The document covers key concepts in digital communication, focusing on pulse modulation techniques including pulse amplitude modulation (PAM), pulse width modulation (PWM), and pulse position modulation (PPM), as well as the principles of sampling and quantization. It explains the advantages of digital signals over analog signals, such as better noise immunity and recovery capabilities. The document also discusses the Nyquist theorem, quantization error, and various encoding techniques used in digital communication systems.

1. EC6651 COMMUNICATION ENGINEERING
UNIT 2
Dr Gnanasekaran Thangavel
Professor and Head
Electronics and Instrumentation Engineering
R M K Engineering College
1

2. UNIT II DIGITAL COMMUNICATION
Pulse modulations – concepts of sampling and sampling
theorems, PAM, PWM, PPM, PTM, quantization and
coding : DCM, DM, slope overload error. ADM, DPCM,
OOK systems – ASK, FSK, PSK, BSK, QPSK, QAM, MSK,
GMSK, applications of Data communication.
2Dr Gnanasekaran Thangavel12/12/2017
YouTube Video Presentation
1. https://www.youtube.com/watch?v=_JMV4ywAJug
2. https://www.youtube.com/watch?v=QEubAxBfqKU

3. DIGITAL COMMUNICATION
3
 A digital signal is superior to an analog signal because it is more robust to noise and can easily be
recovered, corrected and amplified. For this reason, the tendency today is to change an analog signal to
digital data.
 Digital signals carry more information per second than analogue signals. This is the same whether
optical fibers, cables or radio waves are used.
 Digital signals maintain their quality over long distances better than analogue signals.
 Good processing techniques are available for digital signals such as source coding (data compression),
channel coding (error detection and correction), equalization etc.
 Easy to mix signals and data using digital techniques.
 Privacy is preserved by using data encryption. Using data encryption. only permitted receivers can be
allowed to detect the transmitted data. This is very useful in military applications.
 High speed computers and powerful software design tools are available. They make the development of
digital communication systems flexible.
 Internet is spread almost in every cities and towns. The compatibility of digital communication systems
with Internet has opened new area of applications

4. Limitations
• Generally, more bandwidth is required than that for analog systems.
• Synchronization is required.
• High power consumption (Due to various stages of conversion).
• Complex circuit, more sophisticated device making is also drawbacks of
digital system.
• Introduce sampling error
• As square wave is more affected by noise, That’s why while
communicating through channel we send sin waves but while operating
on device we use squire pulses.
4

5. Pulse Code Modulation (PCM)
PCM consists of three steps to digitize an analog signal:
• Sampling
• Quantization
• Binary encoding or Coding
5

6. Sampling
• The process of converting continuous time
signals into equivalent discrete time signals,
can be termed as Sampling. A certain
instant of data is continually sampled in the
sampling process.
• The following figure shows a continuous-
time signal x(t) and the corresponding
sampled signal xs(t). When x(t) is multiplied
by a periodic impulse train, the sampled
signal xs(t) is obtained.
• A sampling signal is a periodic train of
pulses, having unit amplitude, sampled at
equal intervals of time Ts, which is called as
sampling time. This data is transmitted at
the time instants Ts and the carrier signal is
transmitted at the remaining time.
6

7. Sampling Rate
• To discretize the signals, the gap between the samples should be fixed. That
gap can be termed as the sampling period Ts. Reciprocal of the sampling period
is known as sampling frequency or sampling rate fs.
• Mathematically, we can write it as
• Where,
• Fs is the sampling frequency or the sampling rate
• Ts is the sampling period
7

8. Sampling Theorem
• The sampling rate should be such that the data in the message signal should
neither be lost nor it should get over-lapped. The sampling theorem states that,
“a signal can be exactly reproduced if it is sampled at the rate fs, which is greater
than or equal to twice the maximum frequency of the given signal W.”
• Mathematically, we can write it as
• fs is the sampling rate
• W is the highest frequency of the given signal
• If the sampling rate is equal to twice the maximum frequency of the given signal
W, then it is called as Nyquist rate.
8

9. • The sampling theorem, which is also called as Nyquist theorem, delivers the
theory of sufficient sample rate in terms of bandwidth for the class of functions
that are band limited.
• For continuous-time signal x(t), which is band-limited in the frequency domain
is represented as shown in the following figure.
9

10. • If the signal is sampled above Nyquist rate, then the original signal can
be recovered. The following figure explains a signal, if sampled at a
higher rate than 2w in the frequency domain.
10

11. • If the same signal is sampled at a rate less than 2w, then the sampled
signal would look like the following figure.
11
• We can observe from the above pattern that there is over-lapping of information,
which leads to mixing up and loss of information. This unwanted phenomenon
of over-lapping is called as Aliasing.

12. • Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a signal,
taking on the identity of a low-frequency component in the spectrum of its sampled version.”
• Hence, the sampling rate of the signal is chosen to be as Nyquist rate. If the sampling rate is equal to twice
the highest frequency of the given signal W, then the sampled signal would look like the following figure.
12
In this case, the signal can be recovered without any loss. Hence, this is a good sampling rate.

13. • There are 3 sampling methods:
 Ideal - an impulse at each sampling instant
 Natural - a pulse of short width with varying amplitude
 Flattop - sample and hold, like natural but with single amplitude value
13

14. Example
• For an intuitive example of the Nyquist
theorem, let us sample a simple sine wave at
three sampling rates: fs = 4f (2 times the
Nyquist rate), fs = 2f (Nyquist rate), and fs = f
(one-half the Nyquist rate). Figure 4.24 shows
the sampling and the subsequent recovery of
the signal.
• It can be seen that sampling at the Nyquist
rate can create a good approximation of the
original sine wave (part a). Oversampling in
part b can also create the same
approximation, but it is redundant and
unnecessary. Sampling below the Nyquist rate
(part c) does not produce a signal that looks
like the original sine wave.
14

15. 4.15
According to the Nyquist theorem, the sampling rate
must be
at least 2 times the highest frequency contained in the
signal.
Note

16. • Consider the revolution of a hand of a clock.
The second hand of a clock has a period of 60 s.
According to the Nyquist theorem, we need to
sample the hand every 30 s (Ts = T or fs = 2f ).
In Figure 4.25a, the sample points, in order, are
12, 6, 12, 6, 12, and 6. The receiver of the
samples cannot tell if the clock is moving
forward or backward. In part b, we sample at
double the Nyquist rate (every 15 s). The
sample points are 12, 3, 6, 9, and 12. The clock
is moving forward. In part c, we sample below
the Nyquist rate (Ts = T or fs = f ). The sample
points are 12, 9, 6, 3, and 12. Although the
clock is moving forward, the receiver thinks
that the clock is moving backward.
16
Example

17. Quantization
• Sampling results in a series of pulses of varying amplitude
values ranging between two limits: a min and a max.
• The amplitude values are infinite between the two limits.
• We need to map the infinite amplitude values onto a finite
set of known values.
• This is achieved by dividing the distance between min and
max into L zones, each of height 
•  = (max - min)/L
17

18. Quantization Levels
• The midpoint of each zone is assigned a value from 0 to
L-1 (resulting in L values)
• Each sample falling in a zone is then approximated to the
value of the midpoint.
18

19. Quantization Zones
• Assume we have a voltage signal with amplitutes Vmin=-20V
and Vmax=+20V.
• We want to use L=8 quantization levels.
• Zone width = (20 - -20)/8 = 5
• The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to
+5, +5 to +10, +10 to +15, +15 to +20
• The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5
19

20. Assigning Codes to Zones
• Each zone is then assigned a binary code.
• The number of bits required to encode the zones, or the number
of bits per sample as it is commonly referred to, is obtained as
follows:
• nb = log2 L
• Given our example, nb = 3
• The 8 zone (or level) codes are therefore: 000, 001, 010, 011,
100, 101, 110, and 111
• Assigning codes to zones:
 000 will refer to zone -20 to -15
 001 to zone -15 to -10, etc.
20

21. Quantization and encoding of a sampled signal
21

22. Quantization Error
• When a signal is quantized, we introduce an error - the coded
signal is an approximation of the actual amplitude value.
• The difference between actual and coded value (midpoint) is
referred to as the quantization error.
• The more zones, the smaller  which results in smaller
errors.
• BUT, the more zones the more bits required to encode the
samples -> higher bit rate
22

23. Quantization Error and SNQR
• Signals with lower amplitude values will suffer more from quantization
error as the error range: /2, is fixed for all signal levels.
• Non linear quantization is used to alleviate this problem. Goal is to keep
SNQR fixed for all sample values.
• Two approaches:
 The quantization levels follow a logarithmic curve. Smaller ’s at lower amplitudes
and larger’s at higher amplitudes.
 Companding: The sample values are compressed at the sender into logarithmic
zones, and then expanded at the receiver. The zones are fixed in height.
23

