EC6651 COMMUNICATION ENGINEERING

UNIT 2

Dr Gnanasekaran Thangavel

Professor and Head

Electronics and Instrumentation Engineering

R M K Engineering College

1

UNIT II DIGITAL COMMUNICATION

Pulse modulations – concepts of sampling and sampling

theorems, PAM, PWM, PPM, PTM, quantization and

coding : DCM, DM, slope overload error. ADM, DPCM,

OOK systems – ASK, FSK, PSK, BSK, QPSK, QAM, MSK,

GMSK, applications of Data communication.

2Dr Gnanasekaran Thangavel12/12/2017

YouTube Video Presentation

1. https://www.youtube.com/watch?v=_JMV4ywAJug

2. https://www.youtube.com/watch?v=QEubAxBfqKU

DIGITAL COMMUNICATION

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A digital signal is superior to an analog signal because it is more robust to noise and can easily be

recovered, corrected and amplified. For this reason, the tendency today is to change an analog signal to

digital data.

Digital signals carry more information per second than analogue signals. This is the same whether

optical fibers, cables or radio waves are used.

Digital signals maintain their quality over long distances better than analogue signals.

Good processing techniques are available for digital signals such as source coding (data compression),

channel coding (error detection and correction), equalization etc.

Easy to mix signals and data using digital techniques.

Privacy is preserved by using data encryption. Using data encryption. only permitted receivers can be

allowed to detect the transmitted data. This is very useful in military applications.

High speed computers and powerful software design tools are available. They make the development of

digital communication systems flexible.

Internet is spread almost in every cities and towns. The compatibility of digital communication systems

with Internet has opened new area of applications

Limitations

• Generally, more bandwidth is required than that for analog systems.

• Synchronization is required.

• High power consumption (Due to various stages of conversion).

• Complex circuit, more sophisticated device making is also drawbacks of

digital system.

• Introduce sampling error

• As square wave is more affected by noise, That’s why while

communicating through channel we send sin waves but while operating

on device we use squire pulses.

4

Pulse Code Modulation (PCM)

PCM consists of three steps to digitize an analog signal:

• Sampling

• Quantization

• Binary encoding or Coding

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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 shows a continuous-

time signal x(t) and the corresponding

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

sampling time. This data is transmitted at

the time instants Ts and the carrier signal is

transmitted at the remaining time.

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Sampling Rate

• To discretize the signals, the gap between the samples should be fixed. That

gap can be termed as the sampling period Ts. Reciprocal of the sampling period

is known as sampling frequency or sampling rate fs.

• Mathematically, we can write it as

• Where,

• Fs is the sampling frequency or the sampling rate

• Ts is the sampling period

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Sampling Theorem

• The sampling rate 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 of the given signal W.”

• Mathematically, we can write it as

• fs is the sampling rate

• W is the highest frequency of the given signal

• If the sampling rate is equal to twice the maximum frequency of the given signal

W, then it is called as Nyquist rate.

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• 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.

• For continuous-time signal x(t), which is band-limited in the frequency domain

is represented as shown in the following figure.

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• If the signal is sampled above Nyquist rate, then the original signal can

be recovered. The following figure explains a signal, if sampled at a

higher rate than 2w in the frequency domain.

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• If the same signal is sampled at a rate less than 2w, then the sampled

signal would look like the following figure.

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• We can observe from the above pattern that there is over-lapping of information,

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 low-frequency component in the spectrum of its sampled version.”

• Hence, the sampling rate of the signal is chosen to be as Nyquist rate. If the sampling rate is equal to twice

the highest frequency of the given signal W, then the sampled signal would look like the following figure.

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In this case, the signal can be recovered without any loss. Hence, this is a good sampling rate.

