(EC114) DIGITAL COMMUNICATION

Hours / Week Marks

Year Semester C
L T P/D CIE SEE Total
III II 4 1 - 4 30 70 100

Pre-requisite Analog Communication, Probability Theory and Stochastic Process

COURSE OBJECTIVES
Students will be able to
1. Recall different types of samplingtechniques
2. Describe various pulse modulation techniques such as PCM, DPCM, DM andADM.
3. Analyze various digital modulation techniques such as ASK, BPSK, BFSK, DPSK,
QPSK and M-arysignalling.
4. Interpret Inter symbol interference in AWGNchannels.
5. Illustrate the concept of spreadspecturm.

COURSE OUTCOMES
At the end of the course, the student will be able to
1. Apply differentsamplingtechniques to convert analog signal to discrete sequence.
2. Analyze channel capacity ofdifferent communication channels
3. Compute signal-to-noise ratio in various pulse modulationtechniques
4. Construct solution for Inter symbol interference in band limitedchannels
5. Evaluate probability of error in various digital modulationtechniques
6. Describe the applications of Spread Spectrum..

UNIT – I
Introduction to Digital Communication Systems, Block diagram of digital communication
system, Channel capacity, Shannon's limit, sampling Theorem -Mathematical proof of sampling
and reconstruction –ideal and Flat top sampling, Band pass sampling.

UNIT – II
Pulse digital modulation :Pulse code modulation, generation and detection of PCM, Uniform
quantization and companding, Differential PCM; Delta modulation, Adaptive delta modulation;
Signal-to- Noise Ratio calculations in PCM, DM

UNIT – III
Base band data transmission: Communication over Band limited AWGN Channel, ISI in
band limited channels, Zero-ISI condition- the Nyquist criterion, Solution for zero ISI, Raised
cosine filters, Partial response signalling-Duo binary encoding, M-ary baseband system, eye
pattern, adaptive Equalization.

UNIT – IV
Digital carrier modulation schemes :Optimum Receiver for AWGN channel, Matched filter
and Correlation receivers, Digital Modulations-Techniques, Generation and detection of ASK,
BPSK and BFSK, QPSK and DPSK, probability of bit error computation for BPSK, BFSK,
QPSK, M-arysignalling schemes, comparison of Modulation techniques.
UNIT – V
Spread spectrum modulation : Need for Spread Spectrum Modulation, PN sequence and its
properties, Direct sequence SS system- DS/BPSK Transmitter & Receiver, Processing gain,
Jamming margin, Frequency hop SS system- FH-FSK transmitter and Receiver, Fast and slow
hop, Application of SS, CDMA, Multipath fading.

TEXT BOOKS
1. Sam Shanmugam, “Digital and analog communication system”, John Wiley,2005.
2. Herbert Taud, Donald L. Schiling, GoutamSaha, “Principles of Communication Systems”, –
3rdEdition, McGraw – Hill2008.

REFERENCE BOOKS
1. Digital Communications –Simon Haykin, Jon Whiley,2005
2. Wayne Tomasi “Electronic communications systems”-5thedition,pearson publication.
 Unit-I
Introduction to Digital Communication Systems

What Does Communication (or Telecommunication) Mean?

The term communication (or telecommunication) means the transfer of

some form of information from one place (known as the source of
information) to another place (known as the destination of information)
using some system to do this function (known as a communication system).
So What Will we Study in This Course?
In this course, we will study the basic methods that are used for
communication in upon the successful completion of this course, you should
be able to identify the different communication techniques, know the
advantages and disadvantages of each technique, and show the basic construction
of the systems that implement these communication techniques.
Old Methods of Communication

Pigeons
Smoke
Fire
Post Office
Drums

Problems with Old Communication Methods

Slow
Difficult and relatively expensive
Limited amount of information can be sent.
Some methods can be used at specific times of the day
Information is not secure.

All of the following are electric (or electromagnetic) communication systems

Sa

Fast
Easy to use and very cheap
Huge amounts of information can be transmitted
Secure transmission of information can easily be achieved
Can be used 24 hours a day.
Basic Construction of Electrical Communication System

A communication system may transmit information in one direction such as

TV and radio (simplex), two directions but at different times such as the CB (half-
duplex), or two directions simultaneously such as the telephone (full-duplex).

Basic Terminology Used in this Communications Course

A Signal: is a function that specifies how a specific variable changes versus

an independent variable such as time, location, height (examples: the
age of people versus their coordinates on Earth, the amount of
money in your bank account versus time).

A System: operates on an input signal in a predefined way to generate an

output signal.

Analog Signals: are signals with amplitudes that may take any real value out of an
infinite number of values in a specific range (examples: the height
of mercury in a 10cm long thermometer over a period of time is a
function of time that may take any value between 0 and 10cm, the
weight of people setting in a class room is a function of space (x
and y coordinates) that may take any real value between 30 kg to
200 kg (typically)).

Digital Signals: are signals with amplitudes that may take only a specific
number of values (number of possible values is less than
infinite) (examples: the number of days in a year versus the year
is a function that takes one of two values of 365 or 366 days,
number of people sitting on a one-person chair at any instant of
time is either 0 or 1, the number of students registered in
 different classes at KFUPM is an integer number between 1 and
100).

Noise: is an undesired signal that gets added to (or sometimes multiplied

with) a desired transmitted signal at the receiver. The source of
noise may be external to the communication system (noise resulting
from electric machines, other communication systems, and noise
from outer space) or internal to the communication system (noise
resulting from the collision of electrons with atoms in wires and
ICs).

Signal to Noise Ratio (SNR): is the ratio of the power of the desired signal to the
power of the noise signal.

Bandwidth (BW): is the width of the frequency range that the signal occupies. For
example the bandwidth of a radio channel in the AM is around 10
kHz and the bandwidth of a radio channel in the FM band is 150
kHz.
Rate of communication: is the speed at which digital information transmitted.
The maximum rate at which most of today’s modem receives the
information is around 56 k bits/second and transmit digital
information is around 33 k bits/second. A Local Area Network
(LAN) can theoretically receive/transmit information at a rate of 100
M bits/s. Gigabit networks would be able to receive/transmit
information at least 10 times that rate.
Modulation: is changing one or more of the characteristics of a signal (known
as the carrier signal) based on the value of another signal
(known as the information or modulating signal) to produce a
modulated signal.
Analog and Digital communication

Since the introduction of digital communication few decades ago, it has been
gaining a steady increase in use. Today, you can find a digital form of almost all
types of analog communication systems. For example, TV channels are now
broadcasted in digital form (most if not all Ku band satellite TV transmission is
digital). Also, radio now is being broadcasted in digital form (see sirus.com and
xm.com). Home phone systems are starting to go digital (a digital phone system is
available at KFUPM). Almost all cellular phones are now digital, and so on. So, what
makes digital communication more attractive compared to analog communication?

Advantages of Digital Communication over Analog Communication

Immunity to Noise (possibility of regenerating the original digital signal if

signal power to noise power ratio (SNR) is relatively high by using of
devices called repeaters along the path of transmission).
Efficient use of communication bandwidth (through use of techniques like
compression).
Digital communication provides higher security (data encryption). The
ability to detect errors and correct them if necessary.
Design and manufacturing of electronics for digital communication
s ys t em s is much easier and much cheaper than the design and
manufacturing of electronics for analog communication systems.

Modulation:

Famous Types

Amplitude Modulation (AM): varying the amplitude of the carrier based

on the information signal as done for radio
channels that are transmitted in the AM radio
band.
Phase Modulation (PM): varying the phase of the carrier based
on the information signal.
Frequency Modulation (FM): varying the frequency of the carrier based on
the information signal as done for channels
transmitted in the FM radio band.

Purpose of Modulation:

For a signal (like the electric signals coming out of a microphone) to

be transmitted by an antenna, signal wavelength has to be comparable to the
length of the antenna (signal wavelength is equal to 0.1 of the antenna
length or more). If the wavelength is extremely long, modulation must
be used to reduce the wavelength of the signal to make the length of the
required antenna practical.
To receive transmitted signals from multiple sources without interference
between them, they must be transmitted at different frequencies
 (frequency multiplexing) by modulating carriers that have different
frequencies with the different information signals.

