this stage, it may be noted that in context of TDM, we define one important term i.e.,
frame. One frame corresponds to the time period required to transmit all the signals once on the
transmission channel. This has been shown in fig 1.35. Here, we have total four message signals to
be transmitted. Hence, one frame will correspond to the time period required to transmit all the
four signals once on the channel.

The TDM system can be used to multiplex analog or digital signals; however it is more
suitable for the digital signal multiplexing.

PAM/TDM system: -

Now, let us discuss a PAM/TDM system. In fact, this system combines the concepts of
PAM and TDM both as shown in figure 1.36.

Fig.1.36 Block diagram of a PAM/ TDM system

Working Principle:

Here, the multiplexer is a single pole rotating switch or commutator. This switch can be a
mechanical switch or an electronic switch and it rotates as fS rotations per second. As the switch
arm rotates, it is going to make contact with the position 1, 2, 3 or N for a short time. There are N
analog signals, to be multiplexed, which are connected to these contracts. Hence, the switch arm
will connect these N input signals one by one to the communication channel.

The waveform of a TDM signal which is being transmitted has been shown in figure . It
shows that the rotary switch samples each message during each of its rotations.

Since, each rotation corresponds to one frame, therefore, one frame is completed in TS
seconds where TS= .

Hence the function of the commutator is two fold as under:

(i.)To take narrow sample of each input message at a rate fS which is higher than 2fm

(ii.)To sequentially interleave the N samples inside the interval, TS= .

Now, the multiplexed signal at the output of the commutator is applied to a pulse
amplitude modulator. It converts the PAM pulses into a form suitable for transmission over the
communication channel. The input message signals are passed through low pass filters before
applying them to the commutator. These filters are actually the anti-aliasing filters which avoid the
aliasing. The cutoff frequency of each low pass filter (LPF) is fm Hz.

At the receiving end of PAM/TDM system, the received signal is applied to a pulse
amplitude demodulator which performs the reverse operation of pulse amplitude modulator. At the
receiver, there is one more rotating switch or decommutator used for demultiplexing. It will be
interesting to know that this switch must rotate at the same speed as that of the commutator at the
transmitter and its position must be synchronized with commutator at the transmitter and its
position must be synchronized with commutator in order to ensure proper demultiplexing. The low
pass filters (LPFs) on the receiver side are used for the reconstruction of the original message
signals.
Evaluation of signaling Rate in a PAM/TDM system:

As a matter of fact, the signaling rate of a TDM system is defined as the number of pulses
transmitted per second. It is represented by r. Let us now derive an expression for the signaling
rate of the PAM/TDM system in the form of following few points:

(i.) Let fm=maximum frequency of all the input signals x1 to xN .

(ii.) Therfore, as per Nyquist criterion, the sampling frequency fS 2fm. Hence, the speed of

rotation of commutators is fS rotating per second with fS 2fm.

(iii.)As shown in figure, one revolution of commutators corresponding to one frame contains one

sample from each input signal.

Hence,

1 revolution = 1 frame = N pulses

(iv.)One frame period is i.e., TS seconds. Therefore, in TS seconds, N number of pulses are

transmitted. Hence, the pulse to pulse spacing within the frame is given by,

Pulse to pulse spacing= =

(v.)As the period of one pulse (ON+OFF) is seconds, the number of pulses per second is

given by,

Number of pulses per second=

This is nothing but the signaling rate.

Therefore, signaling rate of a TDM system=r = pulses/second.

But as fS 2fm. therefore,

Signaling rate of a TDM system= r 2Nfm pulses per second.

It may be noted that TDM system is supposed to have its signaling rate as high as possible.
It is evident from the expression above that the signaling rate can be increasing the sampling rate fS
and / or the number of input signals N.
Transmission Bandwidth of a PAM/TDM channel: -

The minimum transmission bandwidth of a PAM-TDM channel is given by

Band Width = (signaling rate)

Therefore, minimum transmission bandwidth BW

Hence, minimum transmission bandwidth BW=

Synchronization in PAM/TDM system: -

As a matter of fact, the multiplexed PAM signals can be received properly if and only if the
transmitter and receiver commutators and synchronized to each other in terms of the speed and the
position. In order to ensure synchronization, a marker pulse is introduced at the end of each frame
in the transmitted signals as shown in fig 1.37.

Fig 1.37 illustration of frame synchronization and detection

 The amplitude of this pulse is kept higher than the maximum permissible amplitude of the
multiplexed channels. At the receiver end, the received signal is compared with a DC reference
level. The comparator responds to only the marker pulse to produce output. Thus, the marker pulse
is separated from the remaining multiplexed channels. Due to the introduction of synchronizing
pulse, only three signals instead of four can now be transmitted.

