EEN 07 FUNDAMENTALS OF

ELECTRONIC COMMUNICATIONS
BSEE II GI
BSEE II GJ

ENGR. JOEL ANTHONY L. SEVILLA

NOVEMBER 2023
Introduction to Digital Communications
Systems
Digital Transmission
Transmittal of digital pulses between 2
or more points in a communication
system
Digital Radio
Transmittal of digitally modulated
analog radio frequency
Digital Transmission
Transmittal of digital pulses between two
points in a communication system. The
original source information may be digital
form or it may be analog signals. That must
be converted to digital pulses prior to
transmission and converted back to analog
form at the receive end.
Classification based on analog or digital
communication
Advantages of Analog Communication
•Transmitters and receivers are simple
•Low bandwidth requirements
•FDM can be used
Disadvantages of Analog Communication

•Noise affects the signal quality

•Not possible to separate noise & signal
•Repeaters cannot be used between
transmitters and receivers
•Coding is not possible
•Not suitable for transmission of secret
information
Advantages of Digital over Analog
1. Easier to multiplex
2. Easier to integrate into a switching system
3. Easier to interface with other digital
equipment
4. Noise immunity
Advantages of Digital over Analog
5. Better performance monitor ability
6. Easy to encode, decode, encrypt and
scramble
Disadvantages of Digital over Analog

1. Large bandwidth
2. Need for synchronization
3. Need for additional equipment
4. Not compatible with existing system
5. Need for DA/AD conversion
6. Restriction in wired topology
PULSE MODULATION

Includes many different methods of

converting information into pulse form for
transferring pulses from a source to a
destination.
4 TYPES OF PULSE MODULATION

1. PWM - Pulse Width Modulation

2. PPM - Pulse Position Modulation
3. PAM - Pulse Amplitude Modulation
4. PCM - Pulse Code Modulation
PWM - Pulse Width Modulation
this method is sometimes called pulse
duration modulation (PDM) or pulse length
modulation (PLM). The pulse width (active
portion of the duty cycle) is proportional to
the amplitude of the analog signal.
PPM - Pulse Position Modulation
The position of a constant width pulse within
a prescribed time slot is varied according to
the amplitude of the analog signal.
PAM - Pulse Amplitude Modulation
The amplitude of a constant width, constant
position pulse is varied according to the
amplitude of the analog signal.
PCM - Pulse Code Modulation
The analog signal is sampled and converted
to a fixed length serial binary number for
transmission. The binary number varies
according to the amplitude of the analog
signal.
Analog signal

Sample pulse

PWM
Analog signal

Sample pulse

PPM
Analog signal

Sample pulse

PAM
Analog signal

Sample pulse

PCM
PAM is used as an intermediate form of
modulation with PSK, QAM, and PCM. PWM
and PPM are used in special purpose
communications systems (military) and are
seldom used for commercial systems. PCM is
the most used method of pulse modulation.
PCM is the only one of the digitally encoded
pulse modulation techniques that is used in a
digital transmission system.
Digital Modulation System

Modulation system or technique in which

the transmitted signal is in the form of
digital pulses of constant amplitude,
constant frequency and phase.
PCM and DM

