PULSE MODULATION

INTRODUCTION
 Many Signals in Modern Communication Systems are digital . Also, analog signals
are transmitted digitally.
 Reduced distortion and improvement in signal to noise ratios.
 PAM, PWM , PPM , PCM and DM.
 Data transmission, digital transmission, or digital communications is the physical transfer
of data (a digital bit stream or a digitized analogue signal) over a point-to-point or point-to-
multipoint communication channel.

Ex: optical fibers, wireless channels, computer buses....

 ELEMENTS OF DIGITAL COMMUNICATION SYSTEMS

Fig.Block diagram of Digital Communication

1. Discrete Information Source: It generates message to be transmitted. Examples are the data
from computers, text data or tele type data.

2. Source Encoder: It assigns codes to the symbols (samples) generated from discrete
information source. The code word having n number of bits. Each distinct sample having

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distinct(unique) code word. If code word length is 8 bit(n), we can have 256 distinct
symbols(ie.,2^n).

3. Channel Encoder: We know that channel is the major source of notice due to that there are
more chance of getting errors while propagating through channel. To avoid that channel
encoding is required. In that extra bits are added to the binary sequence generated by the
source encoder. These extra bits are called as redundant bits. These bits are defined with
proper logic. The redundant will be helpful to detect the errors at the receiver bit sequence.

4. Digital Modulator: In digital modulator the message signal is digital data and carrier is
analog one, in most cases we use sinusoidal waves. Some examples are
ASK,FSK,PSK.MRI techniques.

5. Channel: It provides the link between transmitter and rceiver. Channel may be wired
or wireless channel.

 Problems associated with channel:

1. Addictive Noise: This noise is occur due to internal solid state devices or resistors used
in channel.

2. Ampltude and Phase Distortion: This noise is occurred due to non-linear characteristics of the
channel.

3. Attenuation: This is due to internal resistance of the channel.

6. Demodulator: This device is used to detect the digital message signal from the
modulated signal.

7. Channel Decoder: This is used to detect and correct the errors that occur in the digital
message signal.

8. Source Decoder: This produces the sampling signal from the given digital message signal.

9. Destination: The sampled signal is converted into audio signal or video signal or any text
signal depending on the signal.

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 Fig. Basic block diagram of an A/D converter

Advantages of digital communication systems

1. Easy way of transmission of signals

2. Connection of more calls through one channel i.e., Multiplexing is possible using Digital
Communication.
3. Source Encoding and Channel Encoding can be used to detect errors at the
received signal.

4. Using repeaters between source and destination, we can reproduce the original
signal with less distortions.
5. Security is the major advantage of digital communication compared to Analog
Communication.
6. Transmitting analogue signals digitally allows for greater signal processing capability.
7. Digital communication can be done over large distances through internet and
other things.
8. The messages can be stored in the device for longer times, without being damaged.
9. Advancement in communication is achieved through Digital Communication.

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 Disadvantages of digital communication systems

1. Sampling Error
2. Digital communications require greater bandwidth than analogue to transmit the
same information.
3. The detection of digital signals requires the communications system to be
synchronized, whereas generally speaking this is not the case with analogue systems.
4. Digital signals are often the approximation of voice signals, ie, we don‟t get the
exact analogue signal.

 TYPES OF MODULATION – TREE DIAGRAM

In Continuous Wave modulation schemes some parameter of modulated wave varies

continuously with message.
In Analog pulse modulation some parameter of each pulse is modulated by a particular sample
value of the message.

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 Pulse modulation of two types
1. Analog Pulse Modulation
 Pulse Amplitude Modulation (PAM)
 Pulse width Modulation (PWM)
 Pulse Position Modulation (PPM)
2. Digital Pulse Modulation
 Pulse code Modulation (PCM)
 Delta Modulation (DM)

1. Analog Pulse Modulation

Analog pulse modulation results when some attribute of a pulse varies continuously in one-to-one
correspondence with a sample value. In analog pulse modulation systems, the amplitude, width, or
position of a pulse can vary over a continuous range in accordance with the message amplitude at
the sampling instant, as shown in Figure 6.2. These lead to the following

Three types of pulse modulation:

1. Pulse Amplitude Modulation (PAM)
2. Pulse Width Modulation (PWM)
3. Pulse Position Modulation (PPM)

PAM: In this scheme high frequency carrier (pulse) is varied in accordance with sampled value
of message signal.

PWM: In this width of carrier pulses are varied in accordance with sampled values of message
signal. Example: Speed control of DC Motors.

PPM: In this scheme position of high frequency carrier pulse is changed in accordance with
the sampled values of message signal.

