UNIT-3

Pulse Modulation

Sampling: 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.
SamplingFrequency=1/Ts = fs

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.

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, 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.

A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate.
Low-Pass Filter:

A low-pass filter (LPF) is a circuit that only passes signals below its cutoff frequency while
attenuating all signals above it. It is the complement of a high-pass filter, which only passes
signals above its cutoff frequency and attenuates all signals below it.

Low-pass filters have applications such as anti-aliasing, reconstruction, and speech processing,
and can be used in audio amplifiers, equalizers, and speakers.

There are many different low-pass filter circuits, which are characterized by their order and
amplitude characteristic or the type of polynomial that describes it (Butterworth, Chebyshev,
Elliptic, or Bessel).

Figure shows the basic filter structures in the frequency domain of LPF. These ideal filters are
identical for both analogue and digital filters

Bandpass Filter:
The bandpass filter is defined as a device that allows the frequencies which are within the
required frequency range and reject the remaining frequencies which are not in that specific
range.

There are two cuts off frequencies in this bandpass filter, one is from the high pass filter and its
high cut off frequency. This will be the highest frequency limit of that band. There is another
cut-off frequency from the low pass filter and its second cut off frequency known as the lower
cut off frequency which decides the low-frequency range of that band.

This bandpass filter only allows the particular band and attenuates all the remaining signals
above or below that cut off frequencies.
Figure shows the basic filter structures in the frequency domain of BPF. These ideal filters are
identical for both analogue and digital filters

The difference between the lowest cut-off frequency and the highest cut-off frequency is known
as bandwidth.

Bw= fH – fL

Sampling Theorem:
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:

Sampled signal y(t)=x(t). δ(t)......(1)Sampled signal......(1)

The trigonometric Fourier series representation of δ (t) is given by

To reconstruct x(t), you must recover input signal spectrum X(ω) from sampled signal spectrum
Y(ω), which is possible when there is no overlapping between the cycles of Y(ω).

Possibility of sampled frequency spectrum with different conditions is given by the following
diagrams:

Aliasing Effect:

The overlapped region in case of under sampling represents aliasing effect, which can be
removed by
  considering fs >2fm
 By using anti-aliasing filters.

types of sampling techniques:

There are three types of sampling techniques:

 Impulse sampling.
 Natural sampling.
 Flat Top sampling.

Impulse sampling:

Natural sampling:
Major differences between Natural Sampling and Flat-Top Sampling

Natural Sampling Flat-Top Sampling

Natural sampling is a method where Flat-top sampling is a method where samples are
samples are taken at the natural frequency taken at evenly spaced intervals with a specific
of the signal duration.

Natural sampling is used when the signal is Flat-top sampling is used when the signal is
not known beforehand. known beforehand.
 Natural Sampling Flat-Top Sampling

Natural sampling is used to analyze the Flat-top sampling is used to analyze a specific
frequency of a signal. frequency range.

Natural sampling is better suited for non- Flat-top sampling is better suited for periodic
periodic signals. signals.

Natural sampling is sensitive to frequency Flat-top sampling is not sensitive to frequency

variations. variations.

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.

As we can see it consist of a low pass filter, a modulator along with a train generator and a pulse
reshaping circuit.

Here the modulating signal is given to the low pass filter in order to band limit the message
signal.
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. This basically shapes the pulses so that it can be easily detected at the
receiver.

The following figures explain the Pulse Amplitude Modulation.

Though the PAM signal is passed through an LPF, it cannot recover the signal without distortion.
Hence to avoid this noise, flat-top sampling is done as shown in the following figure.

Flat-top sampling is the process in which sampled signal can be represented in pulses for which
the amplitude of the signal cannot be changed with respect to the analog signal, to be sampled.
The tops of amplitude remain flat. This process simplifies the circuit design.

Transmission bandwidth of PAM

During pulse amplitude modulation technique, generally, τ (tau) which is pulse duration of the
modulated signal is assumed to be very small as compared to the time period between two
samples denoted by Ts

Consider the maximum frequency of the modulating signal m(t) to be f m, thus in correspondence
to the sampling theorem

SamplingFrequency=1/Ts = fs
Where fs is the sampling frequency,

Or we can write,

But as we already discussed,

Hence,

The maximum frequency of the pulse modulated signal is achieved when the ON and OFF time
of the modulated pulse is same,

Thus, the transmission bandwidth in case of PAM signal is somewhat equal to the maximum
frequency component.

So, BW ≥ fmax

However, we know,
 Therefore

Thus,

Or we can say,

Thus, the bandwidth for the transmission of the PAM signal is greater than the maximum
frequency component of the message signal.

Advantages of Pulse Amplitude Modulation

 PAM is the simplest form of pulse modulation.

 Its implementation is quite easy.

Disadvantages of Pulse Amplitude Modulation

 The transmission bandwidth required is very large.

 Due to the variation in amplitude, the power required by the generating unit also
varies.

 Less immune to noise due to amplitude variation.

Applications of Pulse Amplitude Modulation

It is used in LED lighting, in microcontrollers in order to produce control signals and in the
Ethernet communication system.
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.

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.

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.

In the above block diagram, a PAM signal is generated from the modulator once, and further, it
is processed at the comparator to produce a PWM signal. After that, the output of the comparator
is given to a monostable multivibrator which is negative edge triggered. Thus, with the trailing
edge of the PWM signal, the output of the monostable goes high.