Exercise 1 1: Specify if the following communication systems are (A)nalog or

(D)igital:

a) TV in the 1970s:

b) TV in the 2030s:

c) Fax machines

d) Local area networks (LANs):

e) First generation cellular phones

f) Second generation cellular phones

g) Third generation cellular phones.

Channel Capacity:

Channel capacity, as we will prove later, specifies the highest rate (number of bits per channel
use signal) at which information can be sent with arbitrarily low error. The problem of data
transmission (over a noisy channel) is dual to data compression. During compression we remove
redundancy in the data, while during data transmission we add redundancy in a controlled fashion
to fight errors in the channel.

Noiseless binary channel:

Noisy channel with non-overlapping outputs:

Binary Symmetric Channel:
Binary erasure channel:
Symmetric channels:
Properties of Channel Capacity:
Unit-II:Pulse digital modulation
 Unit-III
Base band data transmission
Inter-symbol Interference
(ISI)
Unlike analog signals, which are usually smooth in nature, digital signals are composed of pulses with
often vertical transitions. The fact that digital signals sometimes have vertical transitions increases
their bandwidth significantly since it requires infinite bandwidth to represent a signal with vertical
transitions. Compare for example the bandwidth of two baseband signals given by a sine wave with
frequency f 0 and a square wave with frequency f 0 . The sine wave has a single frequency component
at f 0 Hz. However, the square wave has infinite frequency components at f 0 and integer multiples of
it. If we consider the bandwidth of a signal to be the minimum frequency that encloses all frequency
components of the signal (the signal has no frequency components at all above that frequency), then
the sine wave will have a bandwidth of f 0 Hz because it has no frequency components above that
frequency, while the square wave has an infinite bandwidth because it theoretically has frequency
components that extend to infinity.

The fact that any communication system has limited bandwidth to transmit digital data indicates that
certainly a transmitted square pulse will be received differently at the receiver as the channel will
filter some components of it. The difference depends on how narrow the bandwidth of the channel
compared to the symbol rate in the signal. The effect of filtering part of the transmitted signal by the
channel on the quality of the received signal may be significant that a phenomenon called “Inter-
symbol Interference (ISI)” occurs. ISI causes the transmitted pulses to get mixed together, meaning
that a pulse that is transmitted between time instants will smear into adjacent pulses affecting the
process of data detection and possibly causing errors not as a result of noise but as a result of symbols
mixing together.

Effect of Channel Bandwidth Limitation on ISI

Consider a baseband digital signal with symbol rate Rs symbols/second that is composed of a sum of
square pulses and that is transmitted through different baseband channels with different bandwidths:

1. Channel with Infinite Bandwidth: Such a channel passes all signal components. In this
case, the received signal will be exactly the same as the transmitted square wave since the
complete signal is passed. So, the transmitted data will not experience any ISI at all.
 A

Rs f
2

2. Wideband Channel with channel larger than Rs/2: the bandwidth of the channel in this
case is wide but not infinite, so a relatively large amount of the signal power will pass
and a small amount at high frequencies will be rejected. The data in this case experiences
some ISI but data can easily be recovered since the ISI is limited.

t
f
Rs
0 2

3. Channel Bandwidth is Equal to One Half Symbol Rate Rs/2: The first null (zero) in
the power spectrum density of transmitted data occurs at one half the sample rate Rs/2.
The received signal in this situation experiences significant amount of ISI. However, the
data is still recoverable using some signal processing algorithms. This represents the
minimum channel bandwidth that would allow us to recover the data completely. Any
channel bandwidth below this would cause a problem.

t
f
Rs
0
2
 4. Channel Bandwidth is Lower than One Half Symbol Rate Rs/2: in this case, the ISI is
huge and loss of data will occur. It is not possible to recover back the data completely no
matter what signal processing algorithms are used.

t
f
Rs
0 2

We see from the previous 4 cases that when transmitting square pulses and the bandwidth of the
channel is not infinite, then ISI will occur. However, as long as the bandwidth of the channel is
greater than one half the symbol rate, data can be recovered but possibly using some signal processing
algorithms to remove the effect of ISI. If the channel bandwidth is less than that, then loss of data will
certainly occur.

Pulse Shaping to Control ISI

The use of rectangular-shape pulses to transmit digital information makes sense because they have
flat tops which fit the shapes of digital signals perfectly. In addition, a rectangular pulse that extends
over a bit (or symbol) period avoids interference between consecutive pulses as long as the exact
shape of the pulses is preseved. However, the power spectral density of rectangular-pulse shapes is
very wide (remember that the spectrum of a rectangular pulse is a “sinc” function). The wide
spectrum of rectangular pulses means that such pulses must be transmitted over very wideband
channels even for relatively low bit (or symbol) rates or else part of the transmitted signal will be
filtered out by the channel and the received signal will be a distorted version of the transmitted signal.
Filtering out part of the transmitted signal results in the rectangular pulses getting mixed up with
preceding and succeeding pulses in what we called above Inter-Symbol Interference (ISI).

To combat ISI, the pulses that we use to transmit data must have limited bandwidth so that when
transmitted over limited bandwidth channels, the complete spectrum of these signals is retained and
no part of it is filtered out. This will guarantee that the signal does not change as it is transmitted
through the channel. However, limiting the bandwidth of the pulses we use to transmit data causes
their duration in time to be infinite (remember that time limited signals are frequency unlimited and
frequency limited signals are time unlimited). A pulse with an infinite time duration (or at least very
long time duration) means that each pulse extends over a very large number of bit periods. This is not
necessarily bad if the pulse is designed properly. What we mean by designed properly is that each
pulse needs to be equal to a constant (1 V) at the time instant of the start of the bit that this pulse
represents and at which this bit will be sampled and be zero (0 V) at all time instants of future and
past bits so not to interfere with these bits at the moments that they are sampled for detection. A class
of pulses called “Nyquest Pulses” satisfies all these requirements. A famous class of Nyquest pulses
is called “Raised Cosine” pulses

Raised Cosine Pulses

The class of Raised Cosine pulses include the famous “sinc” function. Although the “sinc” The “sinc”
function has the narrowest bandwidth of all Nyquest pulses, it decays at a very slow rate that is
proportional to 1/t. This means that the generation of the “sinc” pulse corresponding to a specific
symbol must start many symbol periods before the time of the symbol represented by this pulse and
must continue for many symbol periods after the time of the symbol represented by this pulse. This
exerts a relatively large computational requirements on the system in additional to a delay before and
after the transmission of data. Other Raised Cosine pulses provide a compromise between the
bandwidth (they require more bandwidth than the “sinc” pulse) with the length of tails of the pulse
(they have much shorter tails than the “sinc” pulse that extend only few symbol periods before and
after the time of their symbol).

The general format for a raised cosine pulse is

⎛ πt ⎞ ⎛ πα t ⎞
sin ⎜ ⎟ cos ⎜ ⎟
sRC (t ) = ⎝ Ts ⎠
⋅ ⎝ Ts ⎠
πt ⎛ 4α t ⎞
2

1− ⎜ ⎟
⎝ 2Ts ⎠

where α is a parameter that provides the tradeoff between the bandwidth and tail length of the
raised cosine function, and Ts is the symbol period. The first component in the raised cosine pulse
shown above is a “sinc” pulse. The tails of the “sinc” pulse are attenuated further by the second
2 3
component at the rate of t . So the raised cosine tails drop at the rate of t which means that for a
properly designed raised cosine, the tails die out after few (3 to 5) bit or symbol periods only. The
raised cosine pulse becomes the “sinc” when the parameter α = 0 .

The spectrum of raised cosine pulses is

 ⎧
0≤ f ≤
(1 − α )
⎪1
⎪ 2Ts
⎪ ⎡ ⎛ π ( 2Ts f − 1 + α ) ⎞ ⎤ (1 − α )
⎪1
S RC ( f ) = ⎨ ⎢1 + cos ⎜ ≤ f ≤
(1 + α )
⎟⎥
⎜ 2α ⎟⎥
⎪ 2 ⎢⎣ ⎝ ⎠⎦ 2Ts 2Ts
⎪
⎪0 (1 + α ) < f
⎪⎩ 2Ts

The spectrum is divided into three regions that are shown in the figure below

S RC ( f )

− (1 + α ) −1 − (1 − α ) (1 − α ) 1 (1 + α )
2Ts 2Ts 2Ts 2Ts 2Ts 2Ts

The spectrums and time-domain pulse shapes of several raised cosine pulses are shown below for
different values of α .