• There are 3 sampling methods:

Ideal - an impulse at each sampling instant

Natural - a pulse of short width with varying amplitude

Flattop - sample and hold, like natural but with single amplitude value

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Example

• For an intuitive example of the Nyquist

theorem, let us sample a simple sine wave at

three sampling rates: fs = 4f (2 times the

Nyquist rate), fs = 2f (Nyquist rate), and fs = f

(one-half the Nyquist rate). Figure 4.24 shows

the sampling and the subsequent recovery of

the signal.

• It can be seen that sampling at the Nyquist

rate can create a good approximation of the

original sine wave (part a). Oversampling in

part b can also create the same

approximation, but it is redundant and

unnecessary. Sampling below the Nyquist rate

(part c) does not produce a signal that looks

like the original sine wave.

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4.15

According to the Nyquist theorem, the sampling rate

must be

at least 2 times the highest frequency contained in the

signal.

Note

• Consider the revolution of a hand of a clock.

The second hand of a clock has a period of 60 s.

According to the Nyquist theorem, we need to

sample the hand every 30 s (Ts = T or fs = 2f ).

In Figure 4.25a, the sample points, in order, are

12, 6, 12, 6, 12, and 6. The receiver of the

samples cannot tell if the clock is moving

forward or backward. In part b, we sample at

double the Nyquist rate (every 15 s). The

sample points are 12, 3, 6, 9, and 12. The clock

is moving forward. In part c, we sample below

the Nyquist rate (Ts = T or fs = f ). The sample

points are 12, 9, 6, 3, and 12. Although the

clock is moving forward, the receiver thinks

that the clock is moving backward.

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Example

Quantization

• Sampling results in a series of pulses of varying amplitude values ranging between two limits: a min and a max.

• The amplitude values are infinite between the two limits.

• We need to map the infinite amplitude values onto a finite set of known values.

• This is achieved by dividing the distance between min and max into L zones, each of height

• = (max - min)/L

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Quantization Levels

• The midpoint of each zone is assigned a value from 0 to

L-1 (resulting in L values)

• Each sample falling in a zone is then approximated to the

value of the midpoint.

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Quantization Zones

• Assume we have a voltage signal with amplitutes Vmin=-20V and Vmax=+20V.

• We want to use L=8 quantization levels.

• Zone width = (20 - -20)/8 = 5

• The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to +5, +5 to +10, +10 to +15, +15 to +20

• The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5

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Assigning Codes to Zones

• Each zone is then assigned a binary code.

• The number of bits required to encode the zones, or the number of bits per sample as it is commonly referred to, is obtained as follows:

• nb = log2 L

• Given our example, nb = 3

• The 8 zone (or level) codes are therefore: 000, 001, 010, 011, 100, 101, 110, and 111

• Assigning codes to zones: 000 will refer to zone -20 to -15

001 to zone -15 to -10, etc.

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Quantization and encoding of a sampled signal

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Quantization Error

• When a signal is quantized, we introduce an error - the coded signal is an approximation of the actual amplitude value.

• The difference between actual and coded value (midpoint) is referred to as the quantization error.

• The more zones, the smaller which results in smaller errors.

• BUT, the more zones the more bits required to encode the samples -> higher bit rate

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Quantization Error and SNQR

• Signals with lower amplitude values will suffer more from quantization error as the error range: /2, is fixed for all signal levels.

• Non linear quantization is used to alleviate this problem. Goal is to keep SNQR fixed for all sample values.

• Two approaches: The quantization levels follow a logarithmic curve. Smaller ’s at lower amplitudes

and larger’s at higher amplitudes.

Companding: The sample values are compressed at the sender into logarithmic zones, and then expanded at the receiver. The zones are fixed in height.

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Bit rate and bandwidth requirements of PCM

• The bit rate of a PCM signal can be calculated form the

number of bits per sample x the sampling rate

• Bit rate = nb x fs

• The bandwidth required to transmit this signal depends on the

type of line encoding used. Refer to previous section for

discussion and formulas.

• A digitized signal will always need more bandwidth than the

original analog signal. Price we pay for robustness and other

features of digital transmission.

