Ce Unit Ii RR

EC6651 CE UNIT II
6th SEM EEE - RMKCET
UNIT II DIGITAL COMMUNICATION
Pulse modulations concepts of sampling and sampling theormes, PAM,

PWM,PPM,PTM,quantization and coding : DCM, DM, slope overload error. ADM, DPCM,
OOK systems ASK, FSK, PSK, BSK, QPSK, QAM, MSK, GMSK, applications of Data
communication.
PART B QUESTIONS
1. Describe the working of pulse code modulation(PCM) & DPCM system with its block
diagram.
2. Describe the working of Delta modulation(DM) & Adaptive Deltal Modulation(ADM)

system.
3. Discuss about the working principle of ASK, FSK, PSK modulator and detector with neat
diagram.
4. Explain QPSK with a block diagram and discuss the phasor diagram for sinusoids. Also
analyze about the bandwidth considerations for QPSK.
5. Explain QAM techniques with neat diagram.
6. Discuss the method of modulation and demodulation in MSK & GMSK with equations
and block diagrams.
I. Merits of Digital Communication:
o Digital signals are very easy to receive. The receiver has to just detect whether
the pulse is low or high.
o AM FM signals become corrupted over much short distances as compared to
digital signals. In digital signals, the original signal can be reproduced accurately.
o The signals lose power as they travel, which is called attenuation. When AM and
FM signals are amplified, the noise also get amplified. But the digital signals can
be cleaned up to restore the quality and amplified by the regenerators.
o The noise may change the shape of the pulses but not the pattern of the pulses.
o AM and FM signals can be received by any one by suitable receiver. But digital
signals can be coded so that only the person, who is intended for, can receive
them.
o AM and FM transmitters are real time systems. I.e. they can be received only at
the time of transmission. But digital signals can be stored at the receiving end.
o The digital signals can be stored, or used to produce a display on a computer
monitor or converted back into analog signal to drive a loud speaker.
RAMACHANDRAN.R Page 1
EC6651 CE UNIT II
Introduction:
1. Sampling:
Sampling is the processes of converting continuous-time analog signal, xa(t), into a

discrete-time signal by taking the samples at discrete-time intervals
Sampling analog signals makes them discrete in time but still continuous
valued
If done properly (Nyquist theorem is satisfied), sampling does not introduce

distortion
Sampled values:
The value of the function at the sampling points
Sampling interval:
The time that separates sampling points (interval b/w samples), Ts

If the signal is slowly varying, then fewer samples per second will be required than if
the waveform is rapidly varying
So, the optimum sampling rate depends on the maximum frequency component
present in the signal
Analog-to-digital conversion is (basically) a 2 step process:
Sampling:
Convert from continuous-time analog signal xa(t) to discrete-time continuous value signal
x(n)
Is obtained by taking the samples of xa(t) at discrete-time intervals, Ts
Quantization:
Convert from discrete-time continuous valued signal to discrete time discrete valued signal
Sampling Rate (or sampling frequency fs):
The rate at which the signal is sampled, expressed as the number of samples
per second (reciprocal of the sampling interval), 1/Ts = fs
Nyquist Sampling Theorem (or Nyquist Criterion):
If the sampling is performed at a proper rate, no info is lost about the original
signal and it can be properly reconstructed later on
EC6651 CE UNIT II
Statement:
If a signal is sampled at a rate at least, but not exactly equal to twice the
max frequency component of the waveform, then the waveform can be exactly
reconstructed from the samples without any distortion
fs > 2fm
Fundamental Rule of Sampling (Nyquist Criterion)
The value of the sampling frequency fs must be greater than twice the highest
signal frequency fmax of the signal
Types of sampling
Ideal Sampling
Natural Sampling
Flat-Top Sampling
2. PULSE MODULATION:
Introduction:
A digital signal is superior to an analog signal because it is more robust to noise and
can easily be recovered, corrected and amplified. For this reason, the tendency today
is to change an analog signal to digital data.
The process of transmitting signals in the form of pulses (discontinuous signals) by

using special techniques.
Types of pulse modulation
Pulse modulation
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2. PULSE ANALOG MODULATION:
Definition:
Pulse modulation consists of sampling analog information signals and then converting
those samples into discrete pulses and transporting the pulses from a source to a destination
over a physical transmission medium
A series of regularly recurring pluses is made to vary in amplitude, duration, shape, or time as
a function of the modulating signal.
TYPES:
pulse-width modulation (PWM)
pulse-duration modulation (PDM)
pulse-position modulation (PPM)
2.1 Pulse Amplitude Modulation(PAM)
DEFNITION:
In PAM,amplitude of pulses is varied in accordance with instantaneous value of