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 Fig. Representation of Various Analog Pulse Modulations

2. Digital Pulse Modulation

In systems utilizing digital pulse modulation, the transmitted samples take on only discrete
values. Two important types of digital pulse modulation are:
1. Delta Modulation (DM)

2. Pulse Code Modulation (PCM)

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 ANALOG PULSE MODULATION

1. Pulse Amplitude Modulation (PAM):

In pulse amplitude modulation, the amplitude of regular interval of periodic pulses or electromagnetic
pulses is varied in proposition to the sample of modulating signal or message signal. This is an
analog type of modulation. In the pulse amplitude modulation, the message signal is sampled at
regular periodic or time intervals and this each sample is made proportional to the magnitude of
the message signal. These sample pulses can be transmitted directly using wired media or we can
use a carrier signal for transmitting through wireless.

Fig. Pulse Amplitude Modulation Signal

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 /*

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Advantages 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. Transmitter
and receiver circuitry is simple and easy to construct.
 PAM can generate other pulse modulation signals and can carry the message or
information at same time.

Disadvantages of Pulse Amplitude Modulation (PAM):

 Bandwidth should be large for transmitting the pulse amplitude modulation signal. Due to
Nyquist criteria also high bandwidth is required.
 The frequency varies according to the modulating signal or 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 techniques. It is almost
equal to amplitude modulation.
 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.

Applications of Pulse Amplitude Modulation (PAM):

 It is mainly used in Ethernet which is type of computer network communication, we know

that we can use Ethernet for connecting two systems and transfer data between the systems.
Pulse amplitude modulation is used for Ethernet communications.
 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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 DIGITAL PULSE MODULATION

Modulation is the process of varying one or more parameters of a carrier signal in accordance
with the instantaneous values of the message signal.

1. PULSE CODE MODULATION(PCM)

The message signal is the signal which is being transmitted for communication and the carrier signal
is a high frequency signal which has no data, but is used for long distance transmission.
There are many modulation techniques, which are classified according to the type of modulation
employed. Of them all, the digital modulation technique used is Pulse Code Modulation
(PCM).
A signal is pulse code modulated to convert its analog information into a binary sequence, i.e., 1s
and 0s. The output of a PCM will resemble a binary sequence. The following figure shows an
example of PCM output with respect to instantaneous values of a given sine wave.

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process is called
as digital. Each one of these digits, though in binary code, represent the approximate amplitude
of the signal sample at that instant.
In Pulse Code Modulation, the message signal is represented by a sequence of coded pulses. This
message signal is achieved by representing the signal in discrete form in both time and
amplitude.

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Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists of Sampling,

Quantizing and Encoding, which are performed in the analog-to-digital converter section. The
low pass filter prior to sampling prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of impaired signals,
decoding, and reconstruction of the quantized pulse train. Following is the block diagram of
PCM which represents the basic elements of both the transmitter and the receiver sections.

 Low Pass Filter

This filter eliminates the high frequency components present in the input analog signal which
is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.
 Sampler
This is the technique which helps to collect the sample data at instantaneous values of message
signal, so as to reconstruct the original signal. The sampling rate must be greater than twice the
highest frequency component W of the message signal, in accordance with the sampling
theorem.
 Quantizer
Quantizing is a process of reducing the excessive bits and confining the data. The
sampled output when given to Quantizer reduces the redundant bits and compresses the

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value.

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  Encoder

Encoder assigns code words to quantized sampled values. This coding techniques uses bits 0 and
1. If number of quantized levels are 16 then each sample is assigned with 4 bit code word.

 Regenerative repeater:

The PCM has an ability to control the distortion and noise caused by the transmission of bits along
the channel. This ability is accomplished by several regenerative repeaters located at sufficient
placing along channel.

Regenerative repeaters have three functions.

1. Equalizing
2. Timing circuits
3. Decision making device

Equalizer shapes the received pulse so as to compensate amplitude and phase distortion caused by the
channel.

Timing circuits provides periodic pulse trains.

 Decision making device compares amplitude of equalized pulse plus noise to the pre-defined
threshold levels to make decisions whether the pulse is present or not.
 If the pulse is present (i.e. decision is yes), clean new pulse is generated and transmitted
through channel to next regenerative pulse. If the pulse is not present (i.e. the decision is no),
it generates clean base line to next regenerative repeater, provided the noise too large caused
bit error by taking the wrong decision

 Decoder

Decoder reboots all the received bits to make more words then it decodes as quantized PAM signals.

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  Reconstruction Filter:
All coded words are passed through low pass filter so that analog signal can be reconstructed from
quantized PAM signal.The cut off frequency of low pass filter is f m Hz which is equal to band width
of message signal.

 Destination
It receives the signal from the reconstructive filter output is analog signal.

Fig.PCM waveform

Bit rate and bandwidth requirements of PCM :

 The bit rate of a PCM signal can be calculated form the number of bits per sample × the
sampling rate. Bit rate =𝑛𝑏×𝑓𝑠 The bandwidth required to transmit this signal depends
on the type of line encoding used.
 A digitized signal will always need more bandwidth than the original analog signal. Price
we pay for robustness and other features of digital transmission.

