DIGITAL COMMUNICATION

III B.Tech-V SEM

20EC503
 UNIT-I
PULSE MODULATION: Introduction,Sampling Process(ideal and flat top),
Pulse-Amplitude Modulation, Pulse-Position Modulation, Quantization
Process, Quantization Noise, Pulse Code Modulation: Encoding,
Regeneration, Decoding, Delta Modulation, Differential Pulse Code
Modulation, Line Codes.
INTRODUCTION
Source
It generates message to be transmitted. Examples are the data from
computers, text data or tele type data. The source can be
an analog signal. Example: A Sound signal

Input Transducer
This is a transducer which takes a physical input and converts it to an
electrical signal (Example: microphone). This block also consists of
an analog to digital converter where a digital signal is needed for further
processes. A digital signal is generally represented by a binary sequence.

Source Encoder
The source encoder compresses the data into minimum number of bits.
This process helps in effective utilization of the bandwidth. It removes
the redundant bits unnecessary excess bits,i.e.,zeros
Channel Encoder
The channel encoder, does the coding for error correction. During the
transmission of the signal, due to the noise in the channel, the signal
may get altered and hence to avoid this, the channel encoder adds
some redundant bits to the transmitted data. These are the error
correcting bits.

Digital Modulator
The signal to be transmitted is modulated here by a carrier. The
signal is also converted to analog from the digital sequence, in order
to make it travel through the channel or medium.

Channel
The channel or a medium, allows the analog signal to transmit from
the transmitter end to the receiver end.
Digital Demodulator
This is the first step at the receiver end. The received signal is
demodulated as well as converted again from analog to digital. The
signal gets reconstructed here.
Channel Decoder
The channel decoder, after detecting the sequence, does some error
corrections. The distortions which might occur during the transmission,
are corrected by adding some redundant bits. This addition of bits helps
in the complete recovery of the original signal.
Source Decoder
The resultant signal is once again digitized by sampling and quantizing
so that the pure digital output is obtained without the loss of
information. The source decoder recreates the source output.
Output Transducer
This is the last block which converts the signal into the original
physical form, which was at the input of the transmitter. It converts the
electrical signal into physical output (Example: loud speaker).
Sampling:-
Sampling is a process where an analog Signal is
converted into a corresponding Sequence of
Samples that are Usually spaced uniformly in
time.
i.e. process of converting Continuous time Signal
into discrete time Signal.
There are two types of Sampling :
1) Ideal Sampling or Impulse Sampling or
Instantaneous Sampling
2) practical Sampling.
i) Natural Sampling & chopper Sampling
ii) Flat Top Sampling or Sample & Hold Sampling.
• The following figure indicates a continuous-time signal ( ) and a sampled
signal ( ). When ( ) is multiplied by a periodic impulse train, the
sampled signal ( ) is obtained.
Sampling frequency is the reciprocal of the sampling
period. This sampling frequency, can be simply called
as Sampling rate. The sampling rate denotes the number of
samples taken per second, or for a finite set of values
Nyquist Rate:
Suppose that a signal is band-limited with no frequency
components higher than Hertz. That means, is the
highest frequency. For such a signal, for effective
reproduction of the original signal, the sampling rate
should be twice the highest frequency.
Which means,
fs = 2W
Where,
fs is the sampling rate
W is the highest frequency
This rate of sampling is called as Nyquist rate.
Statement:

A continuous time signal can be represented in its samples and can be

recovered back when sampling frequency fs is greater than or equal to the
twice the highest frequency component of message signal. i. e.
fs≥2fm
Proof:
Consider a continuous time signal x(t).
The spectrum of x(t) is a band limited to fm Hz i.e. the spectrum of x(t) is
zero for |ω|>ωm.

Sampling of input signal x(t) can be obtained by multiplying x(t) with an

impulse train δ(t) of period Ts.