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Example

• We want to digitize the human voice. What is the bit rate,

assuming 8 bits per sample?

• Solution

• The human voice normally contains frequencies from 0 to 4000 Hz.

So the sampling rate and bit rate are calculated as follows:

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PCM Decoder

• To recover an analog signal from a digitized signal we follow the

following steps:

We use a hold circuit that holds the amplitude value of a pulse till the next

pulse arrives.

We pass this signal through a low pass filter with a cutoff frequency that

is equal to the highest frequency in the pre-sampled signal.

• The higher the value of L, the less distorted a signal is recovered.

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Components of a PCM decoder

• We have a low-pass analog signal of 4 kHz. If we send the analog signal, we need a channel with a minimum

bandwidth of 4 kHz. If we digitize the signal and send 8 bits per sample, we need a channel with a minimum

bandwidth of 8 × 4 kHz = 32 kHz.27

Pulse Amplitude Modulation (PAM)

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Pulse Width Modulation (PWM)

• In Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time

Modulation (PTM) technique, the width or the duration or the time of the pulse carrier varies,

which is proportional to the instantaneous amplitude of the message signal.

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There are three types of PWM.

•The leading edge of the pulse being constant, the

trailing edge varies according to the message

signal. The waveform for this type of PWM is

denoted as (a) in the above figure.

•The trailing edge of the pulse being constant, the

leading edge varies according to the message

signal. The waveform for this type of PWM is

denoted as (b) in the above figure.

•The center of the pulse being constant, the

leading edge and the trailing edge varies

according to the message signal. The waveform

for this type of PWM is denoted as (c) shown in

the above figure.

Pulse Width Modulation

• In Pulse Width Modulation (PWM) or Pulse Duration

Modulation (PDM) or Pulse Time Modulation (PTM)

technique, the width or the duration or the time of the

pulse carrier varies, which is proportional to the

instantaneous amplitude of the message signal.

• There are three types of PWM.

• The leading edge of the pulse being constant, the

trailing edge varies according to the message signal.

The waveform for this type of PWM is denoted as (a) in

the above figure.

• The trailing edge of the pulse being constant, the

leading edge varies according to the message signal.

The waveform for this type of PWM is denoted as (b) in

the above figure.

• The center of the pulse being constant, the leading edge

and the trailing edge varies according to the message

signal. The waveform for this type of PWM is denoted as

(c) shown in the above figure.

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Pulse Position Modulation

• Pulse Position Modulation (PPM) is an analog modulation

scheme in which, the amplitude and the 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 the receiver in sync.

These sync pulses help to 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 edge 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.

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Comparison between PAM, PWM, and PPM

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PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends on the

width of the pulse

Bandwidth depends on the rise

time of the pulse

Bandwidth depends on the rise

time of the pulse

Instantaneous transmitter power

varies with the amplitude of the

pulses

Instantaneous transmitter power

varies with the amplitude and

the width of the pulses

Instantaneous transmitter power

remains constant with the width

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

modulation

It is similar to frequency

modulationIt is similar to phase modulation

Pulse Time Modulation

• Pulse Time Modulation (PTM) is a class of signaling technique

that encodes the sample values of an analog signal onto the

• time axis of a digital signal.

• The two main types of pulse time modulation are:

1.Pulse Width Modulation (PWM)

2.Pulse Position Modulation (PPM)

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Delta Modulation

• This scheme sends only the difference between pulses, if the pulse at time tn+1 is higher in amplitude value than the pulse at time tn, then a single bit, say a “1”, is used to indicate the positive value.

• If the pulse is lower in value, resulting in a negative value, a “0” is used.

• This scheme works well for small changes in signal values between samples.

• If changes in amplitude are large, this will result in large errors.

