Notes - PC Eee 701 - Unit 05

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EEE-7TH SEM

PAPER NAME: - ANALOG & DIGITAL COMMUNICATION


PAPER CODE: - PC-EEE-701

UNIT-5
MODULATION OF DIGITAL SIGNAL
Introduction

⚫ There are basically two types of transmission of Digital Signals

⚫ Baseband data transmission

⚫ The digital data is transmitted over the channel directly. There is no carrier or
any modulation. Suitable for transmission over short distances.

⚫ Pass band data transmission

⚫ The digital data modulates high frequency sinusoidal carrier. Suitable for
transmission over longer distances.

Types of Pass band Modulation

⚫ The digital data can modulate phase, frequency or amplitude of carrier. This gives rise
to three basic techniques:

⚫ Phase Shift Keying (PSK): The digital data modulates the phase of the carrier.

⚫ Frequency Shift Keying(FSK): The digital data modulates the frequency of the
carrier.

⚫ Amplitude Shift Keying (ASK): The digital modulates the amplitude of the carrier.

Digital Modulation Techniques


Types of Reception for Pass band Transmission

⚫ Two Types of methods for detection of pass band signals

⚫ Coherent (Synchronous) Detection: The local carrier generated at the receiver is


phase locked with the carrier at the transmitter. Hence called Synchronous Detection.

⚫ Non Coherent (Envelope) Detection: The receiver carrier need not be phase locked
with the transmitter carrier. It is called Envelope detection. It is simple but it has
higher probability of error.

Requirements of Pass band Transmission Scheme

⚫ Maximum Data transmission rate

⚫ Minimum Probability of symbol error

⚫ Minimum Transmitted power

⚫ Minimum Channel Bandwidth

⚫ Maximum resistance to interfering signals

⚫ Minimum circuit complexity

Advantages of Pass band Transmission over Baseband transmission

⚫ Long Distance Transmission

⚫ Analog Channels, can be used for Transmission

⚫ Multiplexing techniques can be used for BW conservation.

⚫ Problems such as ISI and crosstalk are absent

⚫ Pass band transmission can take place over wireless channels also.

Introduction

⚫ In digital modulation, an analog carrier signal is modulated by a discrete signal.

⚫ Digital modulation can be considered as digital-to-analog and the corresponding


demodulation is considered as analog-to-digital conversion.

⚫ In Digital communications, the modulating wave consists of binary data and the
carrier is sinusoidal wave.
Amplitude Shift Keying (On-Off Keying)

⚫ In this there is only one unit energy carrier and it is switched on or off depending
upon the Binary sequence.

ASK waveform may be represented as

⚫ Signal s(t) contains some complete cycles of carrier frequency (fc).

⚫ Hence the ASK waveform looks like an On-Off of the signal. Therefore it is also
known as the On-Off Keying(OOK)

Generation of ASK Signal

⚫ ASK signal may be generated by simply applying the incoming binary data and the
sinusoidal carrier to the 2 inputs of a product modulator.

⚫ The resulting output will be the ASK waveform.

⚫ Modulation causes the shift of the baseband signal spectrum.


Power Spectral Density (PSD) of Unipolar NRZ:

⚫ The PSD of Unipolar NRZ is given by equ

⚫ PSD of Unipolar NRZ is as shown below

Power Spectral Density (PSD) of ASK

⚫ The PSD of ASK signal is same as that of a baseband on-off signal but shifted in the
frequency domain by ± fc

⚫ It may be noted that 2 impulses occur at ± fc

⚫ The spectrum of ASK shows that it has infinite bandwidth.

⚫ Bandwidth is defined as the BW of an ideal band pass filter centred at fc whose output
contains about 95% of the total average power content of the ASK signal.

⚫ According to this criterion the Bandwidth of ASK signal is approximately 3/T b .


Demodulation of ASK

Coherent Detection of ASK (Integrate and Dump):

⚫ The input to the receiver consists of an ASK signal that is corrupted by AWGN.

⚫ The receiver integrates the product of the signal plus noise & a copy of the noise free
signal over one signal interval.

⚫ Assume that the local signal

is carefully synchronized with the frequency & phase of the carrier received.

⚫ Output of integrator is compared against a set threshold and at the end of each
signalling interval the receiver makes the decision about which of the 2 signals s1(t)
or s2(t) was present at its input during the signalling interval.

⚫ Errors might occur in the demodulation process because of noise.

⚫ Assume

⚫ The signalling components of the receiver output at the end of the signalling interval
are

⚫ The optimum threshold setting in the receiver is


⚫ The receiver decodes the kth transmitted bit as 1 if the output at the kth signalling
interval is greater than Vth , as a ‘0’ otherwise.