A train of digital pulses is transmitted by

the transmitter. All the pulses are
constant amplitude, width and position.
the information is contained in the
combination of the transmitted pulses.
Information Capacity Theorem
Shannon Hartley Theorem
Channel bandwidth and signal to noise ratio
are exchangeable in the sense that we may
trade off one for the other for a prescribed
system performance.
Information Capacity Theorem
Shannon Hartley Theorem
Let B denote the channel bandwidth and SNR
denote the received signal to noise ratio. the
information capacity theorem states that
ideally these two parameters are related by
C = B log2 (1 + SNR) b/s
where C is the information capacity
Alphanumeric Codes
ASCII Code
ASCII is a 7-bit code used for
representing alphanumeric symbols with a
distinctive code word. The ASCII code was
developed by a committee of the American
National Standards Institute (ANSI) for the
purpose of coding binary data. ASCII-77 is the
adopted international standard.
ASCII Code
There are 2⁷ (128) possible 7-bit code
words available with an ASCII system. The
binary codes are ordered sequentially, which
simplifies the grouping and sorting of the
characters. The 7-bit words are ordered with
the least significant bit (LSB) given as bit 1
(b1), while the most significant bit (MSB) is
bit 7 (b7) .
ASCII Code
Parity a common method of error detection,
adding an extra bit to each code
representation to give the word either an
even or odd number of 1s.
In some systems the actual transmission of
these codes includes an extra pulse at the
beginning (start) and ending (stop) for each
character.
When start/stop pulses are used in the
coding of signals, it is called an asynchronous
(nonsynchronous) transmission.
A synchronous transmission (without
start/stop pulses) allows more characters to
be transmitted within a given sequence of
bits.
EBCDIC
Extended Binary-Coded Decimal Interchange
Code (EBCDIC) is an 8-bit alphanumeric code.
The term binary-coded decimal is used in the
name because of the structure present in the
coding scheme, which uses only the 0- 9
positions.
EBCDIC
Code
Baudot Code
The Baudot code was developed in the
days of teletype machines such as the ASR-
33 Teletype terminal. A 5-bit code can have
only 2⁵ or 32 bits of information but
Baudot code actually provides 26 X 2 bits by
transmitting a 11111 to indicate all
following items are "letters“ until a 11011
transmission occurs, indicating "figures."
BAUDOT
Code
PULSE CODE MODULATION
Pulse-code modulation (PCM) is the most
common technique used today in digital
communications for representing an analog
signal by a digital word.
PCM applications:
•telephone system
•digital audio recording
•DAT or digital audio tape
•Laser discs
•digitized video special effects
•voice mail
•digital video
PCM techniques and applications are a
primary building block for many of today's
advanced communications systems.

PCM is a technique for converting the analog

signals into a digital representation. The PCM
architecture consists of a sample-and-hold
(S/H) circuit and a system for converting the
sampled signal into a representative binary
format.
The analog signal is input into a sample-and-
hold circuit. At fixed time intervals, the analog
signal is sampled and held at a fixed voltage
level until the circuitry inside the AD
converter has time to complete the
conversion process of generating a binary
value.
PCM Block diagram
Sample and Hold Circuit
A circuit that is capable of sampling the input
signal applied to its terminal as well as
holding the sampled value up to the last
sample for a particular time interval.
Sample and Hold Circuit
The time required for an S/H circuit to
complete a sample is based partly on the
acquisition and aperture time.
The acquisition time is the amount of time it
takes for the hold circuit to reach its final
value.
Sample and Hold Circuit
The acquisition time is controlled by the
sample pulse. The aperture time is the time
that the S/H circuit must hold the sampled
voltage. The aperture and acquisition times
limit the maximum input frequency that the
S/H circuit can accurately process.
PAM generation
Two basic sampling techniques used to create
a PAM signal.
Natural Sampling - sampling in which the
tops of the sampled waveforms retain their
natural shape.
Flat-Top Sampling - sampling in which the
signal voltage is held constant during
samples, creating a staircase that tracks the
changing input waveform.
Natural Sampling Flat Top Sampling
The sample frequency is governed by the
Nyquist rate.
The Nyquist rate states that the sample
frequency (fs) must be at least twice the
highest input frequency (fa)·
fs ≥ 2fa
Aliasing or Foldover Distortion
the phenomenon associated with the
generation of error signals in the sampling
process

Antialiasing Filter
a filter that band limits the input frequencies
to 1/2 the sampling frequency so that foldover
distortion, or aliasing, is prevented.
Quantization
Process of segmenting a sampled signal in a
PCM system into different voltage levels, each
level corresponding to a different binary
number.
The quantization levels also determine the
resolution of the digitizing system.
Analog signals are quantized to the closest
binary value provided in the digitizing system.
This is an approximation process. If the set of
whole numbers are 1, 2, 3, ... , and the
number 1.4 must be converted (rounded off)
to the closest whole number, then 1.4 is
translated to 1. If the input number is 1.6,
then the number is translated to a 2. If the
number is 1.5, then we have the same error if
the number is rounded off to a 1 or 2.
Voltage levels for a quantized signal

the closest quantization level is selected for representing the

sine-wave signal. The resulting waveform has poor resolution
with respect to the sine-wave input.
COMPANDING
Is the process of compressing the expanding
companded systems – higher amplitude
signals are compressed prior to transmission,
then expanded at the receiver.