Important Relations
 Quantization Noise (𝑁𝑞)=Δ2/2
 Signal to Noise ratio
(𝑆𝑄𝑁𝑅)=32.22𝑛 𝑜𝑟 𝑆𝑄𝑁𝑅 𝑖𝑛 𝑑𝐵=1.76+6.02𝑛≅(1.8+6𝑛)𝑑𝐵
 𝐵𝑖𝑡 𝑟𝑎𝑡𝑒=𝑁𝑜.𝑜𝑓 𝑏𝑖𝑡𝑠 𝑝𝑒𝑟 𝑠𝑎𝑚𝑝𝑙𝑒×𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑟𝑎𝑡𝑒=𝑛𝑓𝑠
 Bandwidth for PCM signal
=n.fm Where,
n – No. of bits in PCM code
Fm – signal bandwidth
fs – sampling rate

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SAMPLING, QUANTIZATION AND CODING
1. Sampling
 Definition: Sampling is defined as ―The process of measuring the
instantaneous values of continuous-time signal in a discrete form.‖
 Sample is a piece of data taken from the whole data which is continuous in the time domain.

When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e., High
or Low, the signal has to be discretized in time. This discretization of analog signal is called as
Sampling.
The following figure indicates a continuous-time signal x (t) and a sampled signal xs (t). When x (t)
is multiplied by a periodic impulse train, the sampled signal xs (t) is obtained.

Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be
termed as a sampling period Ts.
Sampling Frequency fs=1/Ts
Where,
Ts is the sampling time
fs is the sampling frequency or the sampling rate

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.
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For an analog signal to be reconstructed from the digitized signal, the sampling rate should be
highly considered. The rate of sampling should be such that the data in the message signal should
neither be lost nor it should get over-lapped. Hence, a rate was fixed for this, called as Nyquist
rate

Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher than W Hertz.
That means, W is the highest frequency. For such a signal, for effective reproduction of the
original signal, sampling rate should be twice the highest frequency.
This means,
fs=2W
Where,
fs is the sampling rate
W is the highest frequency
This rate of sampling is called as Nyquist rate.
A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate.

Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient
sample rate in terms of bandwidth for the class of functions that are band limited.

The sampling theorem states that, ― a signal can be exactly reproduced if it is sampled at the rate
fs which is greater than twice the maximum frequency W.

To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal
whose value is non-zero between some –W and W Hertz.
Such a signal is represented as x(f)=0for|f|>W

For the continuous-time signal x (t), the band-limited signal in frequency domain, can be
represented as shown in the following figure.

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.

We need a sampling frequency, a frequency at which there should be no loss of information,

even after sampling. For this, we have the Nyquist rate that the sampling frequency should be
two times the maximum frequency. It is the critical rate of sampling.

If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered, and if it
is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency domain.

The above figure shows the Fourier transform of a signal xs(t). Here, the information is
reproduced without any loss. There is no mixing up and hence recovery is possible.

The Fourier Transform of the signal

xs(t) is Xs (w)=1Ts∑n=−∞∞X(w−nw0)
Where Ts = Sampling Period and w0=2πTs

Let us see what happens if the sampling rate is equal to twice the highest
frequency (2W) That means,

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Fs =2W

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 Where,
Fs is the sampling frequency
W is the highest frequency

The result will be as shown in the above figure. The information is replaced without any loss.
Hence, this is also a good sampling rate.
Now, let us look at the condition,
Fs <2W
The resultant pattern will look like the following figure

We can observe from the above pattern that the over-lapping of information is done, which
leads to mixing up and loss of information. This unwanted phenomenon of over-lapping is
called as Aliasing

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Aliasing

Aliasing can be referred to as ―the phenomenon of a high-frequency component in the

spectrum of a signal, taking on the identity of a low-frequency component in the spectrum
of its sampled version.‖
The corrective measures taken to reduce the effect of Aliasing are −
 In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the
sampler, to eliminate the high frequency components, which are unwanted.
 The signal which is sampled after filtering, is sampled at a rate slightly higher than the
Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier
design of the reconstruction filter at the receiver.

Scope of Fourier Transform

It is generally observed that, we seek the help of Fourier series and Fourier transforms in
analyzing the signals and also in proving theorems. It is because −

 The Fourier Transform is the extension of Fourier series for non-periodic signals.
 Fourier transform is a powerful mathematical tool which helps to view the signals in
different domains and helps to analyze the signals easily.
 Any signal can be decomposed in terms of sum of sines and cosines using this Fourier
transform. The digitization of analog signals involves the rounding off of the values which are
approximately equal to the analog values. The method of sampling chooses a few points on
the analog signal and then these points are joined to round off the value to a near
stabilized value. Such a process is called as Quantization.