24. Bit rate and bandwidth requirements of PCM
• The bit rate of a PCM signal can be calculated form the
number of bits per sample x the sampling rate
• Bit rate = nb x fs
• The bandwidth required to transmit this signal depends on the
type of line encoding used. Refer to previous section for
discussion and formulas.
• A digitized signal will always need more bandwidth than the
original analog signal. Price we pay for robustness and other
features of digital transmission.
24

25. Example
• We want to digitize the human voice. What is the bit rate,
assuming 8 bits per sample?
• Solution
• The human voice normally contains frequencies from 0 to 4000 Hz.
So the sampling rate and bit rate are calculated as follows:
25

26. PCM Decoder
• To recover an analog signal from a digitized signal we follow the
following steps:
 We use a hold circuit that holds the amplitude value of a pulse till the next
pulse arrives.
 We pass this signal through a low pass filter with a cutoff frequency that
is equal to the highest frequency in the pre-sampled signal.
• The higher the value of L, the less distorted a signal is recovered.
26

27. Components of a PCM decoder
• We have a low-pass analog signal of 4 kHz. If we send the analog signal, we need a channel with a minimum
bandwidth of 4 kHz. If we digitize the signal and send 8 bits per sample, we need a channel with a minimum
bandwidth of 8 × 4 kHz = 32 kHz.
27

28. Pulse Amplitude Modulation (PAM)
28

29. Pulse Width Modulation (PWM)
• In Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time
Modulation (PTM) technique, the width or the duration or the time of the pulse carrier varies,
which is proportional to the instantaneous amplitude of the message signal.
29
There are three types of PWM.
•The leading edge of the pulse being constant, the
trailing edge varies according to the message
signal. The waveform for this type of PWM is
denoted as (a) in the above figure.
•The trailing edge of the pulse being constant, the
leading edge varies according to the message
signal. The waveform for this type of PWM is
denoted as (b) in the above figure.
•The center of the pulse being constant, the
leading edge and the trailing edge varies
according to the message signal. The waveform
for this type of PWM is denoted as (c) shown in
the above figure.

30. Pulse Width Modulation
• In Pulse Width Modulation (PWM) or Pulse Duration
Modulation (PDM) or Pulse Time Modulation (PTM)
technique, the width or the duration or the time of the
pulse carrier varies, which is proportional to the
instantaneous amplitude of the message signal.
• There are three types of PWM.
• The leading edge of the pulse being constant, the
trailing edge varies according to the message signal.
The waveform for this type of PWM is denoted as (a) in
the above figure.
• The trailing edge of the pulse being constant, the
leading edge varies according to the message signal.
The waveform for this type of PWM is denoted as (b) in
the above figure.
• The center of the pulse being constant, the leading edge
and the trailing edge varies according to the message
signal. The waveform for this type of PWM is denoted as
(c) shown in the above figure.
30

31. Pulse Position Modulation
• Pulse Position Modulation (PPM) is an analog modulation
scheme in which, the amplitude and the width of the pulses
are kept constant, while the position of each pulse, with
reference to the position of a reference pulse varies
according to the instantaneous sampled value of the
message signal.
• The transmitter has to send synchronizing pulses (or simply
sync pulses) to keep the transmitter and the receiver in sync.
These sync pulses help to maintain the position of the
pulses. The following figures explain the Pulse Position
Modulation.
• Pulse position modulation is done in accordance with the
pulse width modulated signal. Each trailing edge of the pulse
width modulated signal becomes the starting point for pulses
in PPM signal. Hence, the position of these pulses is
proportional to the width of the PWM pulses.
31

32. Comparison between PAM, PWM, and PPM
32
PAM PWM PPM
Amplitude is varied Width is varied Position is varied
Bandwidth depends on the
width of the pulse
Bandwidth depends on the rise
time of the pulse
Bandwidth depends on the rise
time of the pulse
Instantaneous transmitter power
varies with the amplitude of the
pulses
Instantaneous transmitter power
varies with the amplitude and
the width of the pulses
Instantaneous transmitter power
remains constant with the width
of the pulses
System complexity is high System complexity is low System complexity is low
Noise interference is high Noise interference is low Noise interference is low
It is similar to amplitude
modulation
It is similar to frequency
modulation
It is similar to phase modulation