modulating signal.
The signal is sampled at regular intervals such that each sample is proportional to the
amplitude of the signal at that sampling instant. This technique is called sampling.
For minimum distortion, the sampling rate should be more than twice the signal
frequency.
There are two classes of PAM signals:
Natural PAM
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Flat top PAM
Generating Natural PAM
The PAM wave form with natural sampling can be generated using a CMOS circuit
consisting of a clock
and analog switch as shown.
Instantaneous Sampling(flat-top )
This type of PAM signal consists of instantaneous samples.
w(t) is sampled at t = kTs
EC6651 CE UNIT II
The sample values w(kTs ) determine the amplitude of the flat-top rectangular pulses.
The carrier is in the form of narrow pulses having frequency fs.The uniform sampling takes
place in multiplier to generate PAM signal.Samples are placed Ts sec away from each other.
Modula PAM
ting Signal
Signal
Depending upon the shape and polarity of the sampled pulses, PAM is of two types,
Natural PAM sampling occurs when top portion of the pulses are subjected to follow
the modulating wave.
Flat topped PAM sampling is often used because of the ease of generating the
modulated wave. In this pulses have flat tops after modulation.
RECEIVER:
EC6651 CE UNIT II
The PAM signal can be detected by passing it through a low pass filter.
2.2 Pulse Width Modulation (PWM or PLM or PDM):
In this type, the amplitude is maintained constant but the duration or length or width
of each pulse is varied in accordance with instantaneous value of the analog signal.
The negative side of the signal is brought to the positive side by adding a fixed d.c.
voltage.
Width Modulated Pulses
That is why the information is contained in width variation. This is similar to FM.
EC6651 CE UNIT II
In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.
PWM GENERATOR OR TRANSMITTER:
A simple method to generate the PWM pulse train corresponding to a given signal is
the intersective PWM: the signal (here the green sinewave) is compared with a
sawtooth waveform (blue). When the latter is less than the former, the PWM signal
(magenta) is in high state (1). Otherwise it is in the low state (0).
In this case a sawtooth signal of frequency fs is a sampling signal.
It is applied to inverting terminal of a comparator with modulating signal at non

inverting terminal.
O/P remains high as long as modulating signal is higher than that of ramp signal.
2.3 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.
EC6651 CE UNIT II
The vertical dotted lines shown in last slide treated as reference lines.
The PPM pulses marked 1,2 and 3 go away from their respective reference lines.This
corresponds to increase in modulating signal amplitude.
Then as modulating signal decreases the PPM pulses 4,5,6,7 come closer to their
respective reference lines.
PPM GENERATION:
The PPM signal can be generated from PWM signal.
The PWM pulses obtained at the comparator output are applied to a monostable
multivibrator which is ve 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 own RC components.
Thus as the trailing edges of the PWM signal keeps shifting in propotion 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.
Application:
PAM is used as intermediate form of modulation with PSK, QAM and PCM
PWM and PPM are used in special purpose communications systems mainly for
the military but seldom used for commercial system
Difference between Pulse amplitude modulation (PAM) , Pulse Width/Duration

modulation and PPM
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S.No PAM PWM / PDM PPM

1. Amplitude of the pulse is Width of the pulse is The relative position of
proportional to the proportional to the the pulse is
amplitude of the amplitude of the proportional to the
modulating signal modulating signal amplitude of the
modulating signal.
2. The bandwidth of the The bandwidth of the The bandwidth of the
transmission channel
transmission channel transmission channel
depends on width of the depends on rise time depends on rising time
pulse. of the pulse. of the pulse.
3. The instantaneous power The instantaneous The instantaneous
of the transmitter varies.
power of the power of the transmitter
transmitter varies. remains constant.
4. Noise is interference is Noise is interference Noise is interference is
high. is minimum. minimum.
5. Similar to amplitude Similar to frequency Similar to phase
modulation. modulation. modulation.
3. Pulse Code Modulation (PCM):

In Pulse Code Modulation (PCM), the analog signal is sampled then converted to a
serial n-bit binary code for transmission.
It is commonly used for digital transmission

Not really a type of modulation but rather a form of digitally coding analog signals
In PCM, pulses are of fixed length & fixed amplitude
PCM is a binary system where a pulse or lack of a pulse within a prescribed time slot
represents either a logic 1 or logic 0 condition
EC6651 CE UNIT II
The band pass filter (BFF) limits the frequency of the analog input signal to a
standard voice-band frequency range of 300Hz to 3000Hz
The sample and hold (S&H) circuit periodically samples the analog input signal and
convert those samples to a multilevel PAM signal
The Analog-to-Digital converter (ADC) converts the PAM sample to parallel PCM
codes, which are converted to serial binary data and outputted to transmission line as
serial digital pulses
The transmission line repeaters are placed at prescribed distances to regenerate the
digital pulses
In the receiver, the serial-to-parallel converter converts the serial pulses received from
the transmission line to parallel PCM code
The Digital-to-Analog Converter (DAC ) converts the parallel PCM codes to a
multilevel PAM signals
Hold circuit and Low pass filter converts the PAM signals back to its original analog
form
An integrated circuit that performs the PCM encoding and decoding functions is
called a codec (coder/decoder)
3.1 PCM Sampling
EC6651 CE UNIT II
The function of a sampling circuit in a PCM transmitter is to periodically sample the