Non Coherent ASK detection

This scheme involves detection in the form of ‘rectifier’ & ‘low pass filter’.

⚫ Input to the receiver is

Where

⚫ ni(t) represents represents AWGN with zero mean at the receiver input.

Now if the BPF is assumed to have BW of 2/T b centred at fc , then it passes the signal
component without much distortion.

⚫ The filter output will be

⚫ Where Ak=A when the kth transmitted bit bk=1, and Ak=0, when bk=0

⚫ The above equ can be written in envelope and phase form as

Where nc(t) and ns(t) are the quadrature components of narrow band noise
Advantages and Disadvantages of ASK

⚫ Advantages

⚫ Simple to design, easy to generate and detect.

⚫ Requires low Bandwidth

⚫ Requires less energy to transmit the binary data.

⚫ Disadvantages

⚫ Susceptible to sudden amplitude variations due to noise and interference.

Applications of ASK

⚫ Mostly used for very low-speed data rate (upto 1200bps) requirements on voice grade
lines in telemetry applications.

⚫ Used to transmit digital data over optical fibre for LED –based optical transmitters.

⚫ Wireless infrared transmissions using a directed beam or diffuse light in wireless


LANs applications.

Frequency Shift Keying

⚫ In Binary FSK, the frequency of the carrier is shifted according to the binary symbol.
Phase unaffected.

⚫ That is there are 2 different frequency signals according to binary symbols.

⚫ Let there be a frequency shift by Ω.

⚫ If b(t)=1, then

b(t)=0, then

⚫ Hence there is increase or decrease in frequency by Ω.

Conversion table for BFSK representation


⚫ FSK equ can be written as

⚫ Hence if symbol ‘1’ is to be transmitted then the carrier frequency will be

⚫ If the symbol ‘0’ is to be transmitted then the carrier frequency will be

⚫ Thus

Generation of BFSK

⚫ PH(t) is same as b(t) and PL(t) is inverted version of b(t)

⚫ PH(t) and PL(t) are Unipolar signals.

⚫ The level shifter converts ‘+1’ to √𝑃𝑠𝑇𝑏 and the zero level is unaffected.

⚫ Further there are product modulators after the level shifters.


⚫ The two carrier signals φ1(t) or φ2(t) are used which are orthogonal to each other. fH-
fL=2fb

⚫ The adder then adds the 2 signals coming from the multipliers, but outputs from the
multipliers are not possible at the same time. This is because P H(t) and PL(t) are
complementary to each other.

PSD of BFSK Signal

⚫ BFSK signal s(t) can be written as

⚫ Let us convert those coefficients in bipolar form as follows

⚫ Here 𝑃′ (𝑡) and 𝑃′ (𝑡) will be bipolar, alternating between +1 and -1, and
𝐻 𝐿
complementary.

⚫ Now s(t) can be written as

⚫ The below equation is used to find the PSD of BFSK Signal


PSD and BW of BFSK Signal

Coherent Detection of BFSK Signal

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.

Non-coherent Detection of BFSK Signal

The FSK input signal is simultaneously applied to the inputs of both band pass 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 pass band, and the comparator responds to the larger of the two powers.
This type of FSK detection is referred to as non coherent detection.
Detection of BFSK signal using PLL
Advantages and Disadvantages of FSK

⚫ Advantages

⚫ It is less susceptible to errors than ASK.

⚫ Better noise immunity than ASK.

⚫ Peak frequency offset is constant and always at its maximum.

⚫ The highest fundamental frequency is equal to half the information bit rate.

⚫ Relatively easy to implement.

⚫ Disadvantages

⚫ Not efficient in terms of transmission bandwidth requirement

⚫ It has poorer error performance than PSK or QAM.

Applications of FSK

⚫ Used in low-speed modems (up to 1200bps) over analog voice-band telephone lines.

⚫ Finds applications in pager systems, HF radio tele-type transmission systems, and


LANs using coaxial cables.

Binary Phase Shift Keying

⚫ Principle of BPSK

⚫ In BPSK the binary symbol ‘1’ and ‘0’ modulate the phase of the carrier.

Let the carrier be

‘A’ represents peak of the sinusoidal carrier

When the symbol is changed, then phase of the carrier is changed by 180o

Consider, for symbol ‘1’


⚫ For symbol ‘0’

⚫ Therefore

⚫ Which implies

Where b(t)=+1; for symbol ‘1’

b(t)=-1; for symbol ‘0’

Generation of BPSK

Coherent Reception of BPSK Signal

Operation of the receiver

⚫ Phase shift in the received signal:


⚫ The signal undergoes the phase change depending upon the time delay from
transmitter to receiver. Let the phase shift be

⚫ Square Law device:

⚫ From the received signal carrier is separated. Since it is coherent detection.