The output of multiplier is a discrete signal called sampled signal which

is represented with y(t) in the following diagrams:
Here, you can observe that the sampled signal takes the period of
impulse.
The process of sampling can be explained by the following mathematical
expression:
 Sampling
 Process of converting analog signal into discrete signal.
 Sampling is common in all pulse modulation techniques .
 The signal is sampled at regular intervals such that each sample is
proportional to amplitude of signal at that instant.
 Analog signal is sampled every s , called sampling interval.
 = 1/ is called sampling rate or sampling frequency.
 =2 is Min. sampling rate called Nyquist rate.
 Sampled spectrum ( ) is repeating periodically without overlapping.
 Original spectrum is centered at =0 and having bandwidth of .
 Spectrum can be recovered by passing through low pass filter with
cut-off .
 For < 2 sampled spectrum will overlap and cannot be recovered
back. This is called as Aliasing.
Sampling can be performed with
the help of sampler.
Sampling methods:
1. Ideal – An impulse at each
sampling instant.
2. Natural – A pulse of Short width
with varying amplitude.
3. Flat Top – Uses sample and
hold, like natural but with single
amplitude value.
Impulse sampling is not used for
information exchange as it has very
low power.
Flat top sampling is most widely
used due to its noise mitigation
capability
 Impulse Sampling

The instantaneous sampling results in the samples whose width approaches zero. Due to this, the power
content in the instantaneously sampled pulse is negligible. Thus, this method is not suitable for
transmission purpose.
Flat Top Sampling or Rectangular Pulse Sampling
 Flat top sampling like natural sampling is also a practically possible sampling method.
But natural sampling is little complex whereas flat top sampling is quite easy.
 In flat-top sampling or rectangular pulse 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.
 The duration or width of each sample is τ and sampling rate is equal to fs = 1 / Ts.
 Figure shows the functional diagram of a sample and hold circuit which is used to
generate the flat top samples.

Fig.:5(a) A sample and hold circuit to generate

flat top samples (b) A general waveform of flat
top sampling.
Fig. 5(b) shows the general waveform of flat
top samples. From fig.5(b), it may be noted
that only starting edge of the pulse represents
instantaneous value of the baseband signal
x(t).

Also the flat top pulse of g(t) is

mathematically equivalent to the convolution
of instantaneous sample and a pulse h(t) as
depicted in fig.6.
 Figure 6:Convolution of any function with delta function is equal to that function

• This means that the width of the pulse in g(t) is determined by the width of h(t) and the
sampling instant is determined by delta function.
• In fig. 5(b), the starting edge of the pulse represents the point where baseband signal is
sampled and width is determined by function h(t). Therefore, g(t) will be expressed as,

Now, from the property of delta function, we know that for any function f(t)
 This property is used to obtain flat top samples.
 It may be noted that to obtain flat top sampling, we are not applying the above equation
directly here i.e., we are applying a modified form of the above equation.
 Thus, in this modified equation, we are taking s(t) in place of delta function δ(t).
 Observe that δ(t) is a constant amplitude delta function whereas s(t) is a varying
amplitude train of impulses. This means that we are taking s(t) which is an
instantaneously sampled signal and this is convolved with function h(t).
 Therefore, on convolution of s(t) and h(t), we get a pulse whose duration is equal to h(t)
only but amplitude is defined by s(t).
 Now, we know that the train of impulses may be represented mathematically as,
7
Fig.8 : (a) Baseband signal x(t), (b)
Instantaneously sample signal s(t), (c)
Constant pulse width function h(t), (d)
Flat top sampled signal g(t) obtained
through convolution of h(t) and s(t)
 PAM GENERATION
• In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced pulses are

varied in proportion to the corresponding sample values of a continuous message signal (

i.e the message signal is multiplied by a periodic train of pulses)

• The pulses can be of rectangular form or some other appropriate shape.

• The waveform of a PAM signal is illustrated in Figure.