S RC ( f )

α =0
α = 0.5
α =1

Rs 1 1
= Rs =
2 2Ts Ts
 sRC (t )
α =0

α = 0.5
α =1

t
−4Ts −3Ts −2Ts −Ts Ts 2Ts 3Ts 4Ts

For baseband transmission , the symbol rate of the transmitted data that can be transmitted using a
Raised Cosine pulse is related to α and the bandwidth of the signal B by the relation

1 2B
Rs = = (baseband transmission)
Ts 1 + α

and for passband transmission, the rate is half of the above value, or

1 B
Rs = = (passband transmission)
Ts 1 + α

Important Notes:

1. The required bandwidth for transmitting a digital data signal is a function of the symbol rate
Rs not the bit rate Rb.
2. For a signal with
M = 2N Bits/Symbol
N Bits/Symbol = log 2 M
1
Tb = Ts
N
Rb = N ⋅ R s
Relation between Probability of Error and C/N Ratio

Unlike analog signals in which quality of the signal is measured in terms of the received signal power
relative to the noise power, the quality of digital signals is measured in terms of number of errors that
occur in the received data as a result of added noise. The probability of error (also called bit error
rate) is related to the C/N ratio of the received signal. To find this relation, let us consider a baseband
binary transmission (for the case of passband transmission the same concept stands) where we are
transmitting one of two pulses that have amplitudes +1 V and –1 V with equal probability. That is

T 1 = +1 V Prob(Transmitting T1) = 0.5

T 2 = −1 V Prob(Transmitting T2) = 0.5

Clearly, in the absence of any thermal noise or other sources of noise, the received pulses
corresponding to the above transmissions are

R 1 = +1 V (No Noise)

R 2 = −1 V (No Noise)

So, clearly the transmitted data can be recovered with zero probability of error.

In the presence of thermal noise, the received pulses become accompanied by normally distributed
noise (noise that has probability density function that follows the Gaussian distribution). So, the
received signals become:

R 1 = +1 + N V (with Noise)

R 2 = −1 + N V (with Noise)

where N is a Gaussian random variable. The probability density functions of the two random
variables R1 and R2 will be similar to the probability density function of N (a zero-mean Gaussian
random variable) except that the means of two will be +1 and –1, respectively. This is shown below:
 

The bit error probability Pb can be computed using conditional probability as

Pb = P ( Detecting R1 T 2 was transmitted ) ⋅ P (T 2 was transmitted )

+ P ( Detecting R 2 T 1 was transmitted ) ⋅ P (T 1 was transmitted )

Since Prob(Transmitting T1) = Prob(Transmitting T2) = 0.5 and since the above two areas are equal
to each other, then the bit error probability Pb becomes

Pb = P ( Detecting R1 T 2 was transmitted ) = P ( Detecting R 2 T 1 was transmitted )

which is equivalent to the area in probability density function of the zero mean Gaussian random
variable shown below:

Therefore, the probability of bit error can be written as

1 ∞
−u 2
Pb = ∫e du
π x
This integration does not have a closed form. Instead, it is often expressed in terms of a function
called the “error function complement (erfc). The above expression becomes

1 ∞
−u 2
Pb = ∫e du
π x

1
= erfc ( x )
2

What is x in the expression above? The value of x is what determines the probability of bit error.
This quantity is related to the signal power and noise power. In fact, this quantity is expressed as

Eb
x =
N0

where Eb is the bit energy (energy contained in a bit) while N0 is the thermal noise power per unit Hz
of bandwidth.

Eb C
Relation between N0 and N

You may ask, why is the probability of bit error expressed in terms of Eb/N0 ratio and not in terms of
C/N ?

The answer is simple. Assuming equal amounts of noise power are added to the digital signals
transmitted by two different systems, the probability of error in the received data of the two systems
may be different. The reason is that although both systems are transmitting equal amounts of power,
what counts is how much energy is allocated per bit in each system. To see this, consider that one of
the systems transmits much more data than the other, yet the transmitted power by each system is the
same. Clearly, the system that transmits more data allocates smaller amounts of energy per bit, and it
is expected that the probability of bit error for that system would be worse (higher proability of bit
error). However, this does not mean that the system has a worse performance than the other. So, to
have fare comparison, it is important to compare two systems with equal bit energy rather than equal
transmitted power. Now, consider two systems that transmit equal amounts of data. However, one of
them uses much more bandwidth than the other. Clearly, the system that uses more bandwidth may
have a lower probability of bit error because of the fact that it is using wider bandwidth. Also, to have
fair comparison, the thermal noise should be evaluated in terms of noise per Hz. For both of these, it
is seen that what determines the probability of bit error is Eb/N0 ratio and not C/N ratio.

The ratios of Eb/N0 and C/N can be related to each other by observing that Energy = Power * Time
and that Noise per Hz = Total Noise / Bandwidth, or
 C = Carrier Power (W)
C
E b = Bit Energy (J) = C ⋅T b =
Rb
N = Noise Power (W)
N
N 0 = Noise Power per Hz (W/Hz = J) =
BW

Therefore,

⎛C ⎞
⎜ ⎟
= ⎝ ⎠
Eb C / Rb N
=
N 0 N / BW ⎛ R b ⎞
⎜ ⎟
⎝ BW ⎠

Rb
The quantity represents the total bit rate of the system divided by the amount of bandwidth the
BW
system uses to transmit this data. This is called the “Throughput” of the communication system,
which is the number of bits/s that the system transmits in each Hz of bandwidth that is allocated for it,
or

Rb
   Throughput =  
BW

Comparison of Different Systems

a) BPSK

T s = Tb
    R s = Rb
1
Throughput BPSK, Zero-ISI =
2

a) 4-PSK (QPSK)
 T s = 2T b
1
Rs = Rb
2
Throughput QPSK, Zero-ISI = 1
   
Ps,QPSK = 2Ps,BPSK = 2Pb,BPSK
1
Pb,QPSK = Pb,QPSK
2
Pb,QPSK = Pb,BPSK

The benefit of using QPSK over BPSK is that a higher bit rate can be achieved without
any deterioration in bit error probability performance.

b) 8-PSK

T s = 3T b
1
R s = Rb
3
    Throughput QPSK, Zero-ISI = 1.5
Ps, 8-PSK = 4Ps, BPSK
3
Pb, 8-PSK = Pb, BPSK
2
 Unit-IV:Digital carrier modulation schemes
Unit-IV Unit-IV UNIT-2IV

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.

if the information signal is digital and the amplitude (lV of the carrier is varied proportional to
the information signal, a digitally modulated signal called amplitude shift keying (ASK) is
produced.
If the frequency (f) is varied proportional to the information signal, frequency shift keying (FSK) is
produced, and if the phase of the carrier (0) is varied proportional to the information signal,
phase shift keying (PSK) is produced. If both the amplitude and the phase are varied proportional to
the information signal, quadrature amplitude modulation (QAM) results. ASK, FSK, PSK, and
QAM are all forms of digital modulation:

a simplified block diagram for a digital modulation system.

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.
Mathematically, amplitude-shift keying is

where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)

In above Equation, the modulating signal [vm(t)] is a normalized binary waveform, where + 1 V =
logic 1 and -1 V = logic 0. Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to

Mathematically, amplitude-shift keying is (2.12) where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts) A/2 = unmodulated carrier amplitude (volts)
 = analog carrier
signal [vm(t)] is a normalized binary waveform, where + 1 V = logic 1 and -1 V = logic 0.
Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to and for a logic 0 input, vm(t)
= -1 V,Equation reduces to

Thus, the modulated wave vask(t),i Hence, the carrier is either "
"off," which is why amplitude-shift keying is sometimes referred to as on-off keying (OOK).
it can be seen that for every change in the input binary data stream, there is one change in the ASK
waveform, and the time of one bit (tb) equals the time of one analog signaling element (t,).
B = fb/1 = fb baud = fb/1 = fb

Example :
Determine the baud and minimum bandwidth necessary to pass a 10 kbps binary signal using
amplitude shift keying. 10Solution For ASK, N = 1, and the baud and minimum bandwidth are
determined from Equations 2.11 and 2.10, respectively:

B = 10,000 / 1 = 10,000
baud = 10, 000 /1 = 10,000
The use of amplitude-modulated analog carriers to transport digital information is a relatively low-
quality, low-cost type of digital modulation and, therefore, is seldom used except for very low-
speed telemetry circuits.
ASK TRANSMITTER:
The input binary sequence is applied to the product modulator. The product modulator amplitude

Coherent ASK DETECTOR:

FREQUENCY SHIFT KEYING

The frequency of the output signal will be either high or low, depending upon the input data
applied.