Analog Companding
In the transmitter, the analog signal is compressed,
sampled then converted to a linear PCM code. In
the receiver the PCM code is converted to a PCM
signal, filtered then expanded back to its original
input characteristics.
Amplitude Companding
process of volume compression before
transmission and volume expansion after
detection

Companding Process
Digital Companding
Involves compression at the transmit and after
the input sample has been converted to a linear
PCM code and expansion at the receive end prior
to PCM decoding
The analog signal is first sampled and converted
to a linear code, then the linear code is digitally
compressed. At the receive end the compressed
PCM code is received, expanded, then decoded.
Vocoders (Voice encoders/decoders)
Used when digitizing speech signals only.
Produces unnatural sounding speech and are
generally used for recording informations,
such as “wrong number” messages,
encrypted voice for transmission over analog
telephone circuits, computer output signals
and educational games.
Vocoders (Voice encoders/decoders)
The purpose of vocoders is to encode the
minimum amount of speech information
necessary to reproduce a perceptible
message with fewer bits than those needed
by a conventional encoder/decoder.
Application – used in limited bandwidth
PCM Communication system
Analog to Digital Converter (ADC)
is used to convert the information signal
to a digital format.
Digital to Analog Converter (DAC)
is to convert a digital (binary) bit stream
to an analog signal.
Serial Transmission
bits are transmitted one at a time
Parallel Transmission
bits are transmitted at the same time
Parallel to Serial Converter
converts parallel PCM into serial data
Serial to Parallel Converter
serial data is converted prior
transmission to parallel PCM
 EEN 07 FUNDAMENTALS OF
ELECTRONIC COMMUNICATIONS
BSEE II GI
BSEE II GJ