Advantages of TDM

(i.) Full available channel bandwidth can be utilized for each channel.

(ii.) Intermodulation distortion is absent.

(iii.) TDM circuitry is not very complex.

(iv.) The problem of crosstalk is not severe.

Disadvantages of TDM

(i.) Synchronization is essential for proper operation.

(ii.) Due to slow narrowband fading, all the TDM channels may get wiped out.

A PCM-TDM System (T1 Carrier System)

When a large number of PCM signals are to be transmitted over a common channel,
multiplexing of these PCM signals is required.Figure 1.38 shows the basic time division
multiplexing scheme, called as the T1-digital system or T1 carrier system. This system is used to
convey multiple signals over telephone lines using wideband coaxial cable.
 Fig 1.38 Digital hierarchy

Working Operation of the T1 Carrier system:-

The working operation of the PCM-TDM system shows in fig1.38 can be explained in the form of
few points as under:

i). This sytem has been designed to accommodate 24 voice channels marked done at a standard
rate of 8KHz. This is higher than the Nyquist rate. The sampling is done by the commutator

switch
ii). These voice signals are selected one by one and connected to a PCM transmitter by the

commutator switch

iii). Each sampled signal is then applied to the PCM transmitter which converts it into a digital

signal by the process of A to D conversion and companding, as explained earlier.

iv). The resulting digital waveform is transmitted over a co-axial cable.

v). Periodically, after ever 6000ft, the PCM-TDM signal is regenerated by amplifiers called
“Repeaters”. They eliminate the distortion introduced by the channel and remove the
superimposed noise and regenerate a clean PCM-TDM signal at their output. This ensures that
the received signal is free from the distortions and noise.
vi).At the destination, the signal is companded, decoded and demultiplexed, using a PCM
 receiver. The PCM receiver output is connected to different low pass filters via the

decommutator switch

vii).Synchronization between the transmitter and receiver commutators and is essential

in order to ensure proper communication.

Now, let us discuss few important terms related to a T1 carrier system as under:

(i) Bits/Frame:

The commutators sweep continuously from and back to at the rate of 8000 revolutions
per second. This will generate 8000 samples per second of each signal ( . Each sample is
then encoded (converted) into an eight bit digital word.

Thus, the digital signal generated during one complete sweep (revolution) of the
commutator is given by

1 frame=1 revolution=24 channels=24×8bits=192

One frame of T1 carrier system is shown in figure 8.18. Each voice signal from is
encoded into eight bits. One frame corresponds to the time corresponding transmission of each
signal once. Hence 1-frame corresponds to one-revolution of the commutator.

(ii) Frame Synchronization:

As discussed earlier, the synchronization between the transmitter and receiver commutator
is essentially. Without such synchronization, the receiver cannot know which bits correspond to
which of the original signals. To provide such synchronization, an extra bit is transmitted
preceding the 192 bits carrying the information in each frame, as shown in figure 8.19. This bit is
called as the frame synchronizing bit “F”. Thus, one frame synchronizing bit is transmitted per
frame. This makes the total number of bits per frame to be 193. The time slots for the 24 signals
and the extra frame synchronizing bit is as shown in figure 1.39.
 Fig 1.39 Frame Synchronization

Further, twelve successive F slots are used to transmit a 12 bit code. The code is 1101 1100 1000.
This code is transmitted repeatedly once every 12 frames and it is used at the receiver to obtain
synchronization.

(iii) Bit Rate

Bit rate means number of bits transmitted by a system per second. In the T1 carrier system,
each signal is sampled 8000 times per second, therefore, 1 frame (1 revolution of commutator) =
1/8000=125μsec.

Also, number of bits in 1 sec =

So, bit rate of T1 carrier system=1.544Mbits/sec.

(iv) Band width of T1 Carrier System:

Minimum bandwidth BW=

Duration of each bit can be found as follows:

Since 193=125μs

Therefore, 1bit=(125/193) μs=0.6476 μs

(v) Channel Associated Signaling:

When the PCM-TDM system is being used for the telephony. It is expected to transmit certain
signaling and supervisory signals along with the speech information. The signaling information
consists of the signals such as a call is being initiated or a call is being terminated, or the address of
calling party etc. In analog system such as signaling information is transmitted over a a separate
channel other than the voice channel. But in the T1 carrier system which is a digital system, a
separate channel is not used. In this system, the signaling information is sent using the same data
bit slots which are used to send the voice information. The technique used is “bit slot sharing”. In
the “bit slot sharing” method, for the first five frames, all the 24 channels are encoded into an 8 bit
digital code. However in the sixth frame, all the channels are coded into a 7 bit code and the LSB
(least significant bit) of each channel is used to transmit the signaling information. This is called as
“channel associated signaling”. This pattern is repeated after every six frames.