continually changing analog input voltage and convert those samples to a series of
constant- amplitude pulses that can more easily be converted to binary PCM code.
A sample-and-hold circuit is a nonlinear device (mixer) with two inputs: the sampling
pulse and the analog input signal.
For the ADC to accurately convert a voltage to a binary code, the voltage must be
relatively constant so that the ADC can complete the conversion before the voltage
level changes. If not, the ADC would be continually attempting to follow the changes
and may never stabilize on any PCM code.
Essentially, there are two basic techniques used to perform the sampling function
natural sampling
flat-top sampling
Natural sampling is when tops of the sample pulses retain their natural shape during
the sample interval, making it difficult for an ADC to convert the sample to a PCM
code.
EC6651 CE UNIT II
EC6651 CE UNIT II
The most common method used for sampling voice signals in PCM systems is flat-
top sampling, which is accomplished in a sample-and-hold circuit. The purpose of a
sample-and-hold circuit is to periodically sample the continually changing analog
input voltage and convert those samples to a series of constant-amplitude PAM
voltage levels.
3.2 PCM Sampling Rate
The Nyquist sampling theorem establishes the minimum Nyquist sampling rate (fs) that
can be used for a given PCM system.
For a sample to be reproduced accurately in a PCM receiver, each cycle of the analog
input signal (fa) must be sampled at least twice.
Consequently, the minimum sampling rate is equal to twice the highest audio input
frequency.
If fs is less than two times fa an impairment called alias or foldover distortion occurs.
Mathematically, the minimum Nyquist sampling rate is:
fs 2fa
3.3 Quantization and the Folded Binary Code
Quantization is the process of converting an infinite number of possibilities to a finite

number of conditions.
Analog signals contain an infinite number of amplitude possibilities.
Converting an analog signal to a PCM code with a limited number of combinations

requires quantization.
EC6651 CE UNIT II
With quantization, the total voltage range is subdivided into a smaller number of
subranges.
The PCM code shown in Table 10-2 is a three-bit sign- magnitude code with eight
possible combinations (four positive and four negative).
The leftmost bit is the sign bit (1 = + and 0 = -), and the two rightmost bits represent
magnitude.
This type of code is called a folded binary code because the codes on the bottom half
of the table are a mirror image of the codes on the top half, except for the sign bit.
With a folded binary code, each voltage level has one code assigned to it except zero
volts, which has two codes, 100 (+0) and 000 (-0).
The magnitude difference between adjacent steps is called the quantization interval or
quantum.
If the magnitude of the sample exceeds the highest quantization interval, overload
distortion (also called peak limiting) occurs.
Assigning PCM codes to absolute magnitudes is called quantizing.
The magnitude of a quantum is also called the resolution.
The resolution is equal to the voltage of the minimum step size, which is equal to the
voltage of the least significant bit (Vlsb) of the PCM code.
The smaller the magnitude of a quantum, the better (smaller) the resolution and the
more accurately the quantized signal will resemble the original analog sample.
EC6651 CE UNIT II
3.4 Quantization
The likelihood of a sample voltage being equal to one of the eight quantization levels
is remote. Therefore, as shown in the figure, each sample voltage is rounded off
(quantized) to the closest available level and then converted to its corresponding PCM
code.
The rounded off error is called the called the quantization error (Qe).
To determine the PCM code for a particular sample voltage, simply divide the voltage
by the resolution, convert the quotient to an n-bit binary code, and then add the sign
bit.
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Companding:
Companding is the process of compressing and then expanding
High amplitude analog signals are compressed prior to txn. and then expanded in the
receiver
It is a means of improving dynamic range
Early PCM systems used analog companding, where as modern systems use digital
companding.
EC6651 CE UNIT II
There are two standard companding methods.
A-law Companding
-law Companding
u-Law is used in North America and Japan
A-Law is used elsewhere to compress digital telephone signals
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-law Companding
A-law Companding
EC6651 CE UNIT II
4. Delta modulation:
Definition:
Delta modulation uses a single-bit PCM code to achieve digital transmission of analog
signals. With conventional PCM, each code is a binary representation of both the sign and
magnitude of a particular sample. Therefore, multiple-bit codes are required to represent the
many values that the sample can be.
In delta modulation, rather than transmit a coded representation of the sample, only a
single bit is transmitted, which simply indicated whether that sample is larger or
smaller than the previous sample
If the current sample is smaller than the previous sample, a logic 0 is transmitted. If
the current sample is larger than the previous sample, a logic 1 is transmitted.Delta
modulation transmitter:
Modulator:
EC6651 CE UNIT II
The input analog is sampled and converted to a PAM signal, which is compared with
the output of the DAC
The up-down counter is incremented or decremented depending on whether the
previous sample is larger or smaller than the current sample
The up-down counter is clocked at a rate equal to the sample rate. Therefore, the up-
down counter is updated after each comparison
Let m n m(nTs ) , n 0,1,2,

where Ts is the sampling period and m(nTs ) is a sample of m(t ).
The error signal is
e n m n mq n 1
e q n sgn( e n )
mq n mq n 1 eq n
where mq n is the quantizer output , e q n is
the quantized version of e n , and is the step size
Waveform:
EC6651 CE UNIT II
Delta modulation receiver:
Figure shows the block diagram of a delta modulation receiver. As you can see, the
receiver is almost identical to the transmitter except for the comparator.
EC6651 CE UNIT II
As the logic is and 0s are received, the up/down-counter is incremented or

decremented accordingly.
Consequently, the output of the DAC in the decoder is identical to the output of the
DAC in the transmitter.
With delta modulation, each sample requires the transmission of only 1 bit; therefore,
the bit rates associated with delta modulation are lower than conventional PCM systems.
However, there are two problems associated with delta modulations that do not occur
with conventional PCM: slope overload and granular noise.
Advantages:
In delta modulation, only one bit is transmitted for each sample which will require
less bandwidth compared to other digital modulation systems.
Disadvantages:
There are two problems associated with delta modulation
i) Slope overload distortion