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The process of delta modulation

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Delta modulation components

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Delta demodulation components

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Adaptive Delta Modulation (ADM)

• it would be better if we can control the adjustment

of step-size, according to our requirement in order

to obtain the sampling in a desired fashion. This is

the concept of Adaptive Delta Modulation.

• Following is the block diagram of Adaptive delta

modulator.

• The gain of the voltage controlled amplifier is

adjusted by the output signal from the sampler. The

amplifier gain determines the step-size and both

are proportional.

• ADM quantizes the difference between the value of

the current sample and the predicted value of the

next sample. It uses a variable step height to

predict the next values, for the faithful reproduction

of the fast varying values.

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Differential PCM

• For 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 a predicted sampled

value, assumed from its previous output

and summarize them with the quantized

values.

• Such a process is called as Differential

PCM (DPCM) technique.

• The DPCM Transmitter consists of

Quantizer and Predictor with two

summer circuits. Following is the block

diagram of DPCM transmitter.

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• The signals at each point are named as −

• x(nTs) is the sampled input

• xˆ(nTs) is the predicted sample

• e(nTs) is the difference of sampled input and predicted output,

often called as prediction error

• v(nTs) is the quantized output

• u(nTs) is the predictor input which is actually the summer

output of the predictor output and the quantizer output

• The predictor produces the assumed samples from the previous outputs of the

transmitter circuit. The input to this predictor is the quantized versions of the input

signal x(nTs).

• Quantizer Output is represented as −

• v(nTs)=Q[e(nTs)]

• =e(nTs)+q(nTs)

• Where q (nTs) is the quantization error

• Predictor input is the sum of quantizer output and predictor output,

• u(nTs)=xˆ(nTs)+v(nTs)

• u(nTs)=xˆ(nTs)+e(nTs)+q(nTs)

• u(nTs)=x(nTs)+q(nTs)

• The same predictor circuit is used in the decoder to reconstruct the original input.

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DPCM Receiver

• The block diagram of DPCM Receiver consists of a

decoder, a predictor, and a summer circuit. Following

is the diagram of DPCM Receiver.

• The notation of the signals is the same as the previous

ones. In the absence of noise, the encoded receiver

input will be the same as the encoded transmitter

output.

• As mentioned before, the predictor assumes a value,

based on the previous outputs. The input given to the

decoder is processed and that output is summed up

with the output of the predictor, to obtain a better

output.

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On-off keying (OOK)

• Amplitude Shift Keying - ASK,

• Frequency Shift Keying - FSK,

• Phase Shift Keying - PSK,

• Binary Shift Keying - BSK –BPSK,BASK, BFSK

• Quadrature Phase Shift Keying - QPSK,

• Quadrature amplitude modulation - QAM

• Minimum shift keying - MSK

• Gaussian Minimum Shift Keying GMSK

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Amplitude Shift Keying

• 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.

• Any modulated signal has a high

frequency carrier. The binary signal

when ASK modulated, gives a zero

value for Low input while it gives the

carrier output for High input.

• The following figure represents ASK

modulated waveform along with its

input.

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ASK Modulator

• The ASK modulator block diagram comprises of

the carrier signal generator, the binary sequence

from the message signal and the band-limited

filter. Following is the block diagram of the ASK

Modulator.

• The carrier generator, sends a continuous high-

frequency carrier. The binary sequence from the

message signal makes the unipolar input to be

either High or Low. The high signal closes the

switch, allowing a carrier wave. Hence, the

output will be the carrier signal at high input.

When there is low input, the switch opens,

allowing no voltage to appear. Hence, the output

will be low.

• The band-limiting filter, shapes the pulse

depending upon the amplitude and phase

characteristics of the band-limiting filter or the

pulse-shaping filter.44

ASK Demodulator

There are two types of ASK Demodulation techniques. They are −

• Asynchronous ASK Demodulation/detection

• Synchronous ASK Demodulation/detection

The clock frequency at the transmitter when matches with the clock frequency at

the receiver, it is known as a Synchronous method, as the frequency gets

synchronized. Otherwise, it is known as Asynchronous.