ENGR. JOEL ANTHONY L. SEVILLA

DECEMBER 2023
ERROR DETECTION AND CORRECTION

Error Detection is the process of

monitoring data transmission and
determining when errors have occurred.
Error detection techniques neither
correct errors nor identify which bits are in
error — they indicate only when an error has
occurred.
 The purpose of error detection is not to
prevent errors from occurring but to prevent
undetected errors from occurring.
The most common error‐detection
techniques are redundancy checking, which
includes vertical redundancy checking,
checksum, longitudinal redundancy checking,
and cyclic redundancy checking.
Redundancy Checking
Adding bits for the sole purpose of
detecting errors.
Duplicating each data unit for the
purpose of detecting errors is a form of error
detection.
It is an effective but rather costly means
of detecting errors, especially with long
messages.
Redundancy Checking
Redundancy involves transmitting each
character twice. If the same character is not
received twice in succession, a transmission
error has occurred.
Vertical Redundancy Checking (VRC)
It is the simplest error‐detection scheme
and is generally referred to as character
parity or parity. With character parity, each
character has its own error‐detection bit
called the parity bit. Since the parity bit is not
actually part of the character, it is considered
a redundant bit. An n‐character message
would have n redundant parity bits.
Vertical Redundancy Checking (VRC)
The number of error‐detection bits is
directly proportional to the length of the
message.
With character parity, a single parity bit
is added to each character to force the total
number of logic 1s in the character, including
the parity bit, to be either an odd number
(odd parity) or an even number (even parity).
ASCII‐77: Odd Parity
Checksum
Is another simple form of redundancy
error checking where each character has a
numerical value assigned to it. The
characters within a message are combined
together to produce an error‐checking
character (checksum), which can be the
arithmetic sum of the numerical values of all
the characters in the message.
Checksum
The checksum is appended to the end of
the message. The receiver replicates the
combining operation and determines its own
checksum. The receiver’s checksum is
compared to the checksum appended to the
message, and if they are the same, it is
assumed that no transmission errors have
occurred. If the two checksums are different,
a transmission error has definitely occurred.
Longitudinal Redundancy Checking (LRC)
Is a redundancy error detection scheme
that uses parity to determine if a
transmission error has occurred within a
message and is therefore sometimes called
message parity.
With LRC, each bit position has a parity bit. In
other words, b0 from each character in the
message is XORed with b0 from all the other
Longitudinal Redundancy Checking (LRC)
characters in the message. Similarly, b1, b2,
and so on are XORed with their respective
bits from all the characters in the message.
Essentially, LRC is the result of XORing the
“character codes” that make up the message,
whereas VRC is the XORing of the bits within
a single character. With LRC, even parity is
generally used, whereas with VRC, odd parity
is generally used.
Cyclic Redundancy Checking (CRC)
Most reliable redundancy checking
technique for error detection. With CRC,
approximately 99.999% of all transmission
errors are detected.
In the United States, the most common
CRC code is CRC‐16. 16 bits are used for the
block check sequence.
Cyclic Redundancy Checking (CRC)
With CRC, the entire data stream is
treated as a long continuous binary number.
Because the BCS is separate from the
message but transported within the same
transmission, CRC is considered a systematic
code.
ERROR CORRECTION
Two primary methods used for error
correction:
Retransmission
Forward Error Correction
Retransmission
When a receive station requests the
transmit station to resend a message (or a
portion of a message) when the message is
received in error, because the receive
terminal automatically calls for a
retransmission of the entire message.
Retransmission
Retransmission is often called ARQ,
which is two‐way radio term that means
automatic repeat request or automatic
retransmission request. ARQ is probably the
most reliable method of error correction,
although it is not necessarily the most
efficient.
Forward Error Correction (FEC)
Is the only error‐correction scheme that
actually detects and corrects transmission
errors when they are received without
requiring a retransmission. Redundant bits
are added to the message before
transmission. When an error is detected, the
redundant bits are used to determine which
bit is in error.
Hamming Code
The Hamming code is an error‐correcting
code used for correcting transmission errors
in synchronous data streams. The Hamming
code will correct only single‐bit errors. It
cannot correct multiple‐bit errors or burst
errors, and it cannot identify errors that
occur in the Hamming bits themselves.
Hamming Code
The Hamming code, as with all FEC
codes, requires the addition of overhead to
the message, consequently increasing the
length of a transmission.
Hamming bits (sometimes called error
bits) are inserted into a character at random
locations. The combination of the data bits
Hamming Code
and the Hamming bits is called the Hamming
code. The only stipulation on the placement
of the Hamming bits is that both the sender
and the receiver must agree on where they
are placed.
To calculate the number of redundant
Hamming bits necessary for a given character
length, a relationship between the character
Hamming Code
bits and the Hamming bits must be
established. The total number of bits in one
data unit is m + n. Since the Hamming bits
must be able to identify which bit is in error,
n Hamming bits must be able to indicate at
least m + n + 1 different codes. Of the m + n
codes, one code indicates that no errors have
occurred, and the remaining m + n codes
Hamming Code
indicate the bit position where an error has
occurred. Therefore, m + n bit positions must
be identified with n bits. Since n bits can
produce ௡ different codes, ௡ must be equal
to or greater than m + n + 1.
Therefore, the number of Hamming bits is
determined by the following expression:
Hamming Code
≥m+n+1
where n number of Hamming bits
m number of bits in each data character
Example:
How many Hamming bits would be added to
a data block containing 128 bits?
Example
How many Hamming bits would be added to
a data block containing 128 bits?
Solution
Number of Hamming bits
≥m+n+1
for n = 6
= 64 ; m + n + 1 = 128 + 6 + 1 < 135
Example
for n = 7
= 128; m + n + 1 = 128 + 7 + 1 < 136

for n = 8
= 256; m + n + 1 = 128 + 8 + 1 > 137

Hamming bits = 8
 EEN 07 FUNDAMENTALS OF
ELECTRONIC COMMUNICATIONS
BSEE II GI
BSEE II GJ

ENGR. JOEL ANTHONY L. SEVILLA

JANUARY 2024
Wireless Digital Communications
The transport of digital data over a wireless
medium

Wireless technologies can be divided into the

following three groups:
1. Fixed wireless: both transmit and receive
units are at fixed locations (wireless local
area networks / wLANs).
2. Mobile wireless: the transmit and/or receive
units are moving (cellular & mobile phones).