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 Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to create a series of digital values out
of the given analog signal. The following figure represents an analog signal. This signal to get converted
into digital has to undergo sampling and quantizing

The quantizing of an analog signal is done by discretizing the signal with a number of quantization
levels.

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.
The following figure shows how an analog signal gets quantized. The blue line represents analog
signal while the brown one represents the quantized signal.

Both sampling and quantization result in the loss of information. The quality of a Quantizer output
depends upon the number of quantization levels used. The discrete amplitudes of the quantized
output are called as representation levels or reconstruction levels. The spacing between the two
adjacent representation levels is called a quantum or step-size.
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The following figure shows the resultant quantized signal which is the digital form for the given
analog signal.

This is also called as Stair-case waveform, in accordance with its shape.

Types of Quantization
There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.

1. The type of quantization in which the quantization levels are uniformly spaced is termed as a
Uniform Quantization.
2. 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.

1. Mid-Rise type
2. Mid-Tread type.

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 The following figures represent the two types of uniform quantization

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of uniform quantization.

1. The Mid-Rise type is so called because the origin lies in the middle of a raising part
of the stair- case like graph. The quantization levels in this type are even in number.
2. The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.

Both the mid-rise and mid-tread type of uniform quantizer are symmetric about the origin.
Δ= (𝑚𝑎𝑥−𝑚𝑖𝑛)𝐿
𝑛𝑏=𝑙𝑜𝑔2𝐿

Quantization Error

For any system, during its functioning, there is always a difference in the values of its input and output.
The processing of the system results in an error, which is the difference of those values.The difference
between an input value and its quantized value is called a Quantization Error.

A Quantizer is a logarithmic function that performs Quantization (rounding off the value). An
analog-to- digital converter (ADC) works as a quantizer.

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The following figure illustrates an example for a quantization error, indicating the difference
between the original signal and the quantized signal.

Quantization Noise

It is a type of quantization error, which usually occurs in analog audio signal, while quantizing it
to digital. For example, in music, the signals keep changing continuously, where a regularity is
not found in errors. Such errors create a wideband noise called as Quantization Noise.

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  COMPANDING IN PCM SYSTEMS

The word Companding is a combination of Compressing and Expanding, which means that it does both.
This is a non-linear technique used in PCM which compresses the data at the transmitter and expands the
same data at the receiver. The effects of noise and crosstalk are reduced by using this technique

Fig. Companding
Companding means it amplifies the low level signals as well as attenuate high level at the
transmitter side. At the receiver side reverse operation done. It attenuates the low level signals and
amplifies the high level signals you get the original signal. Non-uniform quantization cannot be
applied directly by using companding technique.

Fig Companding curves for PCM

Companding is used to maintain constant Signal to Noise Ratio throughout dynamic quantization
ratio

Fig. Non Uniform Quantization

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 SAMPLING PROCESS

Due to the increased use of computers in all engineering applications, including signal
processing, it is important to spend some more time examining issues of sampling. In this
chapter we will look at sampling both in the time domain and the frequency domain.

We have already encountered the sampling theorem and, arguing purely from a trigonometric-
identity point of view, have established the Nyquist sampling criterion for sinusoidal signals.
However, we have not fully addressed the sampling of more general signals, nor provided a
general proof. Nor have we indicated how to reconstruct a signal from its samples. With the tools
of Fourier transforms and Fourier series available to us we are now ready to finish the job that
was started months ago.

To begin with, suppose we have a signal x(t) which we wish to sample. Let us suppose further
that the signal is bandlimited to B Hz. This means that its Fourier transform is nonzero for −2πB
< ω < 2πB. Plot spectrum.

We will model the sampling process as multiplication of x(t) by the “picket fence” function

δT(t) = Xδ(t − nT).

We encountered this periodic function when we studied Fourier series. Recall that by its Fourier
series representation we can write

where . The frequency fs = ωs/(2π) = 1/T is the sampling frequency in samples/sec.

Suppose that the sampling frequency is chosen so that fs > 2B, or equivalently, ωs > 4πB.

THE SAMPLING THEOREM

If x(t) is bandlimited to B Hz then it can be recovered from signals taken at a sampling rate fs >
2B. The recovery formula is

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where

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 .

Show what the formula means: we are interpolating in time between samples using the
sinc function.

We will prove this theorem. Because we are actually lacking a few theoretical tools, it will
take a bit of work. What makes this interesting is we will end up using in a very essential way
most of the transform ideas we have talked about.

1. The first step is to notice that the spectrum of the sampled signal,

is periodic and hence has a Fourier series. The period of the function in frequency is ωs, and the
fundamental frequency is

By the fourier.Series. we can write

where the cn are the F.S. coefficients

But the integral is just the inverse F.T., evaluated at t = −nT:

so

2. Let g(t) = sinc(πfst). Then

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 .

