ECE 421/L Digital

Communication
By: Engr. Rolieven P. Cañizares
Course Outline
1. Introduction to Digital Communication
2. Digital Transmission
3. Pulse Code Modulation
4. Basics of Information Theory
5. Error Detection and Error Correction Schemes
6. Digital Systems Multiplexing
Textbooks
1. Electronic Communication Systems by
Blake
2. Electronic Communication System by
Frenzel
3. Communication System by Carlson
4. Electronic Communication by Wayne
Tomasi
5. Modern Electronic Communication by
Garry Miller
∆
Introduction to Digital
Communication
• Electronic communication is the
transmission, reception, and processing of
information with the use of electronic
circuits
• Simplified block diagram (please see the
figure on the board)
Introduction to Digital
Communication
• Covers a broad area of communication
techniques;
1.Digital Transmission
2.Digital Radio (Digital Modulation)
Types of Information Signal
1. Analog Signal (examples: human voice,
video picture, music, etc.)
2. Digital Signal (examples: binary coded
numbers, alphanumeric codes, graphic
symbols, microprocessor op-codes, data
base information, etc.)
• The source of information is unsuitable for
transmission in its original form and must
be converted to a more suitable form prior
to transmission
• In digital communication analog signal is
converted to digital. In analog
communication, digital information must be
converted to analog
• Analog systems (AM, FM, PM) are rapidly
replaced by modern digital
communications
Digital Transmission
• Transmittal of digital pulses between two
or more points in a communication system
• It requires physical facility between
transmitter and receiver
a) Metallic wire pair
b) Coaxial cable
c) Optical fiber cable
Digital Transmission
Digital Radio (Digital
Modulation)
• Transmittal of digitally modulated analog
carriers between two or more points in
communication systems
Digital Radio (Digital
Modulation)
Advantages of Digital Systems
over Analog Systems
1. Ease of processing
2. Ease of multiplexing
3. Noise immunity
• Digital signals have become very
important in wired and wireless
communications
• To send data by radio, higher frequency
carrier waves are necessary
• Amplitude, frequency, and phase are all
used in digital communication systems
•
Information Capacity (I)
• It represents the number of symbols that
can be carried through the system in a
given unit of time
• Most basic symbol is the binary digit (bit)
• It is expressed in bits per second (bps)
• It commonly denoted as transmission rate
Hartley’s Law
• Describes the relationship between
bandwidth, transmission time, and
information capacity

IαBT
I = kTB
Where:
I = Information Capacity (bps)
B = Bandwidth (Hz)
T = Transmission Time (s)
S/N = signal-to-noise power ratio (unitless)
Shannon Limit

S
I = B log 2 (1 + )
N

S
I = 3.32 B log10 (1 + )
N
Shannon-Heartley Theorem

I = 2 B log 2 M

I = 2B
Digital Modulation
Digital Modulation
• Digital and analog radio systems both use
analog carriers
• Analog modulation – modulating signal is
analog
• Digital Modulation – modulating signal is
digital
Types of Digital Modulation
1. ASK
2. FSK
3. PSK
4. QAM
Bit Rate and Baud Rate
Bit Rate
• Rate of change at the input to the
modulator (bits per second or bps)

Baud Rate
• Rate of change at the output of the
modulator (bps or symbols per second
(sps))
Digital Amplitude Modulation
• It is double-sideband, full carrier ampltude
modulation
• The input modulating signal is binary
• It is also called OOK (On and Off Keying)
• Another name is CW (Continuous Wave)