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Asynchronous ASK Demodulator

• The Asynchronous ASK detector

consists of a half-wave rectifier, a low

pass filter, and a comparator.

Following is the block diagram for the

same.

• The modulated ASK signal is given to

the half-wave rectifier, which delivers a

positive half output. The low pass filter

suppresses the higher frequencies and

gives an envelope detected output

from which the comparator delivers a

digital output.

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Synchronous ASK Demodulator

Synchronous ASK detector consists of

a Square law detector, low pass filter,

a comparator, and a voltage li

The ASK modulated input signal is

given to the Square law detector. A

square law detector is one whose

output voltage is proportional to the

square of the amplitude modulated

input voltage. The low pass filter

minimizes the higher frequencies. The

comparator and the voltage limiter help

to get a clean digital output. miter.

Following is the block diagram for the

same.

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Frequency Shift Keying

• Frequency Shift Keying (FSK) is the

digital modulation technique in which

the frequency of the carrier signal

varies according to the digital signal

changes. FSK is a scheme of frequency

modulation.

• 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.

• The following image is the

diagrammatic representation of FSK

modulated waveform along with its

input.

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FSK Modulator

• The FSK modulator block diagram

comprises of two oscillators with a clock

and the input binary sequence. Following is

its block diagram.

• The two oscillators, producing a higher and

a lower frequency signals, are connected to

a switch along with an internal clock. To

avoid the abrupt phase discontinuities of the

output waveform during the transmission of

the message, a clock is applied to both the

oscillators, internally.

• The binary input sequence is applied to the

transmitter so as to choose the frequencies

according to the binary input.

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FSK Demodulator

• There are different methods for demodulating a FSK

wave. The main methods of FSK detection are

asynchronous detector and synchronous detector.

• The synchronous detector is a coherent one, while

asynchronous detector is a non-coherent one.

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Asynchronous FSK Detector

• The block diagram of Asynchronous FSK detector

consists of two band pass filters, two envelope

detectors, and a decision circuit. Following is the

diagrammatic representation.

• The FSK signal is passed through the two Band

Pass Filters (BPFs), tuned to Space and Mark

frequencies. The output from these two BPFs look

like ASK signal, which is given to the envelope

detector. The signal in each envelope detector is

modulated asynchronously.

• The decision circuit chooses which output is more

likely and selects it from any one of the envelope

detectors. It also re-shapes the waveform to a

rectangular one.

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Synchronous FSK Detector

• The block diagram of Synchronous FSK

detector consists of two mixers with local

oscillator circuits, two band pass filters and a

decision circuit. Following is the diagrammatic

representation.

• The FSK signal input is given to the two mixers

with local oscillator circuits. These two are

connected to two band pass filters. These

combinations act as demodulators and the

decision circuit chooses which output is more

likely and selects it from any one of the

detectors. The two signals have a minimum

frequency separation.

• For both of the demodulators, the bandwidth of

each of them depends on their bit rate. This

synchronous demodulator is a bit complex than

asynchronous type demodulators.52

Phase Shift Keying (PSK)

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.

• PSK is of two types, depending upon the phases the signal gets shifted. They are −

Binary Phase Shift Keying (BPSK)

• This is also called as 2-phase PSK or Phase Reversal Keying. In this technique, the sine wave carrier

takes two phase reversals such as 0° and 180°.

• BPSK is basically a Double Side Band Suppressed Carrier (DSBSC) modulation scheme, for message

being the digital information.

Quadrature Phase Shift Keying (QPSK)

• This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals

such as 0°, 90°, 180°, and 270°.

• If this kind of techniques are further extended, PSK can be done by eight or sixteen values also,

depending upon the requirement.

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BPSK Modulator

• The block diagram of Binary Phase Shift Keying

consists of the balance modulator which has the

carrier sine wave as one input and the binary

sequence as the other input. Following is the

diagrammatic representation.