33. Pulse Time Modulation
• Pulse Time Modulation (PTM) is a class of signaling technique
that encodes the sample values of an analog signal onto the
• time axis of a digital signal.
• The two main types of pulse time modulation are:
1.Pulse Width Modulation (PWM)
2.Pulse Position Modulation (PPM)
33

34. Delta Modulation
• This scheme sends only the difference between pulses, if
the pulse at time tn+1 is higher in amplitude value than the
pulse at time tn, then a single bit, say a “1”, is used to
indicate the positive value.
• If the pulse is lower in value, resulting in a negative value, a
“0” is used.
• This scheme works well for small changes in signal values
between samples.
• If changes in amplitude are large, this will result in large
errors.
34

35. The process of delta modulation
35

36. Delta modulation components
36

37. Delta demodulation components
37

38. Adaptive Delta Modulation (ADM)
• it would be better if we can control the adjustment
of step-size, according to our requirement in order
to obtain the sampling in a desired fashion. This is
the concept of Adaptive Delta Modulation.
• Following is the block diagram of Adaptive delta
modulator.
• The gain of the voltage controlled amplifier is
adjusted by the output signal from the sampler. The
amplifier gain determines the step-size and both
are proportional.
• ADM quantizes the difference between the value of
the current sample and the predicted value of the
next sample. It uses a variable step height to
predict the next values, for the faithful reproduction
of the fast varying values.
38

39. Differential PCM
• For the samples that are highly
correlated, when encoded by PCM
technique, leave redundant information
behind.
• To process this redundant information
and to have a better output, it is a wise
decision to take a predicted sampled
value, assumed from its previous output
and summarize them with the quantized
values.
• Such a process is called as Differential
PCM (DPCM) technique.
• The DPCM Transmitter consists of
Quantizer and Predictor with two
summer circuits. Following is the block
diagram of DPCM transmitter.
39
• The signals at each point are named as −
• x(nTs) is the sampled input
• xˆ(nTs) is the predicted sample
• e(nTs) is the difference of sampled input and predicted output,
often called as prediction error
• v(nTs) is the quantized output
• u(nTs) is the predictor input which is actually the summer
output of the predictor output and the quantizer output

40. • The predictor produces the assumed samples from the previous outputs of the
transmitter circuit. The input to this predictor is the quantized versions of the input
signal x(nTs).
• Quantizer Output is represented as −
• v(nTs)=Q[e(nTs)]
• =e(nTs)+q(nTs)
• Where q (nTs) is the quantization error
• Predictor input is the sum of quantizer output and predictor output,
• u(nTs)=xˆ(nTs)+v(nTs)
• u(nTs)=xˆ(nTs)+e(nTs)+q(nTs)
• u(nTs)=x(nTs)+q(nTs)
• The same predictor circuit is used in the decoder to reconstruct the original input.
40

41. DPCM Receiver
• The block diagram of DPCM Receiver consists of a
decoder, a predictor, and a summer circuit. Following
is the diagram of DPCM Receiver.
• The notation of the signals is the same as the previous
ones. In the absence of noise, the encoded receiver
input will be the same as the encoded transmitter
output.
• As mentioned before, the predictor assumes a value,
based on the previous outputs. The input given to the
decoder is processed and that output is summed up
with the output of the predictor, to obtain a better
output.
41

42. On-off keying (OOK)
• Amplitude Shift Keying - ASK,
• Frequency Shift Keying - FSK,
• Phase Shift Keying - PSK,
• Binary Shift Keying - BSK –BPSK,BASK, BFSK
• Quadrature Phase Shift Keying - QPSK,
• Quadrature amplitude modulation - QAM
• Minimum shift keying - MSK
• Gaussian Minimum Shift Keying GMSK
42

43. Amplitude Shift Keying
• Amplitude Shift Keying (ASK) is a
type of Amplitude Modulation which
represents the binary data in the form
of variations in the amplitude of a
signal.
• Any modulated signal has a high
frequency carrier. The binary signal
when ASK modulated, gives a zero
value for Low input while it gives the
carrier output for High input.
• The following figure represents ASK
modulated waveform along with its
input.
43