A
vam (t ) = [1 + vm (t )][ cos(ω c t )]
2
Frequency Shift Keying (FSK)
Keying
• It is the on and off of the carrier
• It is a low performance type of digital
modulation
• The modulating signal is binary signal that
varies between two discrete voltage levels
• Frequency of the carrier is varied by the
modulating signal
Types of FSK
1. Binary FSK
2. Minimum Shift Keying (MSK)
3. Gaussian Minimum Shift Keying (GMSK)
Binary FSK
• vfsk(t) = binary FSK waveform
• Vc = peak carrier amplitude (volts)
• fc = carrier center frequency
• ∆f = peak frequency deviation
• Vm(t) = binary input modulating signal

v fsk = Vc cos{2π [ f c + vm (t )∆f ]t}

Input and Output Waveforms
FSK Modulator
FSK Transmitter
• It is similar to a conventional FM
modulator and is very often voltage-
controlled oscillator (VCO)
• The carrier or rest (or center) frequency is
chosen such that it falls halfway between
the mark and space frequencies
• Logic 1 input shifts the VCO output to
mark frequency
• Logic 0 – Space frequency
FSK Transmitter
• ∆f = peak frequency deviation (Hz)
• Vm(t) = peak binary modulating-signal
voltage (volts)
• K1 = deviation sensitivity (hertz per volt)

∆f = vm (t )k1
Bandwidth Consideration of
FSK
Frequency Deviation

fm − fs fm > fs
∆f =
2

fs − fm
∆f =
2
fs >fm
Bandwidth Considerations
Minimum Bandwidth

B = 2(∆f + f b )
FSK Reciver
Noncoherent Detection
• The FSK input is simultaneously applied to
the inputs of both bandpass filters through
a power splitter
• The respective filter passes only the mark
and space frequency on to its respective
envelop detector
• The envelop detectors, in turn, indicate the
total power in each passband and the
comparator responds to the largest power
Noncoherent Detection
• There is no frequency involved in the
demodulation process that is synchronized
either in phase, frequency, or both with the
incoming signal
FSK Reciver
Coherent Detection
• The incoming FSK signal is multiplied by a
recovered carrier signal that has the exact
same frequency and phase as the
transmitter reference
• However, the two transmitted frequencies
(mark and space frequencies) are not
generally continuous
• It is not practical to reproduce a local
reference that is coherent with both of
them. This detection is seldom used
PLL Demodulator
PLL Demodulator
• Phase-locked loop (PLL) is the most
common circuit used for demodulating
FSK
• As 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
• Two frequencies (mark and space)
represents two error voltages at the output
(logic 1 or logic 0)
PLL Demodulator
• The natural frequency of PLL is made
equal to the center frequency of FSK
modulator
• As a result, changes in the dc error
voltage follow the changes in the analog
input frequency and are symmetrical
aroung 0 V.
Binary FSK Disadvantages
• It has a poorer error performance than
PSK or QAM and is seldom used for high
performance radio systems
• It use is restricted to low performance,
low-cost, asynchronous data modems
Additional Formulas
Modulating Frequency or Fundamental
Frequency (fa)

fb
fa =
2
Minimum Bandwidth Required

f N = 2nf a
Minimum Shift Keying (MSK)
• It is a binary FSK except that the mark and
space frequencies are synchronized with
the input bit rate
• It is also known as Continuous-Phase Shift
Keying (CP-FSK)
• The mark and space frequencies are
separated from the center frequency by an
odd exact multiple of one half of the bit
rate
• It ensures a smooth phase transition in the
analog output signal when it changes from
mark to space frequency or vice versa

fb
fm = n
2

fb
fs = n
2
Gaussian Minimum-Shift
Keying (GMSK)
• The word Gaussian refers to the shape of
a filter that is used before the modulator to
reduce the transmitted bandwidth of the
signal.
• GMSK uses less bandwidth than
conventional FSK, because the filter
causes the transmitted frequency to move
gradually between the mark and space
frequencies.
GMSK
• With conventional FSK the frequency
transition is theoretically instantaneous,
and in practice as rapid as the hardware
allows, producing sidebands far from the
carrier frequency.
• GMSK is particularly used in GSM cellular
radio and PCS systems.
Problems
1. Determine the bandwidth and baud for an
FSK signal with a mark frequency of 32
kHz, a space frequency of 24 kHz, and a
bit rate of 4 kbps
2. Determine the maximum bit rate for an
FSK with a mark frequency of 48 kHz, a
space frequency of 52 kHz, and an
available bandwidth of 10 kHz
Phase Shift Keying
• It is a form of angle modulated, constant
amplitude digital modulation
• It is similar to conventional phase
modulation except that with PSK the input
signal is binary
• Limited output phases are possible
Variations of PSK
• Binary PSK (BPSK)
• Quaternary PSK (QPSK)
• Eight PSK (8-PSK)
• Sixteen PSK (16-PSK)
• Differential Binary PSK
BPSK
BPSK
• Two output phases are possible for a
single carrier frequency
• One output phase represents logic 1 and
the other is logic 0
• The output shifts between two angles that
are 180 0 out of phase
• Also called phase reversal keying (PRK)
and biphase modulation
Balanced Ring Modulator
Balanced RingModulator
Balanced Ring Modulator
Constellation Diagram
Output Phase
Bandwidth Considerations of
BPSK
• fa = maximum fundamental frequency of
binary input (hertz)
• fc = reference carrier frequency (hertz)