ii) Granular noise
Drawbacks of delta modulation:
There are two problems associated with delta modulation i) slope overload ii) granular
noise
EC6651 CE UNIT II
Granular noise can be reduced by decreasing the step size. Therefore to reduce the
granular noise, a small resolution needed, and to reduce the possibility of slope
overload occurring, a large resolution is required (compromise required)
Granular noise is more prevalent in analog signals that have gradual slopes and whose
amplitudes vary only a small amount. Slope overload is more prevalent in analog
signals that have steep slopes or whose amplitudes vary rapidly.
Slope overload:
The slope of the analog signal is greater than the delta modulator can maintain and is
called slope overload
Increasing the clock frequency reduces the probability of slope overload occurring
Another way to prevent slope overload is to increase the magnitude of the minimum
step size
Granular Noise:
When the original analog input signal has a relatively constant amplitude, the
reconstructed signal has variations that were not present in the original signal. This is
called granular noise
Granular noise in delta modulation is analogous to quantization noise in conventional
PCM
Granular noise can be reduced by decreasing the step size. Therefore to reduce the
granular noise, a small resolution needed, and to reduce the possibility of slope
overload occurring, a large resolution is required (compromise required)
Granular noise is more prevalent in analog signals that have gradual slopes and whose
amplitudes vary only a small amount. Slope overload is more prevalent in analog
signals that have steep slopes or whose amplitudes vary rapidly.
EC6651 CE UNIT II
4.1 Adaptive delta modulation:
Adaptive delta modulation is a delta modulation system where the step size of the
DAC is automatically varied, depending on the amplitude characteristics of the analog
input signals
In adaptive delta modulation, after a predetermined no of consecutive 1s and 0s, the

step size is automatically increased
When an alternate sequence of 1s and 0s is occurring, this indicates that the
possibility of granular noise occurring is high. Consequently, the DAC will
automatically revert to its minimum step size and thus reduce the magnitude of the
noise error
A common algorithms for an adaptive delta modulator is when three consecutive 1s
and 0s occur, the step size of the DAC is increased or decreased by a factor of 1.5
Various other algorithms may be used for adaptive delta modulators, depending on
particular system requirements
EC6651 CE UNIT II
Adaptive delta modulation system: (a) Transmitter. (b) Receiver.
5. Differential pulse code modulation:

In a typical PCM encoded speech waveform, there are often successive
samples taken in which there is little difference between the amplitudes of the two
samples. Which lead to identical PCM code which is redundant.
With DPCM, the difference in the amplitude of two successive samples is

transmitted rather than the actual sample. Because the range of sample differences is
typically less than the range of individual samples, fewer bits are required for DPCM
than conventional PCM.
Encode the changes between consecutive samples
Example
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The value of the differences between samples are much smaller than those of the
original samples. Less bits are used to encode the signal (e.g. 7 bits instead of 8 bits)
DPCM Transmitter:
The analog input signal is bandlimited to one-half the sample rate, then compared
with the preceding accumulated signal level in the differentiator
The output of the differentiator is the difference between the two signals
The difference is PCM encoded and transmitted
The ADC operates the same as in a conventional PCM system, except that it typically
fewer bits per sample
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DPCM system. (a) Transmitter. (b) Receiver
DPCM Receiver:
Received sample is converted back to analog, stored, and then summed with the next
sample received.
6. Types of Digital Modulation:
In digital communications, the high frequency analog carriers are modulated bu relatively
low frequency digital information signals (0s & 1s) and the information are transmitted in
digital forms.
There are three basic types of modulation techniques for the transmission of digital signals.
Amplitude Shift Keying (ASK):
The most basic (binary) form of ASK involves the process of switching the
carrier either on or off, in correspondence to a sequence of digital pulses that
constitute the information signal. One binary digit is represented by the
presence of a carrier, the other binary digit is represented by the absence of a
carrier. Frequency remains fixed
Frequency Shift Keying (FSK):
The most basic (binary) form of FSK involves the process of varying the
frequency of a carrier wave by choosing one of two frequencies (binary FSK)
in correspondence to a sequence of digital pulses that constitute the
information signal. Two binary digits are represented by two frequencies
around the carrier frequency. Amplitude remains fixed
Phase Shift Keying (PSK):
Another form of digital modulation technique which we will not discuss
EC6651 CE UNIT II
6.1 Amplitude shift keying.

Amplitude Shift Keying (ASK) is sometimes called Digital Amplitude
Modulation (DAM), where a binary information signal directly modulated the
amplitude of an analog carrier.
Mathematically,
A
vask (t ) [1 vm (t )] cos(c t )
2
1
vask (t )
Whrere = amplitude-shift keying wave
vm (t )
= digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)
c
= analog carrier radian frequency (rad/sec, 2fct)
Considering Vm(t) = as a normalized binary waveform, where +1V = Logic 1 and -1V
= Logic 0.
For logic 1 input (Vm(t) =+1V) the equation 1 becomes
A
v ask (t ) [1 1] cos( c t ) A cos( c t )
2
For logic 0 input (Vm(t) = -1V) the equation 1 becomes
A
v ask (t ) [111] cos(c t ) 0
2
vask (t ) A cos(ct )
Thus, the modulated wave is either or 0. Hence, the carrier is either
On or Off thats why ASK is sometimes referenced to as On-Off Keying(OOK).
Waveform:
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The bit rate equals the baud since symbol size is one.
6.2 Frequency Shift Keying (FSK)
If the information signal is digital and the frequency (f) is varied proportional to the
information signal Frequency Shift Keying (FSK) is produced.
It is a low performance digital modulation, sometimes called Binary FSK