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.

FSK is a form of constant-amplitude angle modulation similar to standard frequency modulation

(FM) except the modulating signal is a binary signal that varies between two discrete voltage levels
rather than a continuously changing analog waveform.Consequently, FSK is sometimes called
binary FSK (BFSK). The general expression for FSK is
where

vfsk(t) = binary FSK waveform

Vc = peak analog carrier amplitude (volts)

fc = analog carrier center frequency (hertz)

f=peak change (shift)in the analog carrier frequency(hertz)

vm(t) = binary input (modulating) signal (volts)

From Equation 2.13, it can be seen that the peak shift in the carrier frequency ( f) is proportional to
the amplitude of the binary input signal (vm[t]), and the direction of the shift is determined by the
polarity.

The modulating signal is a normalized binary waveform where a logic 1 = + 1 V and a logic 0 = -1
V. Thus, for a logic l input, vm(t) = + 1, Equation 2.13 can be rewritten as

For a logic 0 input, vm(t) = -1, Equation becomes

With binary FSK, the carrier center frequency (fc) is shifted (deviated) up and down in the
frequency domain by the binary input signal as shown in Figure 2-3.

FIGURE: FSK in the frequency domain

As the binary input signal changes from a logic 0 to a logic 1 and vice versa, the output frequency
shifts between two frequencies: a mark, or logic 1 frequency (fm), and a space, or logic 0 frequency
(fs). The mark and space frequencies are separated from the carrier frequency by the peak frequency
deviation ( f) and from each other by 2 f.

Frequency deviation is illustrated in Figure 2-3 and expressed mathematically as

f = |fm fs| / 2 (2.14)

where f = frequency deviation (hertz)

|fm fs| = absolute difference between the mark and space frequencies (hertz)

Figure 2-4a shows in the time domain the binary input to an FSK modulator and the corresponding
FSK output.

When the binary input (fb) changes from a logic 1 to a logic 0 and vice versa, the FSK output
frequency shifts from a mark ( fm) to a space (fs) frequency and vice versa.

In Figure 2-4a, the mark frequency is the higher frequency (fc + f) and the space frequency is the
lower frequency (fc - f), although this relationship could be just the opposite.

Figure 2-4b shows the truth table for a binary FSK modulator. The truth table shows the input and
output possibilities for a given digital modulation scheme.
FSK Bit Rate, Baud, and Bandwidth

In Figure 2-4a, it can be seen that the time of one bit (tb) is the same as the time the FSK output is a
mark of space frequency (ts). Thus, the bit time equals the time of an FSK signaling element, and
the bit rate equals the baud.

The baud for binary FSK can also be determined by substituting N = 1 in Equation 2.11:

baud = fb / 1 = fb
The minimum bandwidth for FSK is given as
B= |(fs fb) (fm fb)|

=|(fs fm)| + 2fb

and since |(fs fm)| equals 2 f, the minimum bandwidth can be approximated as

B= 2( f + fb) (2.15)
where
B= minimum Nyquist bandwidth (hertz)
f= frequency deviation |(fm fs)| (hertz)
fb = input bit rate (bps)
Example 2-2

Determine (a) the peak frequency deviation, (b) minimum bandwidth, and (c) baud for a binary
FSK signal with a mark frequency of 49 kHz, a space frequency of 51 kHz, and an input bit rate of
2 kbps.

Solution

a. The peak frequency deviation is determined from Equation 2.14:

f= |149kHz - 51 kHz| / 2 =1 kHz

b. The minimum bandwidth is determined from Equation 2.15:
B = 2(100+ 2000)
=6 kHz
c. For FSK, N = 1, and the baud is determined from Equation 2.11 as
baud = 2000 / 1 = 2000

FSK TRANSMITTER:
Figure 2-6 shows a simplified binary FSK modulator, which is very similar to a conventional FM
modulator and is very often a voltage-controlled oscillator (VCO).The center frequency (fc) is
chosen such that it falls halfway between the mark and space frequencies.

A logic 1 input shifts the VCO output to the mark frequency, and a logic 0 input shifts the VCO
output to the space frequency. Consequently, as the binary input signal changes back and forth
between logic 1 and logic 0 conditions, the VCO output shifts or deviates back and forth between
the mark and space frequencies.

FIGURE 2-6 FSK modulator

A VCO-FSK modulator can be operated in the sweep mode where the peak frequency deviation is
simply the product of the binary input voltage and the deviation sensitivity of the VCO.
With the sweep mode of modulation, the frequency deviation is expressed mathematically as
f = vm(t)kl (2-19)
vm(t) = peak binary modulating-signal voltage (volts)
kl = deviation sensitivity (hertz per volt).
FSK Receiver
FSK demodulation is quite simple with a circuit such as the one shown in Figure 2-7.

FIGURE 2-7 Noncoherent FSK demodulator

The FSK input signal is simultaneously applied to the inputs of both bandpass filters (BPFs)
through a power splitter.The respective filter passes only the mark or only the space frequency on to
its respective envelope detector.The envelope detectors, in turn, indicate the total power in each
passband, and the comparator responds to the largest of the two powers.This type of FSK detection
is referred to as noncoherent detection.

Figure 2-8 shows the block diagram for a coherent FSK receiver.The incoming FSK signal is
multiplied by a recovered carrier signal that has the exact same frequency and phase as the
transmitter reference.

However, the two transmitted frequencies (the mark and space frequencies) are not generally
continuous; it is not practical to reproduce a local reference that is coherent with both of them.
Consequently, coherent FSK detection is seldom used.

FIGURE 2-8 Coherent FSK demodulator

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.

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 DSB-SC (Double Sideband Suppressed Carrier) modulation scheme, for
message being the digital information.

Following is the image of BPSK Modulated output wave along with its input.
Binary Phase-Shift Keying
The simplest form of PSK is binary phase-shift keying (BPSK), where N = 1 and M =
2.Therefore, with BPSK, two phases (21 = 2) are possible for the carrier.One phase represents a
logic 1, and the other phase represents a logic 0. As the input digital signal changes state (i.e., from
a 1 to a 0 or from a 0 to a 1), the phase of the output carrier shifts between two angles that are
separated by 180°.

Hence, other names for BPSK are phase reversal keying (PRK) and biphase modulation. BPSK
is a form of square-wave modulation of a continuous wave (CW) signal.

FIGURE 2-12 BPSK transmitter

BPSK TRANSMITTER:

Figure 2-12 shows a simplified block diagram of a BPSK transmitter. The balanced modulator acts
as a phase reversing switch. Depending on the logic condition of the digital input, the carrier is
transferred to the output either in phase or 180° out of phase with the reference carrier oscillator.

Figure 2-13 shows the schematic diagram of a balanced ring modulator. The balanced modulator
has two inputs: a carrier that is in phase with the reference oscillator and the binary digital data. For
the balanced modulator to operate properly, the digital input voltage must be much greater than the
peak carrier voltage.

This ensures that the digital input controls the on/off state of diodes D1 to D4. If the binary input is
a logic 1(positive voltage), diodes D 1 and D2 are forward biased and on, while diodes D3 and D4
are reverse biased and off (Figure 2-13b). With the polarities shown, the carrier voltage is
developed across transformer T2 in phase with the carrier voltage across T

1. Consequently, the output signal is in phase with the reference oscillator.

If the binary input is a logic 0 (negative voltage), diodes Dl and D2 are reverse biased and off,
while diodes D3 and D4 are forward biased and on (Figure 9-13c). As a result, the carrier voltage is
developed across transformer T2 180° out of phase with the carrier voltage across T 1.
FIGURE 9-13 (a) Balanced ring modulator; (b) logic 1 input; (c) logic 0 input

FIGURE 2-14 BPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
BANDWIDTH CONSIDERATIONS OF BPSK:

In a BPSK modulator. the carrier input signal is multiplied by the binary data.

If + 1 V is assigned to a logic 1 and -1 V is assigned to a logic 0, the input carrier (sin ct) is
multiplied by either a + or - 1 .