3. Infrared (IR) wireless: the transmit and

receive units use infrared light sources and
detectors to provide the communications
link (building to building communications
links)
Frequency Shift Keying (FSK)
is a form of frequency modulation in which the
modulating wave shifts the output between two
predetermined frequencies –
usually termed the
mark and space
frequencies.
Frequency Shift Keying (FSK)
It may be considered as an FM system in which the
carrier frequency is midway between the mark
and space frequencies and is modulated by a
rectangular wave. The mark condition causes the
carrier frequency to increase by 42.5 Hz, while the
space condition results in a 42.5 Hz downward
shift. The transmitter frequency is constantly
changing by 85 Hz as it is keyed.
Frequency Shift Keying (FSK)
This 85 Hz shift is the standard for narrowband
FSK, while an 850 Hz shift is the standard for
wideband FSK systems.

For FSK the center or carrier frequency is shifted

(deviated by binary input signal)
FSK Modulated waveform
Frequency Shift Keying (FSK)

FSK frequencies
1. Mark Frequency or logic 1 frequency (fm)
2. Space Frequency or logic 0 frequency (fs)
3. Rest Frequency – falls hallway between space
& mark frequencies
Frequency Shift Keying (FSK)

Bit Rate (fb)

The rate of change at the input to the modulator

Baud Rate (fbd)

The rate of change at the output of the modulator
Frequency Shift Keying (FSK)

Fundamental Frequency (fa)

The highest modulating frequency equal to ½ of
input bit rate.

𝑓𝑏
fa =
2
Frequency Shift Keying (FSK)

Modulation Index (MI)

∆𝑓 𝑓𝑚 −𝑓𝑠 /2 𝑓𝑚 −𝑓𝑠
MI = = =
𝑓𝑎 𝑓𝑏/2 𝑓𝑏

In binary FSK the MI is generally kept below 1.0

producing narrow band FM.
PLL FSK DEMODULATOR

Non continuous FSK waveform

Amplitude Shift Keying (ASK)
Is a double sideband, full carrier, amplitude
modulation where the input modulating signal is
a binary waveform.

Digital bit sequence

Carrier wave

ASK modulated wave

Amplitude Shift Keying (ASK)
The amplitude of the resultant output depends
upon the input data whether it should be a zero
level or a variation of positive and negative,
depending upon the carrier frequency.

ASK is a type of Amplitude Modulation which

represents the binary data in the form of
variations in the amplitude of a signal.
Amplitude Shift Keying (ASK)
Any modulated signal has a high frequency
carrier. The binary signal when ASK is
modulated, gives a zero value for LOW input and
gives the carrier output for HIGH input.

Also know as continuous wave modulation and

on off keying (OOK)
Amplitude Shift Keying (ASK)
ASK waveform (baud) is the same as the rate of
change of the binary input (bps)
The bit rate equals the baud. With ASK, the bit
rate is also equal to the minimum Nyquist
bandwidth.