• The modulation of BPSK is done using a balance

modulator, which multiplies the two signals applied at

the input. For a zero binary input, the phase will be 0°

and for a high input, the phase reversal is of 180°.

• Following is the diagrammatic representation of BPSK

Modulated output wave along with its given input.

• The output sine wave of the modulator will be the

direct input carrier or the inverted (180° phase shifted)

input carrier, which is a function of the data signal.

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BPSK Demodulator

• The block diagram of BPSK demodulator consists of

a mixer with local oscillator circuit, a band pass filter,

a two-input detector circuit. The diagram is as

follows.

By recovering the band-limited message signal, with

the help of the mixer circuit and the band pass filter,

the first stage of demodulation gets completed. The

base band signal which is band limited is obtained

and this signal is used to regenerate the binary

message bit stream.

In the next stage of demodulation, the bit clock rate

is needed at the detector circuit to produce the

original binary message signal. If the bit rate is a

sub-multiple of the carrier frequency, then the bit

clock regeneration is simplified. To make the circuit

easily understandable, a decision-making circuit may

also be inserted at the 2nd stage of detection.

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Quadrature Phase Shift Keying

• The Quadrature Phase Shift Keying (QPSK) is a variation of BPSK,

and it is also a Double Side Band Suppressed Carrier (DSBSC)

modulation scheme, which sends two bits of digital information at a

time, called as bigits.

• Instead of the conversion of digital bits into a series of digital stream,

it converts them into bit pairs. This decreases the data bit rate to half,

which allows space for the other users.

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QPSK Modulator

• The QPSK Modulator uses a bit-splitter, two

multipliers with local oscillator, a 2-bit serial

to parallel converter, and a summer circuit.

Following is the block diagram for the same.

At the modulator’s input, the message

signal’s even bits (i.e., 2nd bit, 4th bit, 6th bit,

etc.) and odd bits (i.e., 1st bit, 3rd bit, 5th bit,

etc.) are separated by the bits splitter and

are multiplied with the same carrier to

generate odd BPSK (called as PSKI) and

even BPSK (called as PSKQ). The PSKQ

signal is anyhow phase shifted by 90° before

being modulated.

The QPSK waveform for two-bits input is as

follows, which shows the modulated result

for different instances of binary inputs.

57

QPSK Demodulator

• The QPSK Demodulator uses two

product demodulator circuits with

local oscillator, two band pass filters,

two integrator circuits, and a 2-bit

parallel to serial converter. Following

is the diagram for the same.

• The two product detectors at the input

of demodulator simultaneously

demodulate the two BPSK signals.

The pair of bits are recovered here

from the original data. These signals

after processing, are passed to the

parallel to serial converter.58

Differential Phase Shift Keying

• n Differential Phase Shift Keying (DPSK) the

phase of the modulated signal is shifted relative to

the previous signal element. No reference signal

is considered here. The signal phase follows the

high or low state of the previous element. This

DPSK technique doesn’t need a reference

oscillator.

• The following figure represents the model

waveform of DPSK.

• It is seen from the above figure that, if the data bit

is Low i.e., 0, then the phase of the signal is not

reversed, but continued as it was. If the data is a

High i.e., 1, then the phase of the signal is

reversed, as with NRZI, invert on 1 (a form of

differential encoding).

• If we observe the above waveform, we can say

that the High state represents an M in the

modulating signal and the Low state represents a

W in the modulating signal.

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DPSK Modulator

• DPSK is a technique of BPSK, in which

there is no reference phase signal. Here,

the transmitted signal itself can be used as

a reference signal. Following is the

diagram of DPSK Modulator.

• DPSK encodes two distinct signals, i.e., the

carrier and the modulating signal with 180°

phase shift each. The serial data input is

given to the XNOR gate and the output is

again fed back to the other input through 1-

bit delay. The output of the XNOR gate

along with the carrier signal is given to the

balance modulator, to produce the DPSK

modulated signal.