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.
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 the Bandwidth depends on the rise Bandwidth depends on the rise
width of the pulse time of the pulse time of the pulse

Instantaneous transmitter Instantaneous transmitter power Instantaneous transmitter power

power varies with the varies with the amplitude and remains constant with the width
amplitude of the pulses width of the pulses of the pulses

System complexity is high System complexity is low System complexity is low

Noise interference is high Noise interference is low Noise interference is low

It is similar to amplitude It is similar to frequency

It is similar to phase modulation
modulation modulation

Time Division Multiplexing (TDM):

The Time Division Multiplexing (TDM) is a digital procedure. Here, each sender is given the
entire possession of the whole bandwidth of the channel for a fixed duration of time. After this,
the control is moved to the next sender, and the process continues on a round-robin basis.
An example of TDM is the television broadcast. In a television serial, generally, a 10 minutes'
serial is followed by a 5 minutes' advertisement. The time in which the serial is being
broadcasted, the total frequency is dedicated to the serial.

Variants:

The following variants on TDM are as follows −

 ATDM (Asynchronous Time Division Multiplexing): Multiplexing in which the data is

transmitted asynchronously.
 STDM (Statistical Time Division Multiplexing): A multiplexing method that polls
nodes and immediately skips any nodes with nothing to send.
 STM (Synchronous Transfer Node): Designer for use in BISDN (broadband ISDN)
and also supported in the SONET (Asynchronous Optical Network) architecture.

TDM
A medium can carry only a single signal at any second in time. To transmit multiple
signals to transmit a medium, the medium has to be separated by providing every signal a
segment of the whole bandwidth. This can be possible by using a multiplexing
technique. Multiplexing is a technique is used to combine various signals into a single
signal using a shared medium.

Time-division multiplexing or TDM definition is; a multiplexing technique that is used to

transmit two or above streaming digital signals above a common channel. In this type of
multiplexing technique, incoming signals are separated into equivalent fixed-length time slots.
Once multiplexing is done, these signals are sent over a shared medium & after de-multiplexing,
they are reassembled into their original format.
 Time Division
Multiplexing

Block Diagram of Time Division Multiplexing

The time division multiplexing block diagram is shown below which uses both the sections of
the transmitter and receiver. For data transmission, the multiplexing technique which efficiently
utilizes the whole channel is sometimes called PAM/TDM because; a TDM system utilizes a
PAM. So in this modulation technique, every pulse holds some short time period by allowing
maximal usage of the channel.

It is of the following types:

1. Synchronous TDM:
The time slots are pre-assigned and fixed. This slot is even given if the source is not ready with
data at this time. In this case, the slot is transmitted empty. It is used for multiplexing digitized
voice streams.

2. Asynchronous (or statistical) TDM:

The slots are allocated dynamically depending on the speed of the source or their ready state. It
dynamically allocates the time slots according to different input channels’ needs, thus saving
the channel capacity.

Advantages of Time Division Multiplexing (TDM):

High Capacity: TDM can support a large number of signals over a single communication
channel, making it ideal for applications where many signals need to be transmitted.
Simple Implementation: TDM is a relatively simple technique that is easy to implement,
making it a cost-effective solution for many applications.
Precise Time Synchronization: TDM requires precise time synchronization between the
transmitting and receiving devices, which can help ensure accurate transmission of signals.
Disadvantages of Time Division Multiplexing (TDM):

Inefficient Use of Bandwidth: TDM may not make optimal use of available bandwidth, as
time slots may be left unused if there are no signals to transmit during a particular time slot.
High Implementation Cost: TDM requires sophisticated hardware or software to ensure
precise time synchronization between the transmitting and receiving devices, making it more
expensive to implement than FDM.
Vulnerable to Timing Jitter: TDM can be vulnerable to timing jitter, which can occur when
the timing of the transmitting and receiving devices drifts out of sync, leading to errors in the
transmission of signals.

Digital Representation of Analog Signals.

PCM stands for Pulse Code Modulation and DM stands for Delta Modulation. PCM
is basically used for convert analog to digital signals

Or
Basic Elements of PCM

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

Quantizing and binary code , which are performed in the analog-to-digital converter section.
The low pass filter prior to sampling prevents 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 value.

Binary code

The digitization of analog signal is done by the encoder. It designates each quantized level by a
binary code. The sampling done here is the sample-and-hold process. These three sections (LPF,
Sampler, and Quantizer) will act as an analog to digital converter. Encoding minimizes the
bandwidth used.
Advantages of Pulse Code Modulation (PCM)
 High Noise Immunity: PCM signals are also less likely to be distorted by noises or
interferences as compared to analog signals which is why PCM is normally used in long
distance transmissions.
 Efficient Digital Signal Processing: PCM has been found to be compatible with many
current digital systems for processing, storing and transmission hence can interface easily
with these systems.
 Data Compression: Nevertheless, there are some compression techniques that can be
applied for PCM signals so as to decrease the data width that shall be transmitted without
significant loss in quality.
 Error Detection and Correction: As PCM is a digital signal, therefore the technique of
error detection and correction can be implemented which makes the data transmission is
more reliable.
 High Quality: PCM supplies significantly superior quality sound and video signal
transmission and exhibits a wide range of applications in CDs, telephone line
communication, digital audio systems, etc.
Disadvantages of Pulse Code Modulation (PCM)
 High Bandwidth Requirement: PCM uses a wide bandwidth in its transmission since the
signal is usually sampled in this format so as to provide high quality.
 Complexity: PCM systems are relatively more complicated than
analog modulation schemes since encodings and decoding of the various signals employed
in the system under development are elaborate.
 Quantization Error: Another factor which PCM brings in is the existence of quantization
error through the rounding of the signal values when converting from analog to digital.
 High Power Consumption: Because of the high processing demands, users have typically
reported that PCM systems are power-hungrier and hence not very suitable for low power
conditions.