⚫ The output of square law device

⚫ Band Pass Filter:

⚫ The signal is passed through band pass filter with centre frequency 2fc.

⚫ BPF removes DC level and its output is

⚫ Frequency Divider:

⚫ The signal is passed through a frequency divider by 2.

⚫ Therefore at the output of the frequency divider we get the carrier signal
whose frequency is fc i.e.,

⚫ Synchronous Demodulator:

⚫ The synchronous demodulator multiplies the input signal & the recovered
carrier.

⚫ Therefore at the output of multiplier we get

⚫ Bit synchronizer and integrator:

⚫ The above signal is applied to the bit synchronizer & integrator.


⚫ The integrator integrates the signal over one bit period. The bit synchronizer
takes care of starting and end times of a bit.

⚫ At the end of the bit duration the bit synchronizer closes switch s2 temporarily
connecting the output of integrator to the decision device.
⚫ Synchronizer then opens s2 and closes s1 temporarily to reset the integrator.
⚫ Output of integrator: In the kth bit interval the output can be written as

The signal is then given to the decision device, which decides whether transmitted symbol
was zero or one.
PSD of BPSK

⚫ PSD of polar NRZ baseband signal b(t)=+1 or -1

Power spectral density of BPSK signal

Inter channel interference and ISI

⚫ Inter channel interference avoided by filtering.

⚫ Because of filtering phase distortion takes place resulting in ISI.

⚫ ISI can be reduced to some extent by using equalizers at the receiver.

⚫ Equalizers have reverse effect to the filters adverse effects.

Differential Phase Shift Keying

⚫ DPSK is an alternative form of digital modulation where the binary input information
is contained in the difference between two successive signalling elements rather than
the absolute phase.

⚫ With DPSK, it is not necessary to recover a phase coherent carrier.

⚫ Instead a received signalling element is delayed by one signalling element time slot
and then compared with the next received signalling element.
⚫ The difference in phase of two signalling elements determines the logic condition of
the data.

DPSK Transmitter

The figure (a) below 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 (b) 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 logic 1, the output from the XNOR circuit is simply the complement
of that shown.
In Figure b, the first data bit is XNORed with the reference bit. If they are the same,
the XNOR output is logic 1; if they are different, the XNOR output is logic 0. The balanced
modulator operates the same as a conventional BPSK modulator; a logic I produces +sin ωct
at the output, and A logic 0 produces –sin ωct at the output.

DPSK Receiver

The figure below 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
signalling element in the balanced modulator. If they are the same, a logic 1(+ voltage) is
generated. If they are different, a logic 0 (- voltage) is generated.

If 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
DBPSK.
Advantages

⚫ Simplicity of circuit.

⚫ No carrier recovery circuit needed.

⚫ BW requirement of DPSK (fb) is reduced compared to that of BPSK (2fb).

Disadvantages

 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.

Quadrature Phase Shift Keying (QPSK)

This is the phase shift keying technique, in which the sine wave carrier takes four
phase reversals such as 45°, 135°, -45°, and -135°.
If these kinds 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 transmitter

A block diagram of a QPSK modulator is shown in Figure below. Two 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 the Q 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 ωct and - sin ωct), and two phases are possible at the output of the
Q balanced modulator (+cos ωct), and (-cos ωct).
When the linear summer combines the two quadrature (90° out of phase) signals,
there are four possible resultant phasors given by these expressions: + sin ωct + cos ωct, + sin
ωct - cos ωct, -sin ωct + cos ωct, and -sin ωct - cos ωct.

Example:
For the QPSK modulator shown in the above figure, construct the truth table, Phasor
diagram, and constellation diagram.
Solution:
For a binary data input of Q = 0 and I= 0, the two inputs to the I balanced modulator
are -1 and sin ωct, and the two inputs to the Q balanced modulator are -1 and cos ωct.

Consequently, the outputs are


I balanced modulator =(-1)(sin ωct) = -1 sin ωct
Q balanced modulator =(-1)(cos ωct) = -1 cos ωct and the output of the linear summer is
-1 cos ωct - 1 sin ωct = 1.414 sin (ωct - 135°)
For the remaining dibit codes (01, 10, and 11), the procedure is the same. The results are
shown in figure below.