(BFSK), because it is similar to standard FM except the modulating signal is a binary
signal.
The general expression for FSK is
v fsk (t ) Vc cos{2 [ f c vm (t )f ]t}

1
v fsk (t )
Whrere = Binary FSK waveform
Vc = peak analog carrier amplitude
vm (t )
= digital information (modulating) signal (volts) [binary input]
fc
= analog carrier centre frequency
f
= peak change in analog carrier
If the modulating signal is normalized binary waveform, where +1V = Logic 1 and
-1V = Logic 0.
For logic 1 input (Vm(t) =+1V) the equation 1 becomes
v fsk (t ) Vc cos{2 [ f c f ]t}
EC6651 CE UNIT II
For logic 0 input (Vm(t) =+-V) the equation 1 becomes
v fsk (t ) Vc cos{2 [ f c f ]t}
As the binary input changes between logic 0 and logic 1, the output frequency
shifts between two frequencies i) a mark or logic 1 frequency (fm) and ii) a
space for logic 0 frequency (fs)
The mark and space frequencies are separated from the carrier by the peak
f f
frequency deviation ( ) and from each other by 2 .
fm fs
f
2
Waveform:
The time of one bit (t b) is the same as the time the FSK output is the mark or space
frequency (ts). Thus, the bit time equals to the time of an FSK signaling element, and
the bit rate is equals the baud.
Mathematically,
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fb fb
baud baud fb
N 1
, Where N=1 for FSK (bit rate)
FSK is the exception to the rule for digital modulation. The minimum
bandwidth for FSK is give as
B ( f s fb ) ( f m fb )
( f s f m ) 2 f b 2f 2 f b [ ( f s f m ) 2f ]
B 2(f f b )
fm fb
The above equation is similar to the Carsons rule ( is replaced by )
B
Where = Minimum Nyquist bandwidth
f
= Frequency deviation
fb
= input bit rate
Bessel function can also be used to determine approximate bandwidth for an

FSK wave
The fastest rate of change (highest fundamental frequency) in a non return to

zero (NRZ) binary signals occurs when 1s and 0s are occurring continuously
(Square wave).
Since it takes a low and high to produce a cycle, the highest fundamental
frequency present in the square wave equals the half the bit rate.
f a fb / 2
Therefore, Where fa is maximum fundamental frequency of binary
input and fb is input bit rate (bps)
EC6651 CE UNIT II
The formula used for modulation index in FM is also valid for FSK, thus
f
h
fa
, where h is modulation index called as h-factor in FSK
Deviation ratio: the worst case or widest bandwidth occurs when both the
frequency deviation and the modulating signal frequency are at maximum
( f fm ) / 2 ( fs fm )
h s
fb / 2 fb
Where h = h-factor, fm = Mark frequency, fs = Space frequency and fb = bit rate
various types of FSK receivers with necessary diagrams.
There are three FSK detectors available i) Non-Coherent FSK demodulator

ii) Coherent FSK demodulator and iii) PLL FSK demodulator
Non-Coherent FSK demodulator:
Envelope detector indicate the power in each passband and the comparator respond to
the largest of two powers
This type of FSK detection is referred to as non-coherent detection, since there is no
frequency involved in the demodulation process that is either synchronized in phase,
frequency or both with the incoming FSK signal.
Coherent FSK demodulator:
EC6651 CE UNIT II
The two transmitted frequencies (Mark and Space frequencies) are not generally
continuous; so, it is not practical to reproduce a local reference that is coherent with
both frequencies. Consequently Coherent FSK detection is seldom used.
PLL FSK demodulator:
The most common circuit used for demodulating binary FSK

PLL-FSK demodulator works similar to PLL-FM demodulator
The input to the PLL shifts between the mark and space frequencies, the dc error
voltage at the output of the phase comparator follows the frequency shift
Because there are only two frequencies (Mark and Space) there are also only two
output error voltages
Therefore, the output is a two-level (binary) representation o f the FSK output.
The natural frequency of the PLL is made equal to the center frequency of the FSK
modulator
6.3 BPSK(Binary Phase Shift Keying) :
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It is a simplest form of PSK where N=1 and M=2, with BPSK two phases are
possible for the carrier. One phase (0) represents Logic 1 and other phase
(180) represents Logic 0.
BPSK sometimes called as Phase Reversal Keying (PRK) and Bi-phase
Modulation.
BPSK Transmitter:
The balanced modulator acts as a phase reversing switch

Depending upon 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.
Balanced Ring Modulator:
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a) For logic 1 input
b) For logic 0 input
BPSK receiver.
Block diagram:
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Mathematically,
Sin ( c t )
For a BPSK input signal of + , the output of the balanced modulator is
Sin ( c t ) Sin ( c t )
Output = x
Sin 2 ( c t )
=
1 cos( 2 c t ) 1 cos( 2 c t )