3. Let

We will show that y(t) = x(t) by showing that Y (ω) = X(ω). We can compute the F.T. of
y(t) using linearity and the shifting property:

Observe that the summation on the right is the same as the F.S. we derived in step 1:

Now substituting in the spectrum of the sampled signal (derived above)

since x(t) is bandlimited to −πfs < ω < πfs or −fs/2 < f < fs/2.

Fig. Sampling
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Notice that the reconstruction filter is based upon a sinc function,
whose transform is a rect function: we are really just doing the
filtering implied by our initial intuition.In practice, of course, we want
to sample at a frequency higher than just twice the bandwidth to allow
room for filter rolloff

 QUANTIZATION

Quantization approximates the sampled value to nearest discrete

value from the set of finite discrete levels.
Quantization error

Fig. Quantization effect in PCM

In mid treed quantization the input values lies between ± Δ/2, ±

3Δ/2, ± 5Δ/2, . . . in that output values are quantized values at ± Δ,±
2Δ,±3Δ,……. Suppose the input (i.e. sampled value) lies between ±
Δ/2 which is approximated as zero. If the input values lies between
Δ/2 to 3Δ/2 this quantizer approximates sampled value as Δ. Here
the origin of treed of stair case lies at midpoint so the name is called
mid treed quantizer. In that maximum quantization error is Δ/2 and
minimum quantization error is -Δ/2.
 Page 1 of 6

Analog Pulse
Modulation
After the continuous wave modulation, the next division is Pulse modulation. Pulse
modulation is further divided into analog and digital modulation. The analog modulation
techniques are mainly classified into Pulse Amplitude Modulation, Pulse Duration
Modulation/Pulse Width Modulation, and Pulse Position Modulation.

Pulse Amplitude Modulation

Pulse Amplitude Modulation (PAM) is an analog modulating scheme in which the
amplitude of the pulse carrier varies proportional to the instantaneous amplitude of the
message signal.

The pulse amplitude modulated signal, will follow the amplitude of the original signal, as
the signal traces out the path of the whole wave. In natural PAM, a signal sampled at the
Nyquist rate is reconstructed, by passing it through an efficient Low Pass Frequency
(LPF) with exact cutoff frequency

The following figures explain the Pulse Amplitude Modulation.

 Page 2 of 6

Pulse Width Modulation

Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse
Time Modulation (PTM) is an analog modulating scheme in which the duration or
width or time of the pulse carrier varies proportional to the instantaneous amplitude of
the message signal.

The width of the pulse varies in this method, but the amplitude of the signal remains
constant. Amplitude limiters are used to make the amplitude of the signal constant.
These circuits clip off the amplitude, to a desired level and hence the noise is limited.

The following figures explain the types of Pulse Width Modulations.

 Page 3 of 6

There are three variations of PWM. They are −

The leading edge of the pulse being constant, the trailing edge varies according
to the message signal.

The trailing edge of the pulse being constant, the leading edge varies according
to the message signal.

The center of the pulse being constant, the leading edge and the trailing edge
varies according to the message signal.

These three types are shown in the above given figure, with timing slots.

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Pulse Position Modulation

Pulse Position Modulation (PPM) is an analog modulating scheme in which the
amplitude and width of the pulses are kept constant, while the position of each pulse,
with reference to the position of a reference pulse varies according to the instantaneous
sampled value of the message signal.

The transmitter has to send synchronizing pulses (or simply sync pulses) to keep the
transmitter and receiver in synchronism. These sync pulses help maintain the position of
the pulses. The following figures explain the Pulse Position Modulation.
Page 4 of 6
 Page 5 of 6

Pulse position modulation is done in accordance with the pulse width modulated signal.
Each trailing of the pulse width modulated signal becomes the starting point for pulses in
PPM signal. Hence, the position of these pulses is proportional to the width of the PWM
pulses.

Advantage
As the amplitude and width are constant, the power handled is also constant.

Disadvantage
The synchronization between transmitter and receiver is a must.

Comparison between PAM, PWM, and PPM

The comparison between the above modulation processes is presented in a single table.

PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends on Bandwidth depends on the Bandwidth depends on the

the width of the pulse rise time of the pulse rise time of the pulse

Instantaneous transmitter Instantaneous transmitter Instantaneous transmitter

power varies with the power varies with the power remains constant
amplitude of the pulses with the width of the pulses
 Page 1 of 4

Digital Modulation
Techniques
Digital Modulation provides more information capacity, high data security, quicker system
availability with great quality communication. Hence, digital modulation techniques have
a greater demand, for their capacity to convey larger amounts of data than analog ones.

There are many types of digital modulation techniques and we can even use a
combination of these techniques as well. In this chapter, we will be discussing the most
prominent digital modulation techniques.

Amplitude Shift Keying

The amplitude of the resultant output depends upon the input data whether it should be
a zero level or a variation of positive and negative, depending upon the carrier
frequency.

Amplitude Shift Keying (ASK) is a type of Amplitude Modulation which represents the
binary data in the form of variations in the amplitude of a signal.

Following is the diagram for ASK modulated waveform along with its input.

Any modulated signal has a high frequency carrier. The binary signal when ASK is
modulated, gives a zero value for LOW input and gives the carrier output for HIGH input.

Frequency Shift Keying

The frequency of the output signal will be either high or low, depending upon the input
data applied.
 Page 2 of 4
Frequency Shift Keying (FSK) is the digital modulation technique in which the
frequency of the carrier signal varies according to the discrete digital changes. FSK is a
scheme of frequency modulation.

Following is the diagram for FSK modulated waveform along with its input.

The output of a FSK modulated wave is high in frequency for a binary HIGH input and is
low in frequency for a binary LOW input. The binary 1s and 0s are called Mark and
Space frequencies.

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Phase Shift Keying

The phase of the output signal gets shifted depending upon the input. These are mainly
of two types, namely BPSK and QPSK, according to the number of phase shifts. The
other one is DPSK which changes the phase according to the previous value.

Phase Shift Keying (PSK) is the digital modulation technique in which the phase of the
carrier signal is changed by varying the sine and cosine inputs at a particular time. PSK
technique is widely used for wireless LANs, bio-metric, contactless operations, along with
RFID and Bluetooth communications.
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Modulation
Techniques
There are few modulation techniques which are followed to construct a PCM signal. These
techniques like sampling, quantization, and companding help to create an effective
PCM signal, which can exactly reproduce the original signal.

Quantization
The digitization of analog signals involves the rounding off of the values which are
approximately equal to the analog values. The method of sampling chooses few points
on the analog signal and then these points are joined to round off the value to a near
stabilized value. Such a process is called as Quantization.

The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels. 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.

The following figure shows how an analog signal gets quantized. The blue line represents
analog signal while the red one represents the quantized signal.

Both sampling and quantization results in the loss of information. The quality of a
Quantizer output depends upon the number of quantization levels used. The discrete
amplitudes of the quantized output are called as representation levels or
reconstruction levels. The spacing between two adjacent representation levels is
called a quantum or step-size.
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Companding in
PCM
The word Companding is a combination of Compressing and Expanding, which means
that it does both. This is a non-linear technique used in PCM which compresses the data at
the transmitter and expands the same data at the receiver. The effects of noise and
crosstalk are reduced by using this technique.

There are two types of Companding techniques.

A-law Companding Technique

Uniform quantization is achieved at A = 1, where the characteristic curve is linear

and there is no compression.

A-law has mid-rise at the origin. Hence, it contains a non-zero value.

A-law companding is used for PCM telephone systems.

A-law is used in many parts of the world.

µ-law Companding Technique

Uniform quantization is achieved at µ = 0, where the characteristic curve is linear

and there is no compression.
µ-law has mid-tread at the origin. Hence, it contains a zero value.

µ-law companding is used for speech and music signals.

µ-law is used in North America and Japan.

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Differential PCM
The samples that are highly correlated, when encoded by PCM technique, leave
redundant information behind. To process this redundant information and to have a
better output, it is a wise decision to take predicted sampled values, assumed from its
previous outputs and summarize them with the quantized values.

Such a process is named as Differential PCM technique.

 Sampling Process
Definition:
An analog signal is converted into a corresponding
sequence of samples that are usually spaced
uniformly in time.

Figure: Illustration of sampling process. (a) Analog waveform (b)

Instantaneously sampled representation of the analog Signal
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 (1)

(2)

• Where G(f) is the Fourier Transform of the original

signal g(t), and is the sampling rate.
• The above equation states that the process of
uniformly sampling a continuous time signal of finite
energy results in a periodic spectrum with a period
equal to sampling rate.
(3)

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 Sampling Process Contd.,
• Suppose, that the signal g(t) is strictly band-limited, with no
frequency components higher than W Hertz. That is, the
Fourier transform of g(t) has the property that G(f) is zero for
, as illustrated in below figure. Also we choose, Ts =
1/2W, then the corresponding spectrum shown in figure b.

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

(b) Spectrum of the sampled version of g(t) for a sampling period Ts=1/2W
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 Sampling Process Contd.,
(4)

The Fourier Transform of may also be expressed as

(5)

Hence, under the following two conditions,

1.
2.
We find from equation (5) that
(6)

substitute equation (4) into the above equation, we

may also write
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 Sampling Process Contd.,
(7)

• Therefore, if the sampled values of g(n/2W) of a signal g(t)

are specified for all n, then the Fourier transform G(f) of the
signal is uniquely determined using eq (7). Because the
sequence g(n/2W) has all the information contained in g(t).
• The expression for reconstructing the original signal g(t) from
the sequence of sample values {g(n/2W)} is
(8)

• playing the role of an interpolation function. Each

sample is multiplied by a delayed version of interpolation
function, and all the resulting waveforms are added to obtain
g(t)
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 Sampling Theorem
• A band-limited signal of finite energy, which has no frequency
components higher than W Hertz, is completely described by
specifying the values of the signal at instants of time seperated
by 1/2W seconds.
OR
• A band-limited signal of finite energy, which has no frequency
components higher than W Hertz, may be completely
recovered from a knowledge of its samples taken at the rate of
2W samples per second.
• The sampling rate of 2W samples per second, for a signal
bandwidth of W Hertz, is called Nyquist rate; its reciprocal
1/2W is called Nyquist interval.