𝑓𝑏 𝑓𝑏
B= = fb baud = = fb
1 1
Phase Shift Keying (PSK)
Is a form of angle modulated, constant
amplitude digital modulation similar to
conventional phase modulation except that
with PSK the input is a binary digital signal and
limited number of output phase are available.
The phase of the output signal gets shifted
depending upon the input. These are mainly
of two types, namely BPSK and QPSK,
according to the number of phase shifts.
Phase Shift Keying (PSK)
The other one is DPSK which changes the
phase according to the previous value.
PSK is the digital modulation technique in
which the phase of the carrier signal is
changed by varying the sine and cosine inputs
at a particular time. PSK technique is widely
used for wireless LANs, bio-metric, contactless
operations, along with RFID and Bluetooth
communications.
Binary Phase Shift Keying (BPSK)
Two output phases are possible for a single
carrier frequency.
This is also called as 2-phase PSK (or) Phase
Reversal Keying. In this technique, the sine wave
carrier takes two phase reversals such as 0° and
180°.
BPSK is basically a DSB-SC (Double Sideband
Suppressed Carrier) modulation scheme, for
messages being the digital information.
BPSK modulated output wave
BPSK phasor and constellation diagram
Quadrature Phase Shift Keying (QPSK)
This is the phase shift keying technique, in
which the sine wave carrier takes four phase
reversals such as 0°, 90°, 180°, and 270°.
If this kind of techniques are further extended,
PSK can be done by eight or sixteen values also,
depending upon the requirement. The following
figure represents the QPSK waveform for two
bits input, which shows the modulated result for
different instances of binary inputs.
QPSK MODULATOR
QPSK modulated output wave
QPSK phasor and constellation diagram
QPSK is a variation of BPSK, and it is also a
DSB-SC (Double Sideband Suppressed Carrier)
modulation scheme, which send two bits of
digital information at a time, called as dibits.
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.
Differential Phase Shift Keying (DPSK)
In DPSK (Differential Phase Shift Keying) the
phase of the modulated signal is shifted
relative to the previous signal element. No
reference signal is considered here. The signal
phase follows the high or low state of the
previous element. This DPSK technique
doesn’t need a reference oscillator.
DPSK modulated output wave
Minimum Shift Keying (MSK)
The mark and space frequency are selected such
that they are separated from the center
frequency by an odd exact multiple of one half
of the bit rate.

MSK signal
Gaussian Minimum Shift Keying (GMSK)
It is a form of modulation based on frequency
shift keying that has no phase discontinuities
and provides efficient use of spectrum as well as
enabling high efficiency radio power amplifiers.
The tone frequencies are separated by exactly
half the bit rate. It has high spectral efficiency.
Offset QPSK (OQPSK)
It is a modified form of QPSK where the bit
waveforms on the I and Q channels are offset or
shifted in phase from each other by one half of a
bit time.
Changes in the I channel occur at the midpoints
of the Q channel bits and vice versa, there is
never more than a single bit change in the dibit
code and there is never more than a 90° shift in
the output phase.
Offset QPSK (OQPSK)
In conventional QPSK, a change in the input dibit
from 00 to 11 or 01 to 10 causes a
corresponding 180° shift in the output phase.
An advantage of OQPSK is the limited phase shift
that must be imparted during modulation.
A disadvantage of OQPSK is that changes in the
output phase occur at twice the data rate in
either the I or Q channels.
Offset QPSK (OQPSK)
With OQPSK the baud and minimum bandwidth
are twice that of conventional QPSK for a given
transmission bit rate. OQPSK is sometimes called
Offset-Keyed QPSK.
OQPSK transmitter block diagram

Block diagram Constellation diagram

8 Phase Shift Keying (8 PSK)
Three bits are encoded, forming tribits and
producing eight different output phases.
With 8-PSK, n = 3, M = 8, and there are eight
possible output phases. To encode eight
different phases, the incoming bits are encoded
in groups of three, called tribits.
8 PSK transmitter block diagram
The incoming serial bit stream enters the bit
splitter, where it is converted to a parallel, three-
channel output (the I or in-phase channel, the Q
or in-quadrature channel, and the C or control
channel).
The bit rate in each of the three channels is fb/3.
The bits in the I and C channels enter the I
channel 2-to-4-level converter, and the bits in
the Q and channels enter the Q channel 2-to-4-
level converter.
The 2-to-4-level converters are parallel-input
digital-to-analog converters (DACs). The I or Q
bit determines the polarity of the output analog
signal (logic 1 V and logic 0 V), whereas the C or
C̅ bit determines the magnitude (logic 1 = 1.307 V
and logic 0 = 0.541 V). With two magnitudes and two
polarities, four different output conditions are
possible.
 a b

c d

I- and Q- channel 2-to-4-level converters: (a) I-channel truth

table; (b) Q-channel truth table; (c) PAM levels (d) 8 PSK
truth table
8 PSK Phasor & Constellation Diagram