44. ASK Modulator
• The ASK modulator block diagram comprises of
the carrier signal generator, the binary sequence
from the message signal and the band-limited
filter. Following is the block diagram of the ASK
Modulator.
• The carrier generator, sends a continuous high-
frequency carrier. The binary sequence from the
message signal makes the unipolar input to be
either High or Low. The high signal closes the
switch, allowing a carrier wave. Hence, the
output will be the carrier signal at high input.
When there is low input, the switch opens,
allowing no voltage to appear. Hence, the output
will be low.
• The band-limiting filter, shapes the pulse
depending upon the amplitude and phase
characteristics of the band-limiting filter or the
pulse-shaping filter.
44

45. ASK Demodulator
There are two types of ASK Demodulation techniques. They are −
• Asynchronous ASK Demodulation/detection
• Synchronous ASK Demodulation/detection
The clock frequency at the transmitter when matches with the clock frequency at
the receiver, it is known as a Synchronous method, as the frequency gets
synchronized. Otherwise, it is known as Asynchronous.
45

46. Asynchronous ASK Demodulator
• The Asynchronous ASK detector
consists of a half-wave rectifier, a low
pass filter, and a comparator.
Following is the block diagram for the
same.
• The modulated ASK signal is given to
the half-wave rectifier, which delivers a
positive half output. The low pass filter
suppresses the higher frequencies and
gives an envelope detected output
from which the comparator delivers a
digital output.
46

47. Synchronous ASK Demodulator
Synchronous ASK detector consists of
a Square law detector, low pass filter,
a comparator, and a voltage li
The ASK modulated input signal is
given to the Square law detector. A
square law detector is one whose
output voltage is proportional to the
square of the amplitude modulated
input voltage. The low pass filter
minimizes the higher frequencies. The
comparator and the voltage limiter help
to get a clean digital output. miter.
Following is the block diagram for the
same.
47

48. Frequency Shift Keying
• Frequency Shift Keying (FSK) is the
digital modulation technique in which
the frequency of the carrier signal
varies according to the digital signal
changes. FSK is a scheme of frequency
modulation.
• The output of a FSK modulated wave is
high in frequency for a binary High input
and is low in frequency for a binary Low
input. The binary 1s and 0s are called
Mark and Space frequencies.
• The following image is the
diagrammatic representation of FSK
modulated waveform along with its
input.
48

49. FSK Modulator
• The FSK modulator block diagram
comprises of two oscillators with a clock
and the input binary sequence. Following is
its block diagram.
• The two oscillators, producing a higher and
a lower frequency signals, are connected to
a switch along with an internal clock. To
avoid the abrupt phase discontinuities of the
output waveform during the transmission of
the message, a clock is applied to both the
oscillators, internally.
• The binary input sequence is applied to the
transmitter so as to choose the frequencies
according to the binary input.
49

50. FSK Demodulator
• There are different methods for demodulating a FSK
wave. The main methods of FSK detection are
asynchronous detector and synchronous detector.
• The synchronous detector is a coherent one, while
asynchronous detector is a non-coherent one.
50

51. Asynchronous FSK Detector
• The block diagram of Asynchronous FSK detector
consists of two band pass filters, two envelope
detectors, and a decision circuit. Following is the
diagrammatic representation.
• The FSK signal is passed through the two Band
Pass Filters (BPFs), tuned to Space and Mark
frequencies. The output from these two BPFs look
like ASK signal, which is given to the envelope
detector. The signal in each envelope detector is
modulated asynchronously.
• The decision circuit chooses which output is more
likely and selects it from any one of the envelope
detectors. It also re-shapes the waveform to a
rectangular one.
51

52. Synchronous FSK Detector
• The block diagram of Synchronous FSK
detector consists of two mixers with local
oscillator circuits, two band pass filters and a
decision circuit. Following is the diagrammatic
representation.
• The FSK signal input is given to the two mixers
with local oscillator circuits. These two are
connected to two band pass filters. These
combinations act as demodulators and the
decision circuit chooses which output is more
likely and selects it from any one of the
detectors. The two signals have a minimum
frequency separation.
• For both of the demodulators, the bandwidth of
each of them depends on their bit rate. This
synchronous demodulator is a bit complex than
asynchronous type demodulators.
52