• 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=1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample, so that it occupies some finite value T.
 The LPF at the beginning is placed in order to avoid aliasing of the samples.
 The LPF passes only the low-frequency component of the signal and eliminates the high-
frequency signal component.
 The output of LPF is then provided to a modulator, where it gets mixed with the
rectangular pulse train.
 Basically, the pulsed carrier gets modulated by the message signal here. The rectangular
carrier pulse is generated by the pulse generator circuit.
 The modulator generates a pulse amplitude modulated signal.
 The sampled pulses can be achieved either by natural or flat top sampling.
 The output of the modulator is provided to the pulse reshaping circuit which basically
shapes the pulses so that it can be easily detected at the receiver.
 Aperture Effect:

(1)

 This equation shows that the signal g(t) is obtained by passing the signal s(t)
through a filter having transfer function H(f).
 The corresponding impulse response h(t) in time-domain has been shown in
fig.7 (a)
 Each sample of x(t) [i.e., s(t)] is convolved with this pulse.
 Equation (1) represents that the spectrum of this rectangular pulse is
multiplied with that of s(t).
 Fig.7(b) shows the spectrum of one rectangular pulse of h(t).
 We know that the spectrum of a rectangular pulse is expressed as

(2)
• Hence, from fig.7(b), 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 low-pass filter and thus 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 ‘τ’ of the pulse increases, the aperture effect is more prominent.
Hence, during reconstruction an equalizer is needed to compensate for this effect.
• As shown in figure, the receiver contains a low-pass reconstruction filter with cutoff
frequency slightly higher than the maximum frequency present in the message signal.
The equalizer compensates for the aperture effect. It also compensates for the
attenuation by a low-pass reconstruction filter.

X(f).H(f)

(LPF)
filter

Figure: Recovering the message signal m(t) from the PAM signal s(t)
 From equation (2), it may be noted that the sample function h(t) acts like a low-pass
filter where Fourier transform is expressed as,

 Equalizer used in cascade with the reconstruction filter has the effect of decreasing
the in band loss of the reconstruction filter as the frequency increases in such a way
as to compensate for the aperture effect.
 Also, the transfer function of the equalizer is expressed as,

Here ‘td’ is known as the delay

introduced by low-pass filter which is
equal to τ/2. Therefore

which is the transfer function of an equalizer.

NOTE: The amount of equalization needed in practice is usually small.
 Indeed, for a duty cycle T/Ts  0.1,The amplitude distortion is less than 0.5 percent, in
which case the need for equalization may be omitted altogether.

Transmission Bandwidth in PAM

In a pulse amplitude modulated (PAM) signal the pulse duration ‘τ’ is considered to be
very small in comparison to time period (i.e., sampling period) Ts between any two
samples i.e., (1)

Now, if the maximum frequency in the modulating signal x(t) is fm,

then according to sampling theorem, the sampling frequency fs must
be equal to or higher than the Nyquist rate, i.e.
ADVANTAGES:
 It is a simple process for both modulation and demodulation.
 Transmitter and receiver circuits are simple and easy to construct.
 PAM can generate other pulse modulation signals and can carry the message at the
same time.
 The data can be transmitted quickly, efficiently, and effectively through usual copper
wires in high volume.

DISADVANTAGES:
 Bandwidth should be large for transmission PAM
modulation.
 Noise will be great.
 Pulse amplitude signal varies so the power required for
transmission will be more.
 For transmitting PAM signal, BW must be large.
Applications of PAM
• It is used in Ethernet communication.
• It is used in many micro-controllers for generating control signals.
• It is used in Photo-biology.
• It is used as an electronic driver for LED lighting.
 PULSE-WIDTH MODULATION (PWM)
• In pulse-duration modulation (PDM),the samples of the message signal are used to vary
the duration of the individual pulses. This form of modulation is also referred to as pulse-
width modulation or pulse-length modulation.
 Pulse Position Modulation (PPM)
 In PPM, the amplitude and width of the pulses is kept constant but the position of
each pulse is varied in accordance with the amplitudes of the sampled values of the
modulating signal.
 The position of the pulses is changed with respect to the position of reference
pulses.
 The PPM pulses can be derived from the PWM pulses as shown in fig.1. Here, it
may be noted that with increase in the modulating voltage the PPM pulses shift
further with respect to reference.
 The vertical dotted lines drawn in fig.1 are treated as reference lines to measure the shift
in position of PPM pulses.
 The PPM pulses marked 1, 2 and 3 in fig.1 go away from their respective reference lines.
This is corresponding to increase in the modulating signal amplitude.
 Then, as the modulating voltage decreases, the PPM pulses 4, 5, 6, 7 come progressively
closer to their respective reference lines.