Figure QPSK modulator: (a) truth table; (b) Phasor diagram; (c) constellation diagram

In above figures b and c, it can be seen that with QPSK each of the four possible
outputs Phasor has exactly the same amplitude. Therefore, the binary information must be
encoded entirely in the phase of the output signal

In figure b, it can be seen that the angular separation between any two adjacent Phasor 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.
The figure below shows the output phase-versus-time relationship for a QPSK
modulator.
QPSK Receiver:

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 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 above figures.

To illustrate the demodulation process, let the incoming QPSK signal is -sin ωct + cos
ωct. Mathematically, the demodulation process is as follows.

The received QPSK signal (-sin ωct + cos ωct) is one of the inputs to I product
detector. The other input is the recovered carrier (sin ωct). The output of the I product
detector is
Again, the received QPSK signal (-sin ωct + cos ωct) is one of the inputs to the Q
product detector. The other input is the recovered carrier shifted 90° in phase (cos ωct). 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.
Power spectral Density of QPSK

OPTIMAL RECEPTION OF DIGITAL SIGNALS


Introduction

⚫ Digital data can be transmitted directly or as is usually the case, by modulating a


carrier.

⚫ The received signal is corrupted by noise and hence there is a finite probability that
the receiver will make an error in determining within each time interval, whether a 1
or 0 is transmitted.

Therefore, an optimal receiver design with an objective to reduce error probability in


reception is must.

Baseband Signal Receiver

⚫ Consider that a binary encoded signal consists of a time sequence of voltage levels
+V or –V

⚫ With noise present, the received signal and noise together will yield sample values
generally different from ± V.

⚫ Assumption: Noise is Gaussian and therefore the noise voltage has probability
density which is entirely symmetrical with respect to zero volts.

⚫ Probability that noise has increased the sample value is same as the probability that
the noise has decreased the sample value.

⚫ If sample value is positive the transmitted level was +V, and if the sample value is
negative the transmitted level was –V.

⚫ It is possible that at the sampling time the noise voltage may be of magnitude larger
than V and of a polarity opposite to the polarity assigned to the transmitted bit.
⚫ The probability of error can be reduced by processing the received signal plus noise in
such a manner that we are then able to find sample time where the sample voltage due
to the signal is emphasized relative to the sample voltage due to the noise.

Operation of Baseband signal Receiver

⚫ The operation of the receiver during each bit interval is independent of the waveform
during past and future bit intervals.

⚫ Signal s(t) and white Gaussian noise n(t) of PSD η/2 is presented to an integrator.

⚫ At time t=0+ we require that capacitor C be uncharged which is ensured by a brief


closing of the switch sw1 at time t=0-, thus relieving C of any charge it may have
acquired during the previous interval.

⚫ Sample is taken at the output of the integrator by closing this sampling switch sw2.

⚫ This sample is taken at the end of the bit interval at t=T.

⚫ Signal processing is described by the phrase integrate and dump.

⚫ Dump- refers to the abrupt discharge of the capacitor after each sampling.
Peak Signal to RMS Noise output Voltage Ratio

⚫ The integrator yields an output which is the integral of its input multiplied by 1/RC.
Using τ=RC, we have

⚫ no(T) has a gaussian probability density.

⚫ The output of the integrator, before the sampling switch is

𝒗𝒐(𝒕) = 𝒔𝒐(𝒕) + 𝒏𝒐(𝒕)

The signal output so(t) is a ramp, in each bit interval of duration T. At the end of the
interval the ramp attains the voltage so(T) which is +VT/τ or VT/τ, depending on
whether the bit is 1 or 0

⚫ At the end of each interval the switch SW1 closes momentarily to discharge the
capacitor so that so(t) drops to zero.
⚫ The noise no(t) also starts each interval with no(0)=0 and has the random value no(T)
at the end of each interval.

⚫ The sampling switch SW2 closes briefly just before the closing of SW1 and hence
reads the voltage

𝒗𝒐(𝑻) = 𝒔𝒐(𝑻) + 𝒏𝒐(𝑻)

⚫ The output signal voltage to be as large as possible in comparison with the noise
voltage.

⚫ Hence a figure of merit of interest is the signal-to-noise ratio

⚫ SNR increases with increasing bit duration T and it depends on V 2T which is the
normalized energy of the bit signal.

⚫ The integrator filters the signal and noise such that the signal voltage increases
linearly with time, while the standard deviation (rms value) of the noise increases
more slowly, as .

⚫ Thus the integrator enhances the signal relative to the noise, and this enhancement
increases with time.

Probability of Error (Pe) of Integrate-and-dump receiver

⚫ Function of a receiver: To distinguish the bit 1 from the bit 0 in the presence of
noise.