53. Phase Shift Keying (PSK)
Phase Shift Keying (PSK) is the digital modulation technique in which the phase of the carrier signal is
changed by varying the sine and cosine inputs at a particular time. PSK technique is widely used for
wireless LANs, bio-metric, contactless operations, along with RFID and Bluetooth communications.
• PSK is of two types, depending upon the phases the signal gets shifted. They are −
Binary Phase Shift Keying (BPSK)
• This is also called as 2-phase PSK or Phase Reversal Keying. In this technique, the sine wave carrier
takes two phase reversals such as 0° and 180°.
• BPSK is basically a Double Side Band Suppressed Carrier (DSBSC) modulation scheme, for message
being the digital information.
Quadrature Phase Shift Keying (QPSK)
• This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals
such as 0°, 90°, 180°, and 270°.
• If this kind of techniques are further extended, PSK can be done by eight or sixteen values also,
depending upon the requirement.
53

54. BPSK Modulator
• The block diagram of Binary Phase Shift Keying
consists of the balance modulator which has the
carrier sine wave as one input and the binary
sequence as the other input. Following is the
diagrammatic representation.
• The modulation of BPSK is done using a balance
modulator, which multiplies the two signals applied at
the input. For a zero binary input, the phase will be 0°
and for a high input, the phase reversal is of 180°.
• Following is the diagrammatic representation of BPSK
Modulated output wave along with its given input.
• The output sine wave of the modulator will be the
direct input carrier or the inverted (180° phase shifted)
input carrier, which is a function of the data signal.
54

55. BPSK Demodulator
• The block diagram of BPSK demodulator consists of
a mixer with local oscillator circuit, a band pass filter,
a two-input detector circuit. The diagram is as
follows.
By recovering the band-limited message signal, with
the help of the mixer circuit and the band pass filter,
the first stage of demodulation gets completed. The
base band signal which is band limited is obtained
and this signal is used to regenerate the binary
message bit stream.
In the next stage of demodulation, the bit clock rate
is needed at the detector circuit to produce the
original binary message signal. If the bit rate is a
sub-multiple of the carrier frequency, then the bit
clock regeneration is simplified. To make the circuit
easily understandable, a decision-making circuit may
also be inserted at the 2nd stage of detection.
55

56. Quadrature Phase Shift Keying
• The Quadrature Phase Shift Keying (QPSK) is a variation of BPSK,
and it is also a Double Side Band Suppressed Carrier (DSBSC)
modulation scheme, which sends two bits of digital information at a
time, called as bigits.
• Instead of the conversion of digital bits into a series of digital stream,
it converts them into bit pairs. This decreases the data bit rate to half,
which allows space for the other users.
56

57. QPSK Modulator
• The QPSK Modulator uses a bit-splitter, two
multipliers with local oscillator, a 2-bit serial
to parallel converter, and a summer circuit.
Following is the block diagram for the same.
At the modulator’s input, the message
signal’s even bits (i.e., 2nd bit, 4th bit, 6th bit,
etc.) and odd bits (i.e., 1st bit, 3rd bit, 5th bit,
etc.) are separated by the bits splitter and
are multiplied with the same carrier to
generate odd BPSK (called as PSKI) and
even BPSK (called as PSKQ). The PSKQ
signal is anyhow phase shifted by 90° before
being modulated.
The QPSK waveform for two-bits input is as
follows, which shows the modulated result
for different instances of binary inputs.
57

58. QPSK Demodulator
• The QPSK Demodulator uses two
product demodulator circuits with
local oscillator, two band pass filters,
two integrator circuits, and a 2-bit
parallel to serial converter. Following
is the diagram for the same.
• The two product detectors at the input
of demodulator simultaneously
demodulate the two BPSK signals.
The pair of bits are recovered here
from the original data. These signals
after processing, are passed to the
parallel to serial converter.
58

59. Differential Phase Shift Keying
• n Differential Phase Shift Keying (DPSK) the
phase of the modulated signal is shifted relative to
the previous signal element. No reference signal
is considered here. The signal phase follows the
high or low state of the previous element. This
DPSK technique doesn’t need a reference
oscillator.
• The following figure represents the model
waveform of DPSK.
• It is seen from the above figure that, if the data bit
is Low i.e., 0, then the phase of the signal is not
reversed, but continued as it was. If the data is a
High i.e., 1, then the phase of the signal is
reversed, as with NRZI, invert on 1 (a form of
differential encoding).
• If we observe the above waveform, we can say
that the High state represents an M in the
modulating signal and the Low state represents a
W in the modulating signal.
59