Phasor diagram Constellation diagram

The angular separation between any two
adjacent phasors is 45°, half what it is with
QPSK. An 8-PSK signal can undergo almost a
22.5° phase shift during transmission and still
retain its integrity.
Each phasor is of equal magnitude; the tribit
condition (actual information) is again contained
only in the phase of the signal.
The PAM levels of 1.307 and 0.541 are relative
values. Any levels may be used as long as their
ratio is 0.541/1.307 and their arc tangent is
equal to 22.5°.
For example, if their values were doubled to
2.614 and 1.082, the resulting phase angles
would not change, although the magnitude of
the phasor would increase proportionally.
The tribit code between any two adjacent
phases changes by only one bit. This type of
code is called the Gray code or, sometimes, the
maximum distance code. This code is used to
reduce the number of transmission errors.
If a signal were to undergo a phase shift during
transmission, it would most likely be shifted to
an adjacent phasor. Using the Gray code results
in only a single bit being received in error.
Output phase-versus-time relationship for an 8-PSK modulator
16-PSK
Is an M-ary encoding technique where M = 16;
there are 16 different output phases possible.
With 16-PSK, four bits (called quadbits) are
combined, producing 16 different output
phases. With 16-PSK, n = 4 and M = 16;
therefore, the minimum bandwidth and baud
equal one-fourth the bit rate ( fb/4).
16-PSK truth table and constellation diagram

16 PSK truth table Constellation diagram

16-PSK
With 16-PSK, the angular separation between
adjacent output phases is only 22.5°. Therefore,
16-PSK can undergo only a 11.25° phase shift
during transmission and still retain its integrity.
QUADRATURE-AMPLITUDE MODULATION
Is a form of digital modulation similar to PSK
except the digital information is contained in
both the amplitude and the phase of the
transmitted carrier.
With QAM, amplitude and phase-shift keying are
combined in such a way that the positions of the
signaling elements on the constellation diagrams
are optimized to achieve the greatest distance
QUADRATURE-AMPLITUDE MODULATION
between elements, reducing the likelihood of
one element being misinterpreted as another
element. This reduces the likelihood of errors
occurring.
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.
The only difference between the 8-QAM
transmitter and the 8-PSK transmitter as shown
in the figure is the omission of the inverter
between the C channel and the Q product
modulator.
8 QAM transmitter block diagram & truth table
The incoming data are divided into groups of three
bits (tribits): the I, Q, and C bit streams, each with a
bit rate equal to one-third of the incoming data rate.
Again, the I and Q bits determine the polarity of the
PAM signal at the output of the 2-to-4-level
converters, and the C channel determines the
magnitude. Because the C bit is fed as is to both the
I and the Q channel 2-to-4-level converters, the
magnitudes of the I and Q PAM signals are always
equal. Their polarities depend on the logic condition
of the I and Q bits and, therefore, may be different.
In 8-QAM, the bit rate in the I and Q channels is
one-third of the input binary rate, the same as in
8-PSK.
The highest fundamental modulating frequency
and fastest output rate of change in 8-QAM are
the same as with 8-PSK.
The minimum bandwidth required for 8-QAM is
fb/3, the same as in 8-PSK.
8 QAM Phasor & Constellation Diagram

Phasor diagram Constellation diagram

8 QAM truth table Output phase-versus-time
relationship for an 8-PSK modulator
16-QAM
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). Four bits are serially clocked into the
bit splitter; then they are outputted
simultaneously and in parallel with the I, I’, Q,
and Q’ channels. The I and Q bits determine the
polarity at the output of the 2-to-4 level
16-QAM
converters (a logic 1 = positive and a logic 0 =
negative). The I’ and Q’ bits determine the
magnitude (a logic I 0.821 V and a logic 0 0.22
V). Consequently, the 2-to-4-level converters
generate a 4-level PAM signal. Two polarities and
two magnitudes are possible at the output of
each 2-to-4-level converter. They are 0.22 V and
0.821 V.
16-QAM Transmitter block diagram
 a b

c
16-QAM truth tables for the I- and Q-channel 2-to-4 level
converters: (a) I channel; (b) Q channel (c) truth table
16 QAM Phasor & Constellation Diagram

Phasor diagram Constellation diagram