The output signal is either + 1 sin ct or -1 sin ct the first represents a signal that is in phase with
the reference oscillator, the latter a signal that is 180° out of phase with the reference
oscillator.Each time the input logic condition changes, the output phase changes.
Mathematically, the output of a BPSK modulator is proportional to
BPSK output = [sin (2 fat)] x [sin (2 fct)] (2.20)
where
fa = maximum fundamental frequency of binary input (hertz)
fc = reference carrier frequency (hertz)

Solving for the trig identity for the product of two sine functions,

0.5cos[2 (fc fa)t] 0.5cos[2 (fc + fa)t]

Thus, the minimum double-sided Nyquist bandwidth (B) is

fc + fa fc + fa
-f + f
-(fc + fa) or c a

2fa
and because fa = fb / 2, where fb = input bit rate,

where B is the minimum double-sided Nyquist bandwidth.

Figure 2-15 shows the output phase-versus-time relationship for a BPSK waveform. Logic 1 input
produces an analog output signal with a 0° phase angle, and a logic 0 input produces an analog
output signal with a 180° phase angle.

As the binary input shifts between a logic 1 and a logic 0 condition and vice versa, the phase of the
BPSK waveform shifts between 0° and 180°, respectively.
BPSK signaling element (ts) is equal to the time of one information bit (tb), which indicates that the
bit rate equals the baud.

FIGURE 2-15 Output phase-versus-time relationship for a BPSK modulator

Example:

For a BPSK modulator with a carrier frequency of 70 MHz and an input bit rate of 10 Mbps,
determine the maximum and minimum upper and lower side frequencies, draw the output spectrum,
de-termine the minimum Nyquist bandwidth, and calculate the baud..

Solution

Substituting into Equation 2-20 yields

output = [sin (2 fat)] x [sin (2 fct)]; fa = fb / 2 = 5 MHz

=[sin 2 (5MHz)t)] x [sin 2 (70MHz)t)]

=0.5cos[2 (70MHz 5MHz)t] 0.5cos[2 (70MHz + 5MHz)t]
lower side frequency upper side frequency

Minimum lower side frequency (LSF):

LSF=70MHz - 5MHz = 65MHz

Maximum upper side frequency (USF):

USF = 70 MHz + 5 MHz = 75 MHz

Therefore, the output spectrum for the worst-case binary input conditions is as follows: The
minimum Nyquist bandwidth (B) is

B = 75 MHz - 65 MHz = 10 MHz

and the baud = fb or 10 megabaud.

BPSK receiver:.
Figure 2-16 shows the block diagram of a BPSK receiver.
The input signal maybe+ sin ct or - sin ct .The coherent carrier recovery circuit detects and
regenerates a carrier signal that is both frequency and phase coherent with the original transmit
carrier.
The balanced modulator is a product detector; the output is the product d the two inputs (the BPSK
signal and the recovered carrier).
The low-pass filter (LPF) operates the recovered binary data from the complex demodulated signal.

FIGURE 2-16 Block diagram of a BPSK receiver

Mathematically, the demodulation process is as follows.
For a BPSK input signal of + sin ct (logic 1), the output of the balanced modulator is
output = (sin ct )(sin ct) = sin2 ct (2.21)
or sin2 ct = 0.5(1 cos 2 ct) = 0.5 - 0.5cos 2 ct

filtered out

leaving
output = + 0.5 V = logic 1
It can be seen that the output of the balanced modulator contains a positive voltage (+[1/2]V) and a
cosine wave at twice the carrier frequency (2 ct ).

The LPF has a cutoff frequency much lower than 2 ct, and, thus, blocks the second harmonic of
the carrier and passes only the positive constant component. A positive voltage represents a
demodulated logic 1.

For a BPSK input signal of -sin ct (logic 0), the output of the balanced modulator is

output = (-sin ct )(sin ct) = sin2 ct

or

sin2 ct = -0.5(1 cos 2 ct) = 0.5 + 0.5cos 2 ct

 filtered out
leaving
output = - 0.5 V = logic 0
The output of the balanced modulator contains a negative voltage (-[l/2]V) and a cosine wave at
twice the carrier frequency (2 ct).
Again, the LPF blocks the second harmonic of the carrier and passes only the negative constant
component. A negative voltage represents a demodulated logic 0.
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. The following figure represents the QPSK waveform for two bits
input, which shows the modulated result for different instances of binary inputs.

QPSK is a variation of BPSK, and it is also a DSB-SC (Double Sideband Suppressed Carrier)
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.

2-5-2-1 QPSK transmitter.

A block diagram of a QPSK modulator is shown in Figure 2-17Two bits (a dibit) are clocked into
the bit splitter. After both bits have been serially inputted, they are simultaneously parallel
outputted.

The I bit modulates a carrier that is in phase with the reference oscillator (hence the name "I" for "in
phase" channel), and theQ bit modulate, a carrier that is 90° out of phase.

For a logic 1 = + 1 V and a logic 0= - 1 V, two phases are possible at the output of the I balanced
modulator (+sin ct and - sin ct), and two phases are possible at the output of the Q balanced
modulator (+cos ct), and (-cos ct).

When the linear summer combines the two quadrature (90° out of phase) signals, there are four
possible resultant phasors given by these expressions: + sin ct + cos ct, + sin ct - cos ct, -sin
ct + cos ct, and -sin ct - cos ct.

Example:
For the QPSK modulator shown in Figure 2-17, construct the truthtable, phasor diagram, and
constellation diagram.

Solution

For a binary data input of Q = O and I= 0, the two inputs to the Ibalanced modulator are -1 and sin
ct, and the two inputs to the Q balanced modulator are -1 and cos ct.

Consequently, the outputs are

I balanced modulator =(-1)(sin ct) = -1 sin ct

Q balanced modulator =(-1)(cos ct) = -1 cos ct and the output of the linear summer is
-1 cos ct - 1 sin ct = 1.414 sin( ct - 135°)

For the remaining dibit codes (01, 10, and 11), the procedure is the same. The results are shown in
Figure 2-18a.

FIGURE 2-18 QPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
In Figures 2-18b and c, it can be seen that with QPSK each of the four possible output phasors has
exactly the same amplitude. Therefore, the binary information must be encoded entirely in the
phase of the output signal
Figure 2-18b, it can be seen that the angular separation between any two adjacent phasors in QPSK
is 90°.Therefore, a QPSK signal can undergo almost a+45° or -45° shift in phase during
transmission and still retain the correct encoded information when demodulated at the receiver.

Figure 2-19 shows the output phase-versus-time relationship for a QPSK modulator.

FIGURE 2-19 Output phase-versus-time relationship for a PSK modulator

2-5-2-2 Bandwidth considerations of QPSK

With QPSK, because the input data are divided into two channels, the bit rate in either the I or the Q
channel is equal to one-half of the input data rate (fb/2) (one-half of fb/2 = fb/4).

QPSK RECEIVER:

The block diagram of a QPSK receiver is shown in Figure 2-21

The power splitter directs the input QPSK signal to the I and Q product detectors and the carrier
recovery circuit. The carrier recovery circuit reproduces the original transmit carrier oscillator
signal. The recovered carrier must be frequency and phase coherent with the transmit reference
carrier. The QPSK signal is demodulated in the I and Q product detectors, which generate the
original I and Q data bits. The outputs of the product detectors are fed to the bit combining circuit,
where they are converted from parallel I and Q data channels to a single binary output data stream.
The incoming QPSK signal may be any one of the four possible output phases shown in Figure 2-
18. To illustrate the demodulation process, let the incoming QPSK signal be -sin ct + cos ct.
Mathematically, the demodulation process is as follows.
 FIGURE 2-21 QPSK receiver

The receive QPSK signal (-sin ct + cos ct) is one of the inputs to the I product detector. The
other input is the recovered carrier (sin ct). The output of the I product detector is

Again, the receive QPSK signal (-sin ct + cos ct) is one of the inputs to the Q product detector.
The other input is the recovered carrier shifted 90° in phase (cos ct). The output of the Q product
detector is
The demodulated I and Q bits (0 and 1, respectively) correspond to the constellation diagram and
truth table for the QPSK modulator shown in Figure 2-18.

DIFFERENTIAL PHASE SHIFT KEYING (DPSK):

In DPSK (Differential Phase Shift Keying) 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

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 is continued as it was. If the data is 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.