⚫ A most important characteristic is the probability that an error will be made in such a
determination.

⚫ The probability density of the noise sample no(T) is Gaussian and hence appears as
follows

⚫ The density is therefore given by


⚫ Where 𝝈 𝟐 , the variance, is 𝝈 𝟐 = ̅𝒏̅𝟐̅(̅𝑻̅) given by equ.
𝒐 𝒐 𝒐

⚫ Suppose that during some bit interval the input signal voltage is held at, say –V.

⚫ Then, at the sample time, the signal sample voltage is

while the noise sample is no(T)

 If no (T) is positive and larger in magnitude than VT/τ, the total sample voltage

𝒗𝒐(𝑻) = 𝒔𝒐(𝑻) + 𝒏𝒐(𝑻)

will be positive. Such a positive sample voltage will result in an error.

 The probability of such a misinterpretation, that is, the probability that

is given by the area of the shaded region in the figure.

⚫ The probability of error is given by

𝒏𝒐(𝑻)
⚫ Defining, 𝒙 ≡ ⁄ , and using above equ.
√𝟐𝝈𝒐

⚫ In which Es=V2T, is the signal energy of a bit.

⚫ If the signal voltage were held instead at +V during some bit interval, then it is clear
from the symmetry of the situation that the probability of error would again be given
by Pe in the above equ.
⚫ The probability of error Pe is plotted below

⚫ Pe decreases rapidly as Es/η 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 be wrong more than half the time on the
average.

Optimum Receiver for both Baseband and Pass band

⚫ Assume that the received signal is a binary waveform.

⚫ One binary bit is represented by a signal waveform s1(t) which persists for time T,
while the other bit is represented by the waveform s2(t)which also lasts for an interval.

⚫ In the case of transmission of baseband, s1(t)=+V, while s2(t)=-V.

⚫ For other modulation systems, different waveforms are transmitted. Example:

PSK signalling: s1(t) = A cos (ωct)

s2(t) = - A cos (ωct)

FSK signalling: s1(t)= A cos (ωc+Ω)t

s2(t)= A cos(ωc-Ω)t

⚫ As shown in the above figure, the input, which s1(t) or s2(t), is corrupted by the
addition of noise n(t).

⚫ The noise is Gaussian and has a spectral density G(f).


⚫ The signal & noise are filtered and then sampled at the end of each bit interval.

⚫ The output sample is either Vo(T)=s01(T)+n0(T)

(or)

Vo(T)=s02(T)+n0(T)

⚫ Assumption: Immediately after each sample, every energy storing element in the
filter will be discharged.

⚫ In the absence of noise the output sample would be V0(T)=s01 (T) or s02(T).

⚫ When noise is present to minimize the probability of error one should assume that
s1(t) has been transmitted if V0(T) is closer to s01(T) than to s02(T), similarly it is
assumed s2(t) has been transmitted if V0 (T) is closer to s02(T).

⚫ Decision boundary is midway between s01(T) and s02(T).

⚫ Example: For the Integrate and Dump system, where

and

the decision boundary is Vo(T)=0.

⚫ Decision boundary is

⚫ Example: Suppose that s01(T) > s02(T) and that s2(t) was transmitted.

⚫ If at the sampling time, the noise n0(T) is positive and larger in magnitude than the
voltage difference,

an error will have been made .

⚫ That is , an error will result if

⚫ Hence the probability of error is


𝒏𝒐(𝑻)
Substituting, 𝒙 ≡ ⁄ , then
√𝟐𝝈𝒐

⚫ The complimentary error function is monotonically decreasing function of its


argument.

⚫ Pe decreases as the difference s01(T)-s02(T) becomes larger and as the rms noise
voltage σo becomes smaller.

⚫ The optimum filter, then, is the filter which maximizes the ratio

Eye Diagrams/Eye Patterns

The quality of digital transmission systems are evaluated using the bit error rate. Degradation
of quality occurs in each process modulation, transmission, and detection.

The eye pattern is experimental method that contains all the information concerning the
degradation of quality. Therefore, careful analysis of the eye pattern is important in analyzing
the degradation mechanism.

Eye patterns can be observed using an oscilloscope. The received wave is applied to the
vertical deflection plates of an oscilloscope and the saw tooth wave at a rate equal to
transmitted symbol rate is applied to the horizontal deflection plates, resulting display is eye
pattern as it resembles human eye.

• The interior region of eye pattern is called eye opening.