2 2 2
= [filtered out]
= +1/2 V (Positive voltage represents a demodulated Logic 1)
Sin ( c t )
For a BPSK input signal of - , the output of the balanced modulator is
Output = - x-
Sin 2 ( c t )
=-
1 cos( 2 c t ) 1 cos( 2 c t )
( )
2 2 2
= [filtered out]
= -1/2 V (Negative voltage represents a demodulated Logic 0)
bandwidth consideration of BPSK.
EC6651 CE UNIT II
A balanced modulator is a product modulator. Therefore, the output signal is

the product of two input signals
The data is either at +1V or -1V for Logic 1 and Logic 0 respectively
Therefore, the signal is either at + or - , first is in-pahse with
the reference carrier and later is 180 out of phase with the reference oscillator
The bit rate equals baud, the fundamental frequency (f a) of an alternate 1 or 0
bit sequence is equal to the half the bit rate (fb/2)
Mathematically,
Sin (2f a t ) xSin(2f c t )
BPSK output =
Where fa is maximum fundamental frequency of binary input and
fc is reference carrier frequency
Using Trigonometry formula
1 1
cos[ 2 ( f c f a )t ] cos[ 2 ( f c f a )t ]
2 2
=
Thus, the minimum double-sided Nyquist bandwidth
( fc fa ) ( fc fa ) 2 fa fa fb / 2
B= , Where
B 2( f b / 2) f b
(input bit rate)
EC6651 CE UNIT II
7. Quadrature Phase Shift Keying(QPSK)

QPSK: The two successive bits in a bit stream ar combined together to form a message and
each message is represented by a distinct value of phase shift of the carrier. Each symbol or
message contains two bits so the symbol duration Ts =2Tb.These symbols are transmitted by
the same carrier at four different phase shifts as shown below.
Truth table:
Symbol Phase
00 -135
01 -45
10 135
11 45
It is an M ary encoding technique, where N=2 bits M=4, so four output phases are
possible for a single carrier.
QPSK modulator:
EC6651 CE UNIT II
Two bits (a dibit) are clocked into the bit splitter

One bit is directed to the I channel and the other to the Q channel
The I bit modulates a carrier that is in phase with the reference oscillator and the Q bit
modulates the carrier that is 90 out of phase or in quadrature with the reference
carrier
Once the dibit has been split into the I and Q channels, the operation is the same as
BPSK modulator
A QPSK modulator is two BPSK modulators combined in parallel
For a Logic 1 = +1V and Logic 0 = -1V, two phase are possible at the output of the I
Sin c t Sin c t
balanced modulator (+ &- ) and two phase are possible at the output of
Cosc t Cos c t
the Q balanced modulator (+ &- )
The linear summer combines that two quadrature signals, there are four possible
Sinc t Cosc t Sin c t Cosc t Sinc t Cosc t
resultant phases i) + + ii) + - iii) - + and
Sin c t Cosc t
iv) - -
Receiver:
EC6651 CE UNIT II
the received QPSK signal is given to the power splitter which directs the input
signal to the I and Q product detectors and carrier recovery circuit.
The function of carrier recovery circuit is to reproduce the original transmit
carrier signal.
The recovered carrier is frequency and phase coherent with the transmit
reference carrier.
The I and q product detectors demodulate the QPSK signal and generates I and
Qdata bits.
The output of the product detectors Given to combining circuit, where they are
converted from parallel I and Q data channels to a single binary output data
stream.
Constellation diagram:
Phasor Diagram:
EC6651 CE UNIT II
bandwidth considerations of QPSK system.
EC6651 CE UNIT II
The input data are divided into two channels, the bit rage is either the I or Q channel
is equal to one-half of the input data rate (fb/2)
The highest fundamental frequency present at the data input to the I or Q balanced
modulator is equal to one fourth of the input data rate (one-half of fb/2 = fb/4)
The min double side Nyquist Bandwidth for I or Q balanced modulator
fN = 2 x fb/4 = fb/2
Therefore, the bandwidth compression is realized in QPSK (ie., the minimum
bandwidth is lessthan the incoming bit rage)
The output of the balanced modulators can be expressed mathematically as
Sin a t Sin c t
Output = ( )( )
Sin 2f a t Sin 2f c t
=( )( )
Where fa = fb/4
fb
Sin 2 ( )t Sin 2f c t
4
=( )( )
1 f 1 f
cos[ 2 ( f c b )t ] cos[ 2 ( f c b )t ]
2 4 2 4
=
The output spectrum extends from fc + fb/4 to fc - fb/4 and minimum bandwidth
= ( fc + fb/4 ) ( fc - fb/4)
= 2 (fb/4) = fb/2
EC6651 CE UNIT II
8. Quadrature Amplitude Modulation
Definition:
If both the amplitude and phase of the carrier are varied with respect to the binary
information signal, then it is called quadrature amplitude modulation
QAM is combination of ASK and PSK
8-QAM:
8 QAM is an M-ary encoding technique where M=8
Eight QAM (8-QAM) is an M-ary encoding technique where M = 8. Unlike 8-PSK,