The word binary represents two-bits. M simply represents a digit that corresponds to the number of
conditions, levels, or combinations possible for a given number of binary variables.
This is the type of digital modulation technique used for data transmission in which instead of one-
bit, two or more bits are transmitted at a time. As a single signal is used for multiple bit
transmission, the channel bandwidth is reduced.
DBPSK TRANSMITTER.:

Figure 2-37a shows a simplified block diagram of a differential binary phase-shift keying
(DBPSK) transmitter. An incoming information bit is XNORed with the preceding bit prior to
entering the BPSK modulator (balanced modulator).

For the first data bit, there is no preceding bit with which to compare it. Therefore, an initial
reference bit is assumed. Figure 2-37b shows the relationship between the input data, the XNOR
output data, and the phase at the output of the balanced modulator. If the initial reference bit is
assumed a logic 1, the output from the XNOR circuit is simply the complement of that shown.

In Figure 2-37b, the first data bit is XNORed with the reference bit. If they are the same, the XNOR
output is a logic 1; if they are different, the XNOR output is a logic 0. The balanced modulator
operates the same as a conventional BPSK modulator; a logic I produces +sin ct at the output, and
A logic 0 produces sin ct at the output.
 FIGURE 2-37 DBPSK modulator (a) block diagram (b) timing diagram

BPSK RECEIVER:
Figure 9-38 shows the block diagram and timing sequence for a DBPSK receiver. The received
signal is delayed by one bit time, then compared with the next signaling element in the balanced
modulator. If they are the same. J logic 1(+ voltage) is generated. If they are different, a logic 0 (-
voltage) is generated. [f the reference phase is incorrectly assumed, only the first demodulated bit is
in error. Differential encoding can be implemented with higher-than-binary digital modulation
schemes, although the differential algorithms are much more complicated than for DBPS K.

The primary advantage of DBPSK is the simplicity with which it can be implemented. With
DBPSK, no carrier recovery circuit is needed. A disadvantage of DBPSK is, that it requires
between 1 dB and 3 dB more signal-to-noise ratio to achieve the same bit error rate as that of
absolute PSK.
 FIGURE 2-38 DBPSK demodulator: (a) block diagram; (b) timing sequence

COHERENT RECEPTION OF FSK:

The coherent demodulator for the coherent FSK signal falls in the general form of coherent
demodulators described in Appendix B. The demodulator can be implemented with two correlators
as shown in Figure 3.5, where the two reference signals are cos(27r f t) and cos(27r fit). They must
be synchronized with the received signal. The receiver is optimum in the sense that it minimizes the
error probability for equally likely binary signals. Even though the receiver is rigorously derived in
Appendix B, some heuristic explanation here may help understand its operation. When s 1 (t) is
transmitted, the upper correlator yields a signal 1 with a positive signal component and a noise
component. However, the lower correlator output 12, due to the signals' orthogonality, has only a
noise component. Thus the output of the summer is most likely above zero, and the threshold
detector will most likely produce a 1. When s2(t) is transmitted, opposite things happen to the two
correlators and the threshold detector will most likely produce a 0. However, due to the noise nature
that its values range from -00 to m, occasionally the noise amplitude might overpower the signal
amplitude, and then detection errors will happen. An alternative to Figure 3.5 is to use just one
correlator with the reference signal cos (27r f t) - cos(2s f2t) (Figure 3.6). The correlator in Figure
3.6 can be replaced by a matched filter that matches cos(27r fit) - cos(27r f2t) (Figure 3.7). All
implementations are equivalent in terms of error performance (see Appendix B). Assuming an
AWGN channel, the received signal is

where n(t) is the additive white Gaussian noise with zero mean and a two-sided power spectral
density A',/2. From (B.33) the bit error probability for any equally likely binary signals is

where No/2 is the two-sided power spectral density of the additive white Gaussian noise. For
Sunde's FSK signals El = Ez = Eb, pI2 = 0 (orthogonal). thus the error probability is

where Eb = A2T/2 is the average bit energy of the FSK signal. The above Pb is plotted in Figure 3.8
where Pb of noncoherently demodulated FSK, whose expression will be given shortly, is also
plotted for comparison.
 Figure: Pb of coherently and non-coherently demodulated FSK signal.

NONCOHERENT DEMODULATION AND ERROR PERFORMANCE:

Coherently FSK signals can be noncoherently demodulated to avoid the carrier recovery.
Noncoherently generated FSK can only be noncoherently demodulated. We refer to both cases as
noncoherent FSK. In both cases the demodulation problem becomes a problem of detecting signals
with unknown phases. In Appendix B we have shown that the optimum receiver is a quadrature
receiver. It can be implemented using correlators or equivalently, matched filters. Here we assume
that the binary noncoherent FSK signals are equally likely and with equal energies. Under these
assumptions, the demodulator using correlators is shown in Figure 3.9. Again, like in the coherent
case, the optimality of the receiver has been rigorously proved (Appendix B). However, we can
easily understand its operation by some heuristic argument as follows. The received signal
(ignoring noise for the moment) with an unknown phase can be written as
The signal consists of an in phase component A cos 8 cos 27r f t and a quadrature component A sin
8 sin 2x f,t sin 0. Thus the signal is partially correlated with cos 2s fit and partiah'y correlated with
sin 27r fit. Therefore we use two correlators to collect the signal energy in these two parts. The
outputs of the in phase and quadrature correlators will be cos 19 and sin 8, respectively. Depending
on the value of the unknown phase 8, these two outputs could be anything in (- 5, y). Fortunately
the squared sum of these two signals is not dependent on the unknown phase. That is

This quantity is actually the mean value of the statistics I? when signal si (t) is transmitted and noise
is taken into consideration. When si (t) is not transmitted the mean value of 1: is 0. The comparator
decides which signal is sent by checking these I?. The matched filter equivalence to Figure 3.9 is
shown in Figure 3.10 which has the same error performance. For implementation simplicity we can
replace the matched filters by bandpass filters centered at f and fi, respectively (Figure 3.1 1).
However, if the bandpass filters are not matched to the FSK signals, degradation to

various extents will result. The bit error probability can be derived using the correlator demodulator
(Appendix B). Here we further assume that the FSK signals are orthogonal, then from Appendix B
the error probability is
 PART-2

DATATRANSMISSION

BASE BAND SIGNAL RECEIVER:

Consider that a binary encoded signal consists of a time sequence of voltage levels +V or V.
if there is a guard interval between the bits, the signal forms a sequence of positive and
negative pulses. in either case there is no particular interest in preserving the waveform of the
signal after reception .we are interested only in knowing within each bit interval whether the
transmitted voltage was +V or V. With noise present, the receives signal and noise together
will yield sample values generally different from ±V. In this case, what deduction shall we
make from the sample value concerning the transmitted bit?
PROBABILITY OF ERROR
Since the function of a receiver of a data transmission is to ditinguish the bit 1 from the bit 0
in the presence of noise, a most important charcteristic is the probability that an error will be
made in such a determination.we now calculate this error probabilty P e for the integrate and
dump receiver of fig 11.1-2
The probability of error pe, as given in eq.(11.2-3),is plotted in fig.11.2-2.note that pe decreases
rapidly as Es. increases. The maximum value of pe is ½.thus ,even if the signal is entirely lost in
the noise so that any determination of the receiver is a sheer guess, the receiver cannot bi wrong
more than half the time on the average.

THE OPTIMUM FILTER:

In the receiver system of Fig 11.1-2, the signal was passed through a filter(integrator),so that at the
sampling time the signal voltage might be emphasized in comparison with the noise voltage. We are
naturally led to risk whether the integrator is the optimum filter for the purpose of minimizing the
probability of error. We shall find that the received signal contemplated in system of fig 11.1-2 the
integrator is indeed the optimum filter. However, before returning specifically to the integrator
receiver.