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 Aliasing Effect
• If , an information bearing signal is not strictly band-limited,
some aliasing is produced by the sampling process.
• Aliasing refers to the phenomenon of a high frequency
component in the spectrum of the signal seemingly taking on
the identity of a lower frequency in the spectrum of its
sampled version, as illustrated in below figure.

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

exhibiting the aliasing phenomenon
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 Corrective Measures for Aliasing
1. Prior to sampling, a low-pass anti-aliasing filter is used to
attenuate those high frequency components of the signal that
are not essential to the information being conveyed by the
signal.
2. The filtered signal is sampled at a rate slightly higher than the
Nyquist rate. Also, it has the beneficial effect of easing the
design of the reconstruction filter used to recover the original
signal from its sampled version.

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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) Magnitude
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response of reconstruction filter
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 Pulse Analog Modulation
• In analog modulation systems, some parameter of a sinusoidal
carrier is varied according to the instantaneous value of the
modulating signal.
• In Pulse modulation methods, the carrier is no longer a
continuous signal but consists of a pulse train. Some parameter
of which is varied according to the instantaneous value of the
modulating signal.

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 Types of Pulse Modulation

Pulse Modulation

Pulse Analog Pulse Digital

Modulation Modulation

PAM PCM

PWM DPCM

PPM DM & ADM

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 Pulse Amplitude Modulation
• The amplitude of the pulses of the carrier pulse
train is varied in accordance with the modulating
signal, that is amplitude of the pulses depends on
the value of m(t) during the time of pulse.

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 Demodulation of PAM
PAM signal Reconstruction Message signal
s(t) Equalizer m(t)
Filter

Figure: System for recovering message signal m(t) from PAM signal s(t)

• The distortion caused by the use of PAM 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 fig.

• The equalizer has the effect of decreasing the in-band loss of

the reconstruction filter as the frequency increases in such a
manner as to compensate for the aperture effect.

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 Demodulation of PAM
• Ideally, the magnitude response of the equalizer is given by

• The amount of equalization needed in practice is usually

small.

Advantages of PAM :
• It is the simple and simple process for modulation and
demodulation
• Transmitter and receiver circuits are simple and easy to
construct.

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 Drawbacks of PAM signal
• The bandwidth required for the transmission of a PAM
signal is very large in comparison to the maximum
frequency present in the modulating signal.
• Since the amplitude of the PAM pulses varies in
accordance with the modulating signal therefore the
interference of noise is maximum in a PAM signal. This
noise cannot be removed easily.
• Since the amplitude of the PAM pulses varies, therefore,
this also varies the peak power required by the transmitter
with modulating signal.

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 Pulse Time Modulation (PTM)
• 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.
i. Pulse Width Modulation
ii. Pulse Position Modulation
Pulse Width Modulation
• PWM is also called Pulse Duration Modulation (PDM), Pulse
Length Modulation (PLM) and
Definition:
In PWM, Width of the pulses of the carrier pulse train is varied in
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 accordance with the modulating signal.

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6/5/2020 Prof.Ch.Srinivasa Rao - JNTUK - UCEV 67
 PPM
Advantages of PPM:
• Like PWM, in PPM, amplitude is held constant thus less noise
interference.
• Signal and noise separation is very easy
• Because of constant pulse widths and amplitudes, transmission
power for each pulse is same
Disadvantages of PWM:
• Synchronization between transmitter and receiver is required.
• Large bandwidth is required for the PPM as compared to PAM

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Comparison between PAM, PWM & PPM
S.
PAM PWM/PDM PPM
No
Amplitude of the pulse is The relative position of the
Width of the pulse is
proportional to the pulse is proportional to
1 amplitude of modulating proportional to amplitude the amplitude of
of modulating signal.
signal modulating signal.