60. DPSK Modulator
• DPSK is a technique of BPSK, in which
there is no reference phase signal. Here,
the transmitted signal itself can be used as
a reference signal. Following is the
diagram of DPSK Modulator.
• DPSK encodes two distinct signals, i.e., the
carrier and the modulating signal with 180°
phase shift each. The serial data input is
given to the XNOR gate and the output is
again fed back to the other input through 1-
bit delay. The output of the XNOR gate
along with the carrier signal is given to the
balance modulator, to produce the DPSK
modulated signal.
60

61. DPSK Demodulator
• In DPSK demodulator, the phase of the
reversed bit is compared with the phase of
the previous bit. Following is the block
diagram of DPSK demodulator.
• From the above figure, it is evident that the
balance modulator is given the DPSK signal
along with 1-bit delay input. That signal is
made to confine to lower frequencies with
the help of LPF. Then it is passed to a
shaper circuit, which is a comparator or a
Schmitt trigger circuit, to recover the original
binary data as the output.
61

62. Quadrature Amplitude Modulation
• PSK is limited by the ability of the equipment
to distinguish between small differences in
phases.
 Limits the potential data rate.
• Quadrature amplitude modulation is a
combination of ASK and PSK so that a
maximum contrast between each signal unit
(bit, dibit, tribit, and so on) is achieved.
 We can have x variations in phase and y
variations of amplitude
 x • y possible variation (greater data rates)
• Numerous variations. (4-QAM, 8-QAM)
62
# of phase shifts > # of amplitude shifts

63. MSK Modulation
he problem can be overcome in part by filtering the signal,
but is found that the transitions in the data become
progressively less sharp as the level of filtering is increased
and the bandwidth reduced. To overcome this problem
GMSK is often used and this is based on Minimum Shift
Keying, MSK modulation. The advantage of which is what
is known as a continuous phase scheme. Here there are no
phase discontinuities because the frequency changes
occur at the carrier zero crossing points.
When looking at a plot of a signal using MSK modulation, it
can be seen that the modulating data signal changes the
frequency of the signal and there are no phase
discontinuities. This arises as a result of the unique factor
of MSK that the frequency difference between the logical
one and logical zero states is always equal to half the data
rate. This can be expressed in terms of the modulation
index, and it is always equal to 0.5.
63

64. GMSK Modulation - Gaussian Minimum Shift Keying
There are two main ways in which GMSK modulation can
be generated. The most obvious way is to filter the
modulating signal using a Gaussian filter and then apply this
to a frequency modulator where the modulation index is set
to 0.5. This method is very simple and straightforward but it
has the drawback that the modulation index must exactly
equal 0.5. In practice this analogue method is not suitable
because component tolerances drift and cannot be set
exactly.
64

65. There are two main ways in which
GMSK modulation can be generated.
The most obvious way is to filter the
modulating signal using a Gaussian filter
and then apply this to a frequency
modulator where the modulation index is
set to 0.5. This method is very simple
and straightforward but it has the
drawback that the modulation index must
exactly equal 0.5. In practice this
analogue method is not suitable because
component tolerances drift and cannot
be set exactly.
65

66. 8-QAM and 16-QAM
First example handles noise best
Because of ratio of phases to amplitudes
ITU-T recommendation.
Second example, recommendation of OSI.
not all possibilities are used, to increase
readability of signal, measurable differences
between shifts are increased

67. References
Book:
1. Taub & Schiling “Principles of Communication Systems” Tata McGraw hill 2007.
2. Kennedy and Davis “Electronic Communication Systems” Tata McGraw hill, 4th Edition, 1993.
3. Sklar “Digital Communication Fundamentals and Applications“ Pearson Education, 2001.
4. TG Thomas and S Chandra Sekhar, “Communication Theory” Tata McGraw hill 2006.
Web:
https://www.tutorialspoint.com/analog_communication/analog_communication_pulse_modulation.htm
http://www.rfwireless-world.com/Terminology/MSK-GMSK.html
http://www.radio-electronics.com/info/rf-technology-design/pm-phase-modulation/what-is-gmsk-gaussian-m
https://www.tutorialspoint.com/digital_communication/digital_communication_techniques.htm
PPT:
 www.ics.uci.edu/~magda/Courses/netsys270/ch4_2_v1.ppt
67

68. Other presentations
http://www.slideshare.net/drgst/presentations
68Dr Gnanasekaran Thangavel12/12/2017

69. 69
Thank You
Questions and Comments?