We assume that the received signal is a binary waveform. One binary digit is represented by
a signal waveformS1 (t) which persists for time T, while the4 other bit is represented by the
waveform S2(t) which also lasts for an interval T. For example, in the transmission at baseband, as
shown in fig 11.1-2 S1(t)=+V; for other modulation systems, different waveforms are transmitted.
for example for PSK signaling , S1(t)=Acosw0t and S2(t)=-Acosw0t;while for FSK,
S1(t)=Acos(w t.
Hence probability of error is
 In general, the impulsive response of the matched filter consists of p(t) rotated about t=0 and
then delayed long enough(i.e., a time T) to make the filter realizable. We may note in passing, that
any additional delay that a filter might introduce would in no way interfere with the performance of
the filter ,for both signal and noise would be delayed by the same amount, and at the sampling time
(which would need similarity to be delayed)the ratio of signal to noise would remain unaltered.
COHERENT RECEPTION :CORRELATION:

(11.6-1)
 (11.6-2)
Where s1(t) is either s1(t) or s2 integrator

outputs.

Fig:11.6-1Coherent system of signal reception

If h(t) is the impulsive response of the matched filter ,then the output of the matched filter v0(t) can
be found using the convolution integral. we have

(11.6-3)

The limits on the integral have been charged to 0 and T since we are interested in the filter response
to a bit which extends only over that interval. Using Eq.(11.4-4) which gives h(t) for the matched
filter, we have

(11.6-4)
(11.6-5)

sub 11.6-5 in 11.6-3

(11.6-6)

(11.6-7)

(11.6-8)

Thus s0(T) and n0(T), as calculated from eqs.(11.6-1) and (11.6-2) for the correlation receiver, and
as calculated from eqs.(11.6-7) and (11.6-8) for the matched filter receiver, are identical .hence the
performances of the two systems are identical. The matched filter and the correlator are not simply
two distinct, independent techniques which happens to yield the same result. In fact they are two
techniques of synthesizing the optimum filter h(t)
ERROR PROBABILITY OF BINARY PSK:

To realize a rule for making a decision in favor of symbol 1 or symbol 0,we partition the signal
space into two regions:
 Unit-V
Spread spectrum modulation

MULTIPLEXING
channels ki
• Multiplexing in 4 dimensions
o space (si) k1 k2 k3 k4 k5 k6

o time (t)
c
o frequency (f)
t c
o code (c) t
s1 f
• Goal: multiple use s2 f
of a shared medium
c
t
• Important: guard spaces needed!
s3 f

1
 FREQUENCY MULTIPLEX

 Separation of spectrum into smaller frequency bands

 Channel gets band of the spectrum for the whole time
 Advantages:
• no dynamic coordination needed k3 k4 k5 k6
• works also for analog signals
c f
 Disadvantages:
• waste of bandwidth if traffic distributed unevenly
• inflexible
• guard spaces

2
TIME MULTIPLEX
Channel gets the whole spectrum for a certain
amount of time
Advantages:
• only one carrier in the medium at any time
• throughput high even for many users
Disadvantages:
• Precise synchronization necessary

k1 k2 k3 k4 k5 k6
c

3
 TIME & FREQUENCY MULTIPLEX
 A channel gets a certain frequency band for a certain
amount of time (e.g. GSM)
 Advantages:
• better protection against tapping
• protection against frequency selective interference
• higher data rates compared to code multiplex
 Precise coordination required
k1 k2 k3 k4 k5 k6
c
f

CODE MULTIPLEX
k1 k2 k3 k4 k5 k6

 Each channel has unique code

c
 All channels use same spectrum at same time
 Advantages:
• bandwidth efficient
• no coordination and synchronization
• good protection against interference
 Disadvantages: f
• lower user data rates
• more complex signal regeneration
 Implemented using spread spectrum technology
t

4
 Spread Spectrum
Problem of radio transmission: frequency dependent fading can
wipe out narrow band signals for duration of the interference

Solution: spread the narrow band signal into a broad band signal
using a special code

The spread spectrum techniques was developed initially for

military and intelligence requirements.

The essential idea is to spread the information signal over a wider

bandwidth to make jamming and interception more difficult.

Spread Spectrum (Contd.)

• Spread spectrum is a communication technique that spreads a

narrowband communication signal over a wide range of frequencies for
transmission then de-spreads it into the original data bandwidth at the
receiver.
• Spread spectrum is characterized by:
- wide bandwidth and
- low power
• Jamming and interference have less effect on Spread spectrum
because it is:
- Resembles noise
- Hard to detect
- Hard to intercept

5
 Narrowband vs Spread Spectrum

Power
Narrowband
(High Peak Power)

Spread Spectrum
(Low Peak Power)

Frequency

Signal Spreading

6
SPREADING AND FREQUENCY SELECTIVE FADING
channel
quality

1 2 5 6
narrowband
3
4
channels
Narrowband guard space frequency
signal

channel
quality
2
2
2
2
12 spread spectrum
channels
spread frequency
spectrum

SPREAD SPECTRUM- MAIN OPERATION

b(t) m(t)
x

c(t)

mt   bt  ct 

M  f   B f *C f 

B(f)
M(f)

7
 Why Spread Spectrum..?
 Advantages:
• Spread spectrum signals are distributed over a wide range of frequencies
and then collected back at the receiver
• These wideband signals are noise-like and hence difficult to detect or
interfere with
• Initially adopted in military applications, for its resistance to jamming and
difficulty of interception
• More recently, adopted in commercial wireless communications
• Has the ability to eliminate the effect of multipath interference
• Can share the same frequency band with other users
• Privacy due to the pseudo random code sequence (code division
multiplexing)

 Disadvantages:
• Bandwidth inefficient
• Implementation is somewhat more complex.

Spread Spectrum Technique

• To apply an SS technique, simply inject the corresponding SS code somewhere

in the transmitting chain before the antenna. The effect is to diffuse the
information in a larger bandwidth. Conversely, you can remove the SS code, at
a point in the receive chain before data retrieval. The effect of a de spreading
operation is to reconstitute the information in its original bandwidth. Obviously,
the same code must be known in advance at both ends of the transmission
channel.

8
 Narrow Band vs Spread Spectrum

• Narrow Band
- Uses only enough frequency spectrum to carry the signal
- High peak power
- Easily jammed
• Spread Spectrum
- The bandwidth is much wider than required to send to the signal.
- Low peak power
-Hard to detect
-Hard to intercept
- Difficult to jam

Spread Spectrum Use

• In the 1980s FCC implemented a set of rules making Spread Spectrum
available to the public.
- Cordless Telephones
- Global Positioning Systems (GPS)
- Cell Phones
- Personal Communication Systems
- Wireless video cameras
• Local Area Networks
- Wireless Local Area Networks (WLAN)
- Wireless Personal Area Network (WPAN)
- Wireless Metropolitan Area Network (WMAN)
- Wireless Wide Area Network (WWAN)

9
 FCC Specifications
• The Code of Federal Regulations (CFR) Part 15 originally only described two
spread spectrum techniques to be used in the licensed free Industrial, Scientific,
Medical (ISM) band, 2.4 GHz, thus 802.11 and 802.11b.
- Frequency Hopping Spread Spectrum (FHSS) and
- Direct Sequence spread Spectrum (DSSS)

• Orthogonal Frequency Division Multiplexing (OFDM) was not covered by the

CFR and would have required licensing.
- 802.11a, employing OFDM, was created to work in the 5GHz Unlicensed
National Information Infrastructure (UNII)

• In May, 2001 CFR, Part 15 was modified to allow alternative "digital modulation
techniques".
- This resulted in 802.11g which employs OFDM in the 2.4 GHz range

• Direct Sequence (DS) - A carrier is modulated by a digital

code sequence in which bit rate is much higher than the
information signal bandwidth.

• Frequency Hopping (FH) - A carrier frequency is shifted in

discrete increments in a pattern dictated by a code sequence.

10
 What is pseudorandom number sequences?
is a sequence of numbers that has been computed by
some defined arithmetic process but is effectively a random
number sequence for the purpose for which it is required.

• Input is fed into a channel encoder

• Produce an analog signal with a relatively narrow bandwidth around
some center frequency.
• Further modulated using a sequence of digits known as a spreading code
or spreading sequence.
• The spreading code is generated by a pseudo noise, or
pseudorandom number generator.
• The effect of this modulation is to increase significantly the
bandwidth (spread the spectrum) of the signal to be transmitted.
• At the receiver, the same digit sequence is used to demodulate the spread
spectrum signal.
• The signal is fed into a channel decoder to recover the data.
• Spread Spectrum signals use fast codes that run many times the
information bandwidth or data rate.

• These special "Spreading" codes are called "Pseudo Random" or

"Pseudo Noise" codes. They are called "Pseudo" because they are not real
Gaussian noise.

11
 What can we gain from this apparent waste of spectrum?
 Gain immunity from various kinds of noise and multipath
distortion

 Can be used for hiding and encrypting signals. Only a

recipient who knows the spreading code can recover the
encoded information.

 Several users can independently use the same higher

bandwidth with very little interference (CDMA).

GENERATING THE SPREADING (PSEUDO-NOISE) SEQUENCE

• The pseudo-noise (PN) sequence is a periodic binary sequence with a

noise like waveform that is generated by means of a feedback shift
register.

• The feedback shift register consists of m-stage shift registers and a

logic circuit that perform modulo-2 (X-OR) arithmetic.

• A sequence with period 2m-1 is called Maximal-Length sequence

12
PN SEQUENCE: EXAMPLE

s1 s2 s3
1 0 0
1 1 0
1 1 1
0 1 1
1 0 1
0 1 0
0 0 1
1 0 0

Spreading code  0 0 1 1 1 0 1 0 . . .

Here N=2m – 1=7, length of the sequence

PROPERTIES OF MAXIMAL-LENGTH SEQUENCES

• In each period of a maximal-length sequence, the number

of 1’s and the number of 0’s in the sequence always differ
by 1.
• The autocorrelation function of a maximal-length sequence
is periodic and binary valued.
Tb
Rc    ct c t  dt
1
Tb  2
T
 b
2

 N 1 
 1    Tc
R c    NT c
 1 for the rest of the period
 N

13
 Chip rate, Rc is the rate at which the
no. of bits of the PN sequence
occur.

The duration of each bit is

TC= 1/Rc

Therefore, the period of the

waveform is Tb = NTc

W1 and W2N

14
Sequence Generation

Chip Sequences

Encoding Rules

15
CDMA Multiplexer

CDMA Demultiplexer

16
 DIRECT SEQUENCE SPREAD SPECTRUM (DSSS)

Direct Sequence (DS) - A carrier is modulated by a digital code

sequence in which bit rate is much higher than the information signal
bandwidth.

• Most widely recognized technology for spread spectrum.

• This method generates a redundant bit pattern for each bit to be

transmitted. This bit pattern is called a chip.

• The longer the chip, the greater the probability that the Original
data can be recovered, and the more bandwidth required.

In a spread spectrum system, the process gain (or ‘processing gain') is

the ratio of the spread bandwidth to the unspread bandwidth. It is
usually expressed in decibels (dB).

• The amount of spreading is dependent upon the ratio of chips per bit
of information (which is the processing gain Gp for DSSS)

• A direct sequence modulator is then used to carrier modulate the

carrier using binary phase shift keying (BPSK)

• At the receiver, the information is recovered by multiplying the signal

with a locally generated replica of the code sequence.

• Each bit represented by multiple bits using spreading code

• Spreading code spreads signal across wider frequency band

• In proportion to number of bits used
• 10 bit spreading code spreads signal across 10 times bandwidth
of 1 bit code

17
 Processing Gain

The process gain (or ‘processing gain') is the ratio of the spread
bandwidth to the unspread bandwidth. It is usually expressed in
decibels (dB).

DSSS USING BPSK

18
19
 RANGING USING DSSS

Frequency Hopping Spread Spectrum (FHSS)

• When using FHSS, the frequency spectrum is divided into channels.
Data packets are split up and transmitted on these channels in a
random pattern known only to the transmitter and receiver.
At the transmitter, the original signal is broadcasted
over a series of radio frequencies, hopping from
different frequencies in a fixed pattern.

The receiver should use the same hopping pattern

simultaneously with the transmitter in order to
receive the data correctly.

The spreading code specifies the sequence of

channels and the receiver should use the same
code to tune into the sequence of channels that are
used by the sender.

20
• The information signal is transmitted on different frequencies
• Time is divided in slots
• In each slot the frequency is changed
• The change of the frequency is referred to as slow if more than one
bit is transmitted on one frequency, and as fast if one bit is
transmitted over multiple frequencies
• The frequencies are chosen based on the spreading sequences

 Each channel used for fixed interval

• Eg: 300 ms in IEEE 802.11
• Sequence dictated by spreading cod

FREQUENCY HOPPING SPREAD SPECTRUM (FHSS)

21
FREQUENCY SELECTION IN FHSS (EXAMPLE 1)

Normally ‘K’ successive bits of input data sequence represents 2 k = M

symbols. Those distinct M symbols are transmitted with the help of M-
ary FSK modulation system. When spread spectrum modulation is to
be used, then the M-ary FSK signal is further modulated to generate
wideband signal.

FHSS CYCLES

22
Two versions
Fast Hopping: several frequencies per user bit (FFH) --- symbol rate
is lesser than the hop rate i.e., hop rate is faster.
Slow Hopping: several user bits per frequency (SFH) --- symbol rate
is higher than hop rate i.e., hop rate is slower.
Tb

user data
0 1 0 1 1 t
f
Td
f3 slow
f2 hopping
f1 (3 bits/hop)

Td t
f
f3 fast
f2 hopping
f1 (3 hops/bit)

(EXAMPLE 2)

• FHSS uses the 2.402 – 2.480 GHz frequency range in the ISM
band.
• It splits the band into 79 non-overlapping channels with each
channel 1 MHz wide.
Transmission Frequency (GH z)

2.479
1 MHz Channels
Divided into 79

2.401

200 400 600 800 1000 1200 1400 1600

Elapsed Time in Milliseconds (ms)

Channel 1 Channel 2 Channel 78

23
BANDWIDTH SHARING (EXAMPLE 3)

(EXAMPLE 4)

• A number of channels are

allocated for the FH signal.

• Typically, there are 2k

carrier frequencies forming
2k channels.

• The width of each channel

corresponds to the
bandwidth of the input
signal.

24
Transmitter of FH/MFSK

FSK FH/MFSK
signal signal
M-ary FSK
Mixer
Modulator
Binary data
Sequence

Frequency hops

Frequency
Synthesizer
‘t’ successive bits of
PN sequence
1 2 t
generator
PN
Sequence
Generator

25
Receiver of FH/MFSK

Received FSK
FH/MFSK signal signal Non coherent
Mixer M-ary FSK
detector Binary
Sequence output

Frequency hops

Frequency
Synthesizer

PN
Sequence
Generator

Application of FHSS – Multipath Suppression

Multipath Interference Problem
In mobile communication, the signal reach to the receiver from different
paths.
There is one direct path and many indirect paths due to reflections from
nearby objects.
The signal due to indirect paths interfere with the required signal in
amplitude as well as phase. It is called multipath fading.
How FHSS overcomes multipath effect?
The carrier frequency of the transmitted signal hops faster than the
differential time delay between the direct signal and reflected signals.
Therefore the reflected signal energy will fall in different frequency slots.
This signal energy will be treated as interference by the matched filter of the
receiver. It is then filtered out and only signal from direct path is available at
the output.
The hopping rate must be fast enough to eliminate interference due to small
time delays between direct and reflected paths.

26
 Synchronization in Spread Spectrum Systems

Spread spectrum systems are essentially synchronous. The PN sequences

generated at the receiver and the transmitter must be same and locked to
each other so that the transmitted signal can be extracted.

The synchronization of the spread spectrum systems can be considered in

two parts:
• Acquisition
• Tracking

The acquisition means initial synchronization of the spread spectrum signal.

The tracking starts after acquisition is complete. The tracking maintains the
PN generator at the receiver in synchronism with the transmitter.

Acquisition of DS signal using Serial search

27
The received signal is correlated with the generated PN sequence. This
cross correlation is performed over the time interval of NTC.

The output of the Correlator is compared with a threshold. If it exceeds

the threshold, then the required signal is obtained. If the threshold is
not exceeded, then the PN generator output is advanced by half chip
duration ( 12 T ) and the correlation is performed.
C

The output of the Correlator is again compared with the threshold and
the procedure is repeated.

Acquisition of FH signal using Serial search

28
The VCO consists of frequency synthesizer, PN generator and clock
generator. The received signal is correlated with the output of VCO.

The tuned filter passes only the intermediate frequency f 0.

The envelop detector generates the output which is compared with the
threshold voltage.

When the input frequency and frequency of VCO are same, then output
of threshold detector is high and the clock generator starts running
continuously. Then the signal is said to have acquired and tracking
starts.

29