Fig.1 : PPM pulses generated from PWM signal

 Generation of PPM Signal
The PPM signal can be generated from PWM signal as shown in fig.2 (a).

 The PWM pulses obtained at the comparator output are applied to a monostable
multivibrator. The monostable is negative edge triggered.
 Hence, corresponding to each trailing edge of PWM signal, the monostable output
goes high.
 It remains high for a fixed time decided by its own RC components.
 Thus, as the trailing edges of the PWM signal keep shifting in proportion with the
modulating signal x(t), the PPM pulses also keep shifting, as shown in fig.3.
 Demodulation of PPM Signal
The PPM demodulator block diagram has been shown in fig.4 .

 The operation of the demodulator circuit

may be explained as under:
 The noise corrupted PPM waveform is
received by the PPM demodulator circuit.
 The pulse generator develops a pulsed
waveform at its output of fixed duration and
applies these pulses to the reset pin (R) of a
SR flip-flop.
 A fixed period reference pulse is generated
from the incoming PPM waveform and the
SR flip-flop is set by the reference pulses.
 Due to the set and reset signals applied to
the flip-flop, we get a PWM signal at its
output.
 The PWM signal can be demodulated using
the PWM demodulator.
Advantages:
1. PPM has the most power efficiency as compared to other modulations.
2. This modulation has less stable amplitude noise interference.
3. This modulation separates the signal easily from a noisy signal.
4. It has constant transmitted power output.
5. It needs extremely less power as compared to PAM & PDM because of amplitude
& short duration pulse.
6. Easy noise removal & separation is extremely easy in this type of modulation.
7. Power utilization is also extremely low as compared to other modulations because
of stable pulse amplitude & width.
8. PPM communicates only simple commands from a Tx to an Rx, so it is frequently
used in lightweight applications because of its low system necessities.
Disadvantages
1. PPM is very complex.
2. It needs more bandwidth for transmission as compared to PAM.
3. It is extremely sensitive to multi-pathway interference like echoing that can disturb a
transmission by changing the difference in arrival times of every signal.
4. Synchronization is necessary between transmitter & receiver which is not feasible each time &
we require a dedicated channel for it.
5. Special devices are required for this kind of modulation.

Applications
1. The PPM is mainly used in telecommunication systems & air traffic control systems.
2. This modulation is used in radio control, an optical communication system & military
applications.
3. This technique is used in planes, remote-controlled cars, trains, etc.
4. PPM is used in noncoherent detection wherever a receiver does not require any Phase lock
loop or PLL to track the carrier’s phase.
5. It is used in RF (radio frequency) communication.
6. It is also utilized in high-frequency, contactless smart cards, radio frequency ID tags, etc.
Difference Between PAM, PWM, and PPM
In continuous-wave (CW) modulation, some parameter of
a sinusoidal carrier wave is varied continuously in
accordance with the message signal.

In pulse modulation, some parameter of a pulse train is

varied in accordance with the message signal.

In this context, we may distinguish two families of pulse

modulation, analog pulse modulation and digital pulse
modulation, depending on how the modulation is
performed.
In analog pulse modulation, a periodic pulse train is used
as the carrier wave, and some characteristic feature of each
pulse (e.g., amplitude, duration, or position) is varied in a
continuous manner in accordance with the corresponding
sample value of the message signal.

Thus, in analog pulse modulation, 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 represented in a form that is discrete in
both time and amplitude, thereby permitting its
transmission in digital form as a sequence of coded pulses.
Quantization:
The process of converting continuous amplitude levels into
discrete amplitude levels is called as quantization

Quantization process:- The process of representing m(kTs) which is

sampled value of m(t) at t=kTs is represented by finite set of
representation levels.