PULSE MODULATION
 Carrier is a train of pulses
 Example: Pulse Amplitude Modulation (PAM), Pulse width modulation (PWM) ,
Pulse Position Modulation (PPM)

Types of Pulse Modulation:

⚫ The immediate result of sampling is a pulse-amplitude modulation (PAM) signal

⚫ PAM is an analog scheme in which the amplitude of the pulse is proportional to the
amplitude of the signal at the instant of sampling

⚫ Another analog pulse-forming technique is known as pulse-duration modulation


(PDM). This is also known as pulse-width modulation (PWM)

⚫ Pulse-position modulation is closely related to PDM

Pulse Amplitude Modulation:

In PAM, amplitude of pulses is varied in accordance with instantaneous value of


modulating signal.

PAM Generation:

The carrier is in the form of narrow pulses having frequency fc. The uniform
sampling takes place in multiplier to generate PAM signal. Samples are placed Ts sec
away from each other.
Figure PAM Modulator

⚫ The circuit is simple emitter follower.


⚫ In the absence of the clock signal, the output follows input.
⚫ The modulating signal is applied as the input signal.
⚫ Another input to the base of the transistor is the clock signal.
⚫ The frequency of the clock signal is made equal to the desired carrier pulse train
frequency.

⚫ The amplitude of the clock signal is chosen the high level is at ground level(0v) and
low level at some negative voltage sufficient to bring the transistor in cutoff region.

⚫ When clock is high, circuit operates as emitter follower and the output follows in the
input modulating signal.

⚫ When clock signal is low, transistor is cutoff and output is zero.


⚫ Thus the output is the desired PAM signal.

PAM Demodulator:
⚫ The PAM demodulator circuit which is just an envelope detector followed by a
second order op-amp low pass filter (to have good filtering characteristics) is as
shown below

Figure PAM Demodulator


Pulse Width Modulation:
⚫ In this type, the amplitude is maintained constant but the width of each pulse is varied
in accordance with instantaneous value of the analog signal.

⚫ In PWM information is contained in width variation. This is similar to FM.

⚫ In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.

Pulse Position Modulation:

⚫ In this type, the sampled waveform has fixed amplitude and width whereas the
position of each pulse is varied as per instantaneous value of the analog signal.

⚫ PPM signal is further modification of a PWM signal.

PPM & PWM Modulator:

Figure PWM & PPM Modulator

• The PPM signal can be generated from PWM signal.

• The PWM pulses obtained at the comparator output are applied to a mono stable multi
vibrator which is negative edge triggered.
• Hence for each trailing edge of PWM signal, the monostable output goes high. It
remains high for a fixed time decided by its RC components.

• Thus as the trailing edges of the PWM signal keeps shifting in proportion with the
modulating signal, the PPM pulses also keep shifting.

• Therefore all the PPM pulses have the same amplitude and width. The information is
conveyed via changing position of pulses.

Figure PWM & PPM Modulation waveforms

PWM Demodulator:

Figure PWM Demodulator


⚫ Transistor T1 works as an inverter.

⚫ During time interval A-B when the PWM signal is high the input to transistor T2 is
low.

⚫ Therefore, during this time interval T2 is cut-off and capacitor C is charged through
an R-C combination.

⚫ During time interval B-C when PWM signal is low, the input to transistor T2 is high,
and it gets saturated.

⚫ The capacitor C discharges rapidly through T2. The collector voltage of T2 during B-
C is low.

⚫ Thus, the waveform at the collector of T2is similar to saw-tooth waveform whose
envelope is the modulating signal.

⚫ Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.

PPM Demodulator:

Figure PPM Demodulator

⚫ The gaps between the pulses of a PPM signal contain the information regarding the
modulating signal.

⚫ During gap A-B between the pulses the transistor is cut-off and the capacitor C gets
charged through R-C combination.

⚫ During the pulse duration B-C the capacitor discharges through transistor and the
collector voltage becomes low.

⚫ Thus, waveform across collector is saw-tooth waveform whose envelope is the


modulating signal.

⚫ Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.
Multiplexing

Multiplexing is the set of techniques that allows the simultaneous transmission of multiple
signals across a single common communications channel.

Multiplexing is the transmission of analog or digital information from one or more sources to
one or more destination over the same transmission link.

Although transmissions occur on the same transmitting medium, they do not necessarily
occupy the same bandwidth or even occur at the same time.

Frequency Division Multiplexing

Frequency division multiplexing (FDM) is a technique of multiplexing which means


combining more than one signal over a shared medium. In FDM, signals of different
frequencies are combined for concurrent transmission.

In FDM, the total bandwidth is divided to a set of frequency bands that do not
overlap. Each of these bands is a carrier of a different signal that is generated and modulated
by one of the sending devices. The frequency bands are separated from one another by strips
of unused frequencies called the guard bands, to prevent overlapping of signals.