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DPSK Demodulator

• In DPSK demodulator, the phase of the

reversed bit is compared with the phase of

the previous bit. Following is the block

diagram of DPSK demodulator.

• From the above figure, it is evident that the

balance modulator is given the DPSK signal

along with 1-bit delay input. That signal is

made to confine to lower frequencies with

the help of LPF. Then it is passed to a

shaper circuit, which is a comparator or a

Schmitt trigger circuit, to recover the original

binary data as the output.

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Quadrature Amplitude Modulation

• PSK is limited by the ability of the equipment

to distinguish between small differences in

phases.

Limits the potential data rate.

• Quadrature amplitude modulation is a

combination of ASK and PSK so that a

maximum contrast between each signal unit

(bit, dibit, tribit, and so on) is achieved.

We can have x variations in phase and y

variations of amplitude

x • y possible variation (greater data rates)

• Numerous variations. (4-QAM, 8-QAM)

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# of phase shifts > # of amplitude shifts

MSK Modulation

he problem can be overcome in part by filtering the signal,

but is found that the transitions in the data become

progressively less sharp as the level of filtering is increased

and the bandwidth reduced. To overcome this problem

GMSK is often used and this is based on Minimum Shift

Keying, MSK modulation. The advantage of which is what

is known as a continuous phase scheme. Here there are no

phase discontinuities because the frequency changes

occur at the carrier zero crossing points.

When looking at a plot of a signal using MSK modulation, it

can be seen that the modulating data signal changes the

frequency of the signal and there are no phase

discontinuities. This arises as a result of the unique factor

of MSK that the frequency difference between the logical

one and logical zero states is always equal to half the data

rate. This can be expressed in terms of the modulation

index, and it is always equal to 0.5.

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GMSK Modulation - Gaussian Minimum Shift Keying

There are two main ways in which GMSK modulation can

be generated. The most obvious way is to filter the

modulating signal using a Gaussian filter and then apply this

to a frequency modulator where the modulation index is set

to 0.5. This method is very simple and straightforward but it

has the drawback that the modulation index must exactly

equal 0.5. In practice this analogue method is not suitable

because component tolerances drift and cannot be set

exactly.

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There are two main ways in which

GMSK modulation can be generated.

The most obvious way is to filter the

modulating signal using a Gaussian filter

and then apply this to a frequency

modulator where the modulation index is

set to 0.5. This method is very simple

and straightforward but it has the

drawback that the modulation index must

exactly equal 0.5. In practice this

analogue method is not suitable because

component tolerances drift and cannot

be set exactly.

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8-QAM and 16-QAM

First example handles noise best

Because of ratio of phases to amplitudes

ITU-T recommendation.

Second example, recommendation of OSI.

not all possibilities are used, to increase

readability of signal, measurable differences

between shifts are increased

References

Book:

1. Taub & Schiling “Principles of Communication Systems” Tata McGraw hill 2007.

2. Kennedy and Davis “Electronic Communication Systems” Tata McGraw hill, 4th Edition, 1993.

3. Sklar “Digital Communication Fundamentals and Applications“ Pearson Education, 2001.

4. TG Thomas and S Chandra Sekhar, “Communication Theory” Tata McGraw hill 2006.

Web:

https://www.tutorialspoint.com/analog_communication/analog_communication_pulse_modulation.htm

http://www.rfwireless-world.com/Terminology/MSK-GMSK.html

http://www.radio-electronics.com/info/rf-technology-design/pm-phase-modulation/what-is-gmsk-gaussian-m

https://www.tutorialspoint.com/digital_communication/digital_communication_techniques.htm

PPT:

www.ics.uci.edu/~magda/Courses/netsys270/ch4_2_v1.ppt

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68Dr Gnanasekaran Thangavel12/12/2017

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Thank You

Questions and Comments?