the output signal from an 8-QAM modulator is not a constant-amplitude signal.
Bandwidth:
The minimum bandwidth require for QAM transmission is same as that required for ASK and
PSK.
B=fb
16-QAM:
16-QAM is an M-ary system where M =16. The input data are acted on in groups of four (24
= 16). As with 8-QAM, both the phase and the amplitude of the transmit carrier are varied.
16 QAM transmitter:
EC6651 CE UNIT II
The input binary data are divided into four channels: I, I', Q, and Q'. The bit
rate in each channel is equal to one-fourth of the input bit rate (fb/4).
The I and Q bits determine the polarity at the output of the 2- to-4-level
converters (a logic 1 = positive and a logic 0 = negative). The I' and Q' buy
determine the magnitude (a logic 1 = 0.821 V and a logic 0 = 0.22 V).
For the I product modulator they are +0.821 sin c t, -0.821 sin c t,
+0.22 sin c t, and -0.22 sin c t. For the Q product modulator, they are +0.821
cos c t, +0.22 cos c t, -0.821 cos c t, and -0.22 cos c t.
The linear summer combines the outputs from the I and Q channel product
modulators and produces the 16 output conditions necessary for 16-QAM. Figure
shows the truth table for the I and Q channel 2-to-4-level converters.
Receiver:
The received signal is given to power splitter, it splits the I and Q signals
Carrier recovery circuit regenerates the carrier signal
Incoming signal is mixed with recovered carrier in product detecter, it gives the
outout as PAM signal.
Th e binary signals occurs at the output of analog to digital converter circuit.
TRUTH TABLE
EC6651 CE UNIT II
Phasor diagram; constellation diagram.
Comparison between QPSK and QAM
EC6651 CE UNIT II
Comparison between BPSK and DPSK:
Advantages and dis adv of digital modulation over analog modulation
ADVANTAGES:
1.Digital signals are very easy to receive. The receiver has to just detect whether the pulse is
low or high.
2.AM & FM signals become corrupted over much short distances as compared to digital
signals. In digital signals, the original signal can be reproduced accurately.
3.The signals lose power as they travel, which is called attenuation. When AM and FM
signals are amplified, the noise also get amplified. But the digital signals can be cleaned up
to restore the quality and amplified by the regenerators.
4.The noise may change the shape of the pulses but not the pattern of the pulses.
5.AM and FM signals can be received by any one by suitable receiver. But digital signals can
be coded so that only the person, who is intended for, can receive them.
EC6651 CE UNIT II
6.AM and FM transmitters are real time systems. i.e. they can be received only at the time
of transmission. But digital signals can be stored at the receiving end.
7.The digital signals can be stored.
DIS ADVANTAGES:
Special encoding and decoding techniques may be necessary to increase transmission

rates, thus making the pulse stream more difficult to recover
9. Minimum-shift keying(MSK)
In digital modulation, minimum-shift keying(MSK) is a type of continuous-phase frequency-

shift keying that was developed in the late 1950s and 1960s.
Similar to OQPSK(Offset quadrature phase-shift keying), MSK is encoded with bits

alternating between quadrature components, with the Q component delayed by half the
symbol period.
EC6651 CE UNIT II
EC6651 CE UNIT II
Gaussian Minimum Shift Keying (GMSK)
Gaussian Minimum Shift Keying (GMSK) is a form of continuous-phase FSK in

which the phase is changed between symbols to provide a constant envelope
Consequently it is a popular alternative to QPSK
A Gaussian filter is used before frequency modulation
GMSK is advanced version of MSK
Gaussian Minimum Shift keying (GMSK) Gaussian Minimum Shift Keying (GMSK) is a
modification of MSK (i.e. CPFSK with h = 1/2). A filter used to reduce the bandwidth of a
baseband pulse train prior to modulation is called a pre-modulation filter.
The Gaussian pre-modulation filter smooths the phase trajectory of the MSK signal thus
limiting the instantaneous frequency variations. The result is an FM modulated signal with a
much narrower bandwidth. This bandwidth reduction does not come for free since the pre-
modulation filter smears the individual pulses in pulse train.
As a consequence of this smearing in time, adjacent pulses interfere with each other
generating what is commonly called inter-symbol interference or ISI. In the applications
where GMSK is used, the trade-off between power efficiency and bandwidth efficiency is
well worth the cost.
EC6651 CE UNIT II
The requirements for the filter are:
should have a sharp cut-off
narrow bandwidth
impulse response should show no overshoot
Gaussian shaped response to an impulse and no ringing
In this way the basic MSK signal is converted to GMSK modulation
response of the filter to a single 1 is a phase change of /2, is equivalent to choosing

the constant K to satisfy the following equation
PART-A
1.Define bit time and baud rate.
Bit time: It is the reciprocal of the bit rate
Baud rate: The rate of change of a signal on the transmission medium after encoding
and modulation have occurred.
Baud = 1/ts
2.What is difference between coherent and non coherent detection?
Coherent detection Non- Coherent detection
Carrier which is in perfect No carrier recovery circuit needed

coherence with that used in for detection.
transmitter is used for
demodulation.
EC6651 CE UNIT II
Carrier recovery circuit is

needed for detection
Relatively complex Simple implementation
3.Define Digital Amplitude Modulation.
Amplitude Shift Keying (ASK) is sometimes called Digital Amplitude

Modulation (DAM), where a binary information signal directly modulated the
amplitude of an analog carrier.
Mathematically,
A
vask (t ) [1 vm (t )] cos(c t )
2
vask (t )
Whrere = amplitude-shift keying wave
vm (t )
= digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)
c
= analog carrier radian frequency (rad/sec, 2fct)
4.What is the minimum bandwidth required for an FSK system?
Bandwidth required=fm-fs+ 2/tb
1/tb =fb=bit rate,fm=mark frequency,fs=space frequency
5. What is difference between coherent and non coherent detection?
Coherent detection Non- Coherent detection
Carrier which is in perfect No carrier recovery circuit needed

coherence with that used in for detection.
transmitter is used for
demodulation.
Carrier recovery circuit is

needed for detection
Relatively complex Simple implementation
EC6651 CE UNIT II
6. Compare binary PSK with QPSK.