The bandwidth of the Bandwidth of

Bandwidth of transmission
transmission channel transmission channel
2 channel depends on rise
depends on width of the depends on rise time of
time of the pulse.
pulse the pulse.
The instantaneous The instantaneous
The instantaneous power
power of the power of the
of the transmitter varies
3 transmitter varies transmitter
with width of pulses
with amplitude of remains constant
pulses. with width of
pulses.
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 Comparison between PAM, PWM & PPM
Contd.,

S.
PAM PWM/PDM PPM
No

Noise interference is Noise interference is Noise interference is

4
high minimum minimum

5 System is complex Simple is implement Simple is implement

6
Similar to Amplitude Similar to frequency Simple to Phase
modulation modulation modulation

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 SYNCHRONIZATION
• PAM
• PWM
• PPM

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 Time Division Multiplexing (TDM)
• An important feature of the sampling process is a conservation
of time. That is, the transmission of the message samples
engages the communication channel for only a fraction of the
sampling interval on a periodic basis, and in this way some of
the time interval between adjacent samples is cleared for use
by other independent message sources on a time-shared basis.
• Time Division Multiplex (TDM) is a system, which enables
the joint utilization of common communication channel by a
plurality of independent message sources without mutual
interference among them.
• The concept of TDM is illustrated by the block diagram shown
below.

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 TDM Contd.,

Figure: Block diagram of TDM system

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 TDM Contd.,
• Each input message signal is first restricted in bandwidth by a
low-pass anti-aliasing filter to remove the frequencies that are
nonessential to an adequate signal representation.
• The LPF outputs are applied to commutator. The function of
the commutator is two fold
1. To take a narrow sample of each of the N input messages at a
rate fs that is slightly higher than 2W, where W is the cutoff
frequency of the anti-aliasing filter.
2. To sequentially interleave these N samples inside the
sampling interval Ts.
• The multiplexed signal is applied to a pulse modulator, the
purpose of which is to transform the multiplexed signal into a
form suitable for transmission over the common channel.

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 TDM Contd.,
• At the receiving end of the system, the received signal is
applied to the pulse demodulator, which performs the reverse
operation of the pulse modulator.
• The narrow samples produced at the pulse demodulator output
are distributed to the appropriate low-pass reconstruction
filters by means of a decommutator, which operates in
synchronism with the commutator in the transmitter.
Synchronization depends on the method of pulse modulation
used to transmit the multiplexed sequence of samples.

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 Page 1 of 5

Pulse Modulation
So far, we have discussed about continuous-wave modulation. Now it’s time for discrete
signals. The Pulse modulation techniques, deals with discrete signals. Let us see how
to convert a continuous signal into a discrete one. The process called Sampling helps us
with this.

Sampling
The process of converting continuous time signals into equivalent discrete time signals,
can be termed as Sampling. A certain instant of data is continually sampled in the
sampling process.

The following figure indicates a continuous-time signal x(t) and a sampled signal xs(t).
When x(t) is multiplied by a periodic impulse train, the sampled signal xs(t) is obtained.

A sampling signal is a periodic train of pulses, having unit amplitude, sampled at

equal intervals of time Ts, which is called as the Sampling time. This data is
transmitted at the time instants Ts and the carrier signal is transmitted at the remaining
time.

Sampling Rate
 Page 2 of 5

To discretize the signals, the gap between the samples should be fixed. That gap can be
termed as the sampling period Ts.

1
Sampling Frequency = = fs
Ts
Where,

Ts = the sampling time

fs = the sampling frequency or sampling rate

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Sampling Theorem
While considering the sampling rate, an important point regarding how much the rate
has to be, should be considered. The rate of sampling should be such that the data in
the message signal should neither be lost nor it should get over-lapped.

The sampling theorem states that, “a signal can be exactly reproduced if it is sampled
at the rate fs which is greater than or equal to twice the maximum frequency W.”

To put it in simpler words, for the effective reproduction of the original signal, the
sampling rate should be twice the highest frequency.

Which means,

fs ≥ 2 W
Where,

fs = the sampling frequency

W is the highest frequency

This rate of sampling is called as Nyquist rate.

The sampling theorem, which is also called as Nyquist theorem, delivers the theory of
sufficient sample rate in terms of bandwidth for the class of functions that are
bandlimited.

For the continuous-time signal x(t), the band-limited signal in frequency domain, can be
represented as shown in the following figure.
 Page 3 of 5

If the signal is sampled above the Nyquist rate, the original signal can be recovered. The
following figure explains a signal, if sampled at a higher rate than 2w in the frequency
domain.

If the same signal is sampled at a rate less than 2w, then the sampled signal would look
like the following figure.
 Page 4 of 5

We can observe from the above pattern that the over-lapping of information is done,
which leads to mixing up and loss of information. This unwanted phenomenon of over-
lapping is called as Aliasing.

Aliasing can be referred to as “the phenomenon of a high-frequency component in the

spectrum of a signal, taking on the identity of a lower-frequency component in the
spectrum of its sampled version.”

Hence, the sampling of the signal is chosen to be at the Nyquist rate, as was stated in
the sampling theorem. If the sampling rate is equal to twice the highest frequency (2W).

That means,

fs = 2 W
Where,

fs = the sampling frequency

W is the highest frequency

 Page 5 of 5

The result will be as shown in the above figure. The information is replaced without any
loss. Hence, this is a good sampling rate.