The modulated signals are combined together using a multiplexer (MUX) in the
sending end. The combined signal is transmitted over the communication channel, thus
allowing multiple independent data streams to be transmitted simultaneously. At the
receiving end, the individual signals are extracted from the combined signal by the process of
demultiplexing (DEMUX).

FDM system Transmitter

 Analog or digital inputs: mi (t); i = 1,2, ... n


 Each input modulates a subcarrier of frequency fi; i=1, 2, .... n
 Signals are summed to produce a composite baseband signal denoted as mb(t)
 fi is chosen such that there is no overlap.
Spectrum of composite baseband modulating signal

FDM system Receiver

 The Composite base band signal mb(t) is passed through n band pass filters with
response centred on fi
 Each si(t) component is demodulated to recover the original analog/digital data.
Time Division Multiplexing

TDM technique combines time-domain samples from different message signals (sampled at
same rate) and transmits them together across the same channel.

The multiplexing is performed using a commutator (switch). At the receiver a decommutator


(switch) is used in synchronism with the commutator to demultiplex the data.

The input signals, all band limited to fm (max) by the LPFs are sequentially sampled at the
transmitter by a commutator.

The Switch makes one complete revolution in Ts,(1/fs) extracting one sample from each
input. Hence the output is a PAM waveform containing the individual message sampled
periodically interlaced in time.

A set of pulses consisting of one sample from each input signal is called a frame.

At the receiver the de-commutator separates the samples and distributes them to a bank of
LPFs, which in turn reconstruct the original messages.

Synchronizing is provided to keep the de-commutator in step with the commutator.


Elements of Digital Communication Systems

Figure Elements of Digital Communication Systems

1. Information Source and Input Transducer:


The source of information can be analog or digital, e.g. analog: audio or video
signal, digital: like teletype signal. In digital communication the signal produced by
this source is converted into digital signal which consists of 1′s and 0′s. For this we
need a source encoder.
2. Source Encoder:
In digital communication we convert the signal from source into digital signal
as mentioned above. The point to remember is we should like to use as few binary
digits as possible to represent the signal. In such a way this efficient representation of
the source output results in little or no redundancy. This sequence of binary digits is
called information sequence.

Source Encoding or Data Compression: the process of efficiently converting


the output of whether analog or digital source into a sequence of binary digits is
known as source encoding.

3. Channel Encoder:
The information sequence is passed through the channel encoder. The purpose
of the channel encoder is to introduce, in controlled manner, some redundancy in the
binary information sequence that can be used at the receiver to overcome the effects
of noise and interference encountered in the transmission on the signal through the
channel.
For example take k bits of the information sequence and map that k bits to
unique n bit sequence called code word. The amount of redundancy introduced is
measured by the ratio n/k and the reciprocal of this ratio (k/n) is known as rate of code
or code rate.
4. Digital Modulator:
The binary sequence is passed to digital modulator which in turns convert the
sequence into electric signals so that we can transmit them on channel (we will see
channel later). The digital modulator maps the binary sequences into signal wave
forms , for example if we represent 1 by sin x and 0 by cos x then we will transmit sin
x for 1 and cos x for 0. ( a case similar to BPSK)
5. Channel:
The communication channel is the physical medium that is used for
transmitting signals from transmitter to receiver. In wireless system, this channel
consists of atmosphere , for traditional telephony, this channel is wired , there are
optical channels, under water acoustic channels etc.We further discriminate this
channels on the basis of their property and characteristics, like AWGN channel etc.
6. Digital Demodulator:
The digital demodulator processes the channel corrupted transmitted
waveform and reduces the waveform to the sequence of numbers that represents
estimates of the transmitted data symbols.
7. Channel Decoder:
This sequence of numbers then passed through the channel decoder which
attempts to reconstruct the original information sequence from the knowledge of the
code used by the channel encoder and the redundancy contained in the received data

Note: The average probability of a bit error at the output of the decoder is a
measure of the performance of the demodulator – decoder combination.

8. Source Decoder:
At the end, if an analog signal is desired then source decoder tries to decode
the sequence from the knowledge of the encoding algorithm. And which results in the
approximate replica of the input at the transmitter end.

9. Output Transducer:
Finally we get the desired signal in desired format analog or digital.

Advantages of digital communication

 Can withstand channel noise and distortion much better as long as the noise and the
distortion are within limits.
 Regenerative repeaters prevent accumulation of noise along the path.
 Digital hardware implementation is flexible.
 Digital signals can be coded to yield extremely low error rates, high fidelity and
well as privacy.
 Digital communication is inherently more efficient than analog in realizing the
exchange of SNR for bandwidth.
 It is easier and more efficient to multiplex several digital signals.
 Digital signal storage is relatively easy and inexpensive.
 Reproduction with digital messages is extremely reliable without deterioation.
 The cost of digital hardware continues to halve every two or three years while
performance or capacity doubles over the same time period.

Disadvantages

 TDM digital transmission is not compatible with FDM


 A Digital system requires large bandwidth.

Elements of PCM System

Sampling:

 Process of converting analog signal into discrete signal.


 Sampling is common in all pulse modulation techniques
 The signal is sampled at regular intervals such that each sample is proportional to
amplitude of signal at that instant
 Analog signal is sampled every 𝑇𝑠 𝑆𝑒𝑐𝑠, called sampling interval. 𝑓𝑠=1/𝑇𝑆 is called
sampling rate or sampling frequency.
 𝑓𝑠=2𝑓𝑚 is Min. sampling rate called Nyquist rate. Sampled spectrum (𝜔) is repeating
periodically without overlapping.
 Original spectrum is centered at 𝜔=0 and having bandwidth of 𝜔𝑚. Spectrum can be
recovered by passing through low pass filter with cut-off 𝜔𝑚.
 For 𝑓𝑠<2𝑓𝑚 sampled spectrum will overlap and cannot be recovered back. This is
called aliasing.
Sampling methods:

 Ideal – An impulse at each sampling instant.


 Natural – A pulse of Short width with varying amplitude.
 Flat Top – Uses sample and hold, like natural but with single amplitude value.

Fig. 4 Types of Sampling

Sampling of band-pass Signals:

 A band-pass signal of bandwidth 2fm can be completely recovered from its samples.
Min. sampling rate =2×𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ

=2×2𝑓𝑚=4𝑓𝑚

 Range of minimum sampling frequencies is in the range of 2×𝐵𝑊 𝑡𝑜 4×𝐵𝑊

Instantaneous Sampling or Impulse Sampling:

 Sampling function is train of spectrum remains constant impulses throughout


frequency range. It is not practical.

Natural sampling:

 The spectrum is weighted by a sinc function.


 Amplitude of high frequency components reduces.

Flat top sampling:

 Here top of the samples remains constant.


 In the spectrum high frequency components are attenuated due sinc pulse roll off.
This is known as Aperture effect.
 If pulse width increases aperture effect is more i.e. more attenuation of high frequency
components.
PCM Generator
Transmission BW in PCM
PCM Receiver

Quantization

 The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels.
 Quantization is representing the sampled values of the amplitude by a finite set of
levels, which means converting a continuous-amplitude sample into a discrete-time
signal
 Both sampling and quantization result in the loss of information.
 The quality of a Quantizer output depends upon the number of quantization levels
used.
 The discrete amplitudes of the quantized output are called as representation levels or
reconstruction levels.
 The spacing between the two adjacent representation levels is called a quantum or
step-size.
 There are two types of Quantization
o Uniform Quantization
o Non-uniform Quantization.
 The type of quantization in which the quantization levels are uniformly spaced is
termed as a Uniform Quantization.
 The type of quantization in which the quantization levels are unequal and mostly the
relation between them is logarithmic, is termed as a Non-uniform Quantization.

Uniform Quantization:
• There are two types of uniform quantization.
– Mid-Rise type
– Mid-Tread type.
• The following figures represent the two types of uniform quantization.

• The Mid-Rise type is so called because the origin lies in the middle of a raising part
of the stair-case like graph. The quantization levels in this type are even in number.
• The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
• Both the mid-rise and mid-tread type of uniform quantizer is symmetric about the
origin.
Quantization Noise and Signal to Noise ratio in PCM System
Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization:
Non-Uniform Quantization:
In non-uniform quantization, the step size is not fixed. It varies according to certain
law or as per input signal amplitude. The following fig shows the characteristics of Non
uniform quantizer.
Companding PCM System
• Non-uniform quantizers are difficult to make and expensive.
• An alternative is to first pass the speech signal through nonlinearity before quantizing
with a uniform quantizer.
• The nonlinearity causes the signal amplitude to be compressed.
– The input to the quantizer will have a more uniform distribution.
• At the receiver, the signal is expanded by an inverse to the nonlinearity.
• The process of compressing and expanding is called Companding.
Differential Pulse Code Modulation (DPCM)
Redundant Information in PCM
Introduction to Delta Modulation
Condition for Slope overload distortion occurrence
Slope overload distortion will occur if
Expression for Signal to Quantization Noise power ratio for Delta Modulation

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