Let m(t) is a analog input message signal m(kTs) represent the

sampled amplitude of m(t) at t=kTs.
 For uniform Quantization Process

Where k= 1,2,3,…, L
 Total voltage range is divided into q equal intervals of
step size S

where VH = Max. voltage value

VL = Min. voltage value
Draw mid lines representing quantization levels
Assign binary codes (pre-defined) to each quantization level
Calculate quantization error
A 2-bit PCM modulator is used with a 0-1 V signal.
What is the binary digital value that will occur for the
following inputs: 0.4 V, 0.78 V. What is the
quantization error for these two samples?
Quantizing 0.4 V sample value: See that level 2 is near to 0.4
V. So, for sample voltage 0.4 V, 01 code is transmitted.
Quantization error e = 0.4 – 0.375 = 0.025 V

Quantizing 0.78 V sample: Level 4 is nearest to 0.78 V. So,

digital code 11 is transmitted.
Quantization error e = 0.875 -0.78 = 0.095 V
The major steps involved in PCM is
Sampling
Quantizing
Encoding
In pulse code modulation, the analog message signal is first
sampled, and then the amplitude of the sample is approximated to
the nearest set of quantization level.

This allows the representation of time and amplitude in a

discrete manner. Thereby, generating a discrete signal.

This discrete signal is then converted into its binary form for
the transmission of the signal.
It is basically composed of a transmitter, a transmission path and a
receiver.

The transmitter performs the sampling, quantizing and encoding

of the signal.

The transmission path includes regenerative receivers that recover

the signal from the undesired noise effects.

The receiver section that performs decoding of the coded signal

after regeneration of the signal at the receiver
The figure below shows the sampling of analog signal and further quantization of the samples
Non-Uniform Quantization:
In non-uniform quantization, the step size is not fixed. It
varies according to certain law or as per input signal
amplitude.
The following fig shows the characteristics of Non
uniform quantizer.
 Companding PCM System:
• Non-uniform quantizers are
difficult to make and expensive.
• An alternative is to first pass the
speech signal through
nonlinearity before quantizing
with a uniform quantizer.
• The nonlinearity causes the
signal amplitude to be
compressed.
• The input to the quantizer will
have a more uniform
distribution.
• At the receiver, the signal is
expanded by an inverse to the
nonlinearity.
• The process of compressing and
expanding is called
Companding.
In Delta Modulation the step size is constant , so that its
slope overload distortion and granular noise both can not
be controlled.

This drawback is overcome by Adaptive delta modulation

,in which the step size is variable.
Slope overload distortion will take place if slope of sine wave is greater than slope of delta
modulator
Differential pulse code modulation is a technique of analog to
digital signal conversion. This technique samples the analog
signal and then quantizes the difference between the sampled
value and its predicted value, then encodes the signal to form a
digital value.
Before going to discuss differential pulse code modulation, we
have to know the demerits of PCM (Pulse Code Modulation).
The samples of a signal are highly correlated with each other.
The signal’s value from the present sample to the next sample
does not differ by a large amount.
The adjacent samples of the signal carry the same information
with a small difference.
When these samples are encoded by the standard PCM system,
the resulting encoded signal contains some redundant
information bits. The below figure illustrates this.
The above figure shows a continuing time signal x(t) denoted by a
dotted line.
This signal is sampled by flat-top sampling at intervals Ts, 2Ts,
3Ts…nTs.
The sampling frequency is selected to be higher than the Nyquist
rate.
These samples are encoded by using 3-bit (7 levels) PCM.
The samples are quantized to the nearest digital level as shown by
small circles in the above figure.
The encoded binary value of each sample is written on the top of
the samples.
Just observe the above figure at samples taken at 4Ts, 5Ts, and
6Ts are encoded to the same value of (110).
This information can be carried only by one sample value. But
three samples are carrying the same information means redundant.
Now let consider the samples at 9Ts and 10Ts, the difference
between these samples only due to the last bit and first two bits
are redundant since they do not change.
So in order to make the process this redundant information
and to have a better output.
It is an intelligent decision to take a predicted sampled value,
assumed from its previous output and summarize them with the
quantized values.
Such a process is called a Differential PCM (DPCM)
technique.
PCM-Each and every sample is encoded and
transmitted

DPCM-Difference between samples is encoded and

transmitted..so that some band width is saved.

DM-Difference between samples is encoded to

only one bit and transmitted..so that more band
width is saved.

ADM-Variable step size