BPSK QPSK
Binary Phase Shift Keying Quadrature Phase Shift Keying
One bit form a symbol Two bits form a symbol
Two possible symbols Four possible symbols
Minimum bandwidth required = f b Minimum bandwidth required = f b / 2
where f b is bit rate where f b is bit rate
7. Define Bandwidth efficiency. What is the bandwidth efficiency of BPSK and 8-

PSK system?
It is the ratio of the transmission bit rate to the minimum bandwidth required for a
particular modulation scheme.
For BPSK , transmission rate = f b and minimum bandwidth = f b
Band width efficiency = 1
For 8-PSK , transmission rate = f b and minimum bandwidth = f b/3
Band width efficiency = 3
8. What is the difference between probability of error P(e) and bit error rate BER?
P(e) Probability of error is a theoretical (mathematical) expectation of the bit
error rate for a given system.
BER is an empirical record of a systems actual bit error performance.
For Example, if a system has a P(e) of 10 -5 , this mean that, you can expect one bit
error in every 100,000 bits transmitted.
-5
If a system has a BER of 10 , this mean that, there was one bit error for every
100,000 bits transmitted.
BER is measured and then compared to the expected probability of error to evaluate
the systems performance.
EC6651 CE UNIT II
9. Define PAM, PWM modulation? [AU- April/May- 2004]

pulse width modulation:
The pulse width modulation is defined as the width of the carrier pulse is varied in
accordance with the variation in the modulating signal at sampling instant.
pulse amplitude modulation. [AU-Nov/Dec- 2005]
The pulse amplitude modulation is defined as the amplitude of the carrier pulse
is varied in accordance with the variation in the modulating signal at sampling instant.
10. For a PCM system with a maximum input frequency of 4 KHz. Determine the
minimum sample rate and alias frequency produced if 5 KHz audio signal were
allowed to enter the sample hold circuit.
Solution:-
We know the sampling rate fs > 2 fm ; fs > 2 4 KHz = 8 KHz
Alias frequency = Minimum sampling rate highest frequency of signal to be

sampled
= 8 KHz 5 KHz = 3 KHz.
11. Draw the phasor diagram of QPSK signal.

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EC6651 CE UNIT II

6th SEM EEE - RMKCET

UNIT II DIGITAL COMMUNICATION

Pulse modulations concepts of sampling and sampling theormes, PAM,

2. Describe the working of Delta modulation(DM) & Adaptive Deltal Modulation(ADM)

5. Explain QAM techniques with neat diagram.

I. Merits of Digital Communication:

Sampling is the processes of converting continuous-time analog signal, xa(t), into a

If done properly (Nyquist theorem is satisfied), sampling does not introduce

The value of the function at the sampling points

The time that separates sampling points (interval b/w samples), Ts

Analog-to-digital conversion is (basically) a 2 step process:

Is obtained by taking the samples of xa(t) at discrete-time intervals, Ts

Sampling Rate (or sampling frequency fs):

Nyquist Sampling Theorem (or Nyquist Criterion):

Fundamental Rule of Sampling (Nyquist Criterion)

The process of transmitting signals in the form of pulses (discontinuous signals) by

Types of pulse modulation

2. PULSE ANALOG MODULATION:

pulse-width modulation (PWM)

pulse-duration modulation (PDM)

pulse-position modulation (PPM)

2.1 Pulse Amplitude Modulation(PAM)

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

There are two classes of PAM signals:

Flat top PAM

Generating Natural PAM

and analog switch as shown.

This type of PAM signal consists of instantaneous samples.

w(t) is sampled at t = kTs

2.2 Pulse Width Modulation (PWM or PLM or PDM):

Width Modulated Pulses

PWM GENERATOR OR TRANSMITTER:

In this case a sawtooth signal of frequency fs is a sampling signal.

It is applied to inverting terminal of a comparator with modulating signal at non

2.3 Pulse Position Modulation

PPM signal is further modification of a PWM signal.

The PPM signal can be generated from PWM signal.

Difference between Pulse amplitude modulation (PAM) , Pulse Width/Duration

S.No PAM PWM / PDM PPM

3. Pulse Code Modulation (PCM):

It is commonly used for digital transmission

3.1 PCM Sampling

The function of a sampling circuit in a PCM transmitter is to periodically sample the

3.2 PCM Sampling Rate

Mathematically, the minimum Nyquist sampling rate is:

3.3 Quantization and the Folded Binary Code

Quantization is the process of converting an infinite number of possibilities to a finite

Analog signals contain an infinite number of amplitude possibilities.

Converting an analog signal to a PCM code with a limited number of combinations

Assigning PCM codes to absolute magnitudes is called quantizing.

The magnitude of a quantum is also called the resolution.

Companding is the process of compressing and then expanding

It is a means of improving dynamic range

There are two standard companding methods.

u-Law is used in North America and Japan

A-Law is used elsewhere to compress digital telephone signals

Let m n m(nTs ) , n 0,1,2,

Delta modulation receiver:

As the logic is and 0s are received, the up/down-counter is incremented or

There are two problems associated with delta modulation

i) Slope overload distortion

Drawbacks of delta modulation: