Notes - PC Eee 701 - Unit 05
Notes - PC Eee 701 - Unit 05
Notes - PC Eee 701 - Unit 05
UNIT-5
MODULATION OF DIGITAL SIGNAL
Introduction
⚫ The digital data is transmitted over the channel directly. There is no carrier or
any modulation. Suitable for transmission over short distances.
⚫ The digital data modulates high frequency sinusoidal carrier. Suitable for
transmission over longer distances.
⚫ 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.
⚫ 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.
⚫ Pass band transmission can take place over wireless channels also.
Introduction
⚫ 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.
⚫ Hence the ASK waveform looks like an On-Off of the signal. Therefore it is also
known as the On-Off Keying(OOK)
⚫ 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 PSD of ASK signal is same as that of a baseband on-off signal but shifted in the
frequency domain by ± fc
⚫ 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.
⚫ 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.
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.
⚫ Assume
⚫ The signalling components of the receiver output at the end of the signalling interval
are
This scheme involves detection in the form of ‘rectifier’ & ‘low pass filter’.
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.
⚫ Where Ak=A when the kth transmitted bit bk=1, and Ak=0, when bk=0
Where nc(t) and ns(t) are the quadrature components of narrow band noise
Advantages and Disadvantages of ASK
⚫ Advantages
⚫ Disadvantages
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.
⚫ In Binary FSK, the frequency of the carrier is shifted according to the binary symbol.
Phase unaffected.
⚫ If b(t)=1, then
b(t)=0, then
⚫ Thus
Generation of BFSK
⚫ The level shifter converts ‘+1’ to √𝑃𝑠𝑇𝑏 and the zero level is unaffected.
⚫ 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.
⚫ Here 𝑃′ (𝑡) and 𝑃′ (𝑡) will be bipolar, alternating between +1 and -1, and
𝐻 𝐿
complementary.
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.
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
⚫ The highest fundamental frequency is equal to half the information bit rate.
⚫ Disadvantages
Applications of FSK
⚫ Used in low-speed modems (up to 1200bps) over analog voice-band telephone lines.
⚫ Principle of BPSK
⚫ In BPSK the binary symbol ‘1’ and ‘0’ modulate the phase of the carrier.
When the symbol is changed, then phase of the carrier is changed by 180o
⚫ Therefore
⚫ Which implies
Generation of BPSK
⚫ The signal is passed through band pass filter with centre frequency 2fc.
⚫ Frequency Divider:
⚫ 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.
⚫ 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
⚫ 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.
⚫ 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.
Disadvantages
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.
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
⚫ 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.
⚫ 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.
⚫ 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.
⚫ Sample is taken at the output of the integrator by closing this sampling switch sw2.
⚫ 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
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
⚫ The output signal voltage to be as large as possible in comparison with the noise
voltage.
⚫ 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.
⚫ 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
⚫ Suppose that during some bit interval the input signal voltage is held at, say –V.
If no (T) is positive and larger in magnitude than VT/τ, the total sample voltage
𝒏𝒐(𝑻)
⚫ Defining, 𝒙 ≡ ⁄ , and using above equ.
√𝟐𝝈𝒐
⚫ 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
⚫ 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.
⚫ 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.
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).
(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).
and
⚫ 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,
⚫ 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
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.
⚫ 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
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 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.
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
⚫ In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.
⚫ 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.
• 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.
PWM Demodulator:
⚫ 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:
⚫ 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.
⚫ 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.
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).
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 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.
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.
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
Sampling:
A band-pass signal of bandwidth 2fm can be completely recovered from its samples.
Min. sampling rate =2×𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ
=2×2𝑓𝑚=4𝑓𝑚
Natural sampling:
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