BPSK output = [sin( 2πf a t )][sin( 2πf c t )]

1 1
BPSK output = cos[2π ( f c − f a )t ] − cos[2π ( f c + f a )t ]
2 2
M-ary Encoding
• N = number of bits encoded
• M = number of output conditions possible
with N bits
• B = minimum bandwidth (Hertz)
• fb = input bit rate (bps)
N = log 2 M

fb fb
B= B=
log 2 M N
BPSK Receiver
Problems
1. Determine the minimum bandwidth and
baud for a BPSK modulator with a carrier
frequency of 40 MHz, and an input bit
rate of 500 kbps. Sketch the spectrum
Quaternary Phase Shift Keying
(QPSK)
• It is also called Quadrature PSK
• M = 4 (Quaternary)
• The data are group in dibits
QPSK Modulator
• I Channel (In Phase Channel)
• Q Channel (Quadrature Channel)
Minimum Bandwidth Required (fN)

fb
fN = 2
4

fb
fN =
2
Output Baud Rate

fb
fN =
2
QPSK Modulator
Constellation / Phasor Diagram
Output Wave
Bandwidth Consideration
QPSK Receiver
Offset QPSK
• Modified form of QPSK
• The bit on I and Q channels are offset or
shifted in phase from each other by one
half of a bit time
Offset QPSK
Offset QPSK
Problems
1. For the QPSK modulator change the + 90
0 phase shift to – 90 0 and sketch the new

constellation diagram.
2. For the QPSK Demodulator determine
the I and Q bits for an input signal of
sin ωc t − cos ω c t
8 - PSK
• M = 8 (output phases)
• The data is tribits
8 – PSK Transmitter
• I = in phase channel
• Q = quadrature channel
• C = Control Channel
• Baud rate is fb/3
• The 2-to-4 level converters are parallel-
input digital-analog converters
2 to 4 Level Converters
Phasor Diagram
Constellation Diagram
8 – PSK Output Wave
Bandwidth Considerations
8-PSK Receiver
Problems
1. For an 8 – PSK modulator with an input
data rate (fb) equal to 20 Mbps and a
carrier frequency of 100 Mbps, determine
the minimum double-sided Nyquist
bandwidth (fN) and the baud. Sketch the
output spectrum.
2. For the 8 – PSK modulator, change the
reference oscillator to cosωc t
Sketch the new constellation diagram
Sixteen-Phase PSK
• M = 16
• The data is four bits (quad bits)
• The output phase does not change until
the four bits have been inputted into the
modulator
Truth Table / Constellation
Problems
1. For an 16– PSK modulator with an input
data rate (fb) equal to 20 Mbps and a
carrier frequency of 100 Mbps, determine
the minimum double-sided Nyquist
bandwidth (fN) and the baud. Sketch the
output spectrum.
2. For the 16 – PSK modulator, change the
reference oscillator to cosωc t
sketch the new constellation diagram
Quadrature Amplitude
Modulation (QAM)
• The digital information is contained in both
the amplitude and phase of the transmitted
carrier
Types of QAM
1.) 8-QAM
2.) 16-QAM
8-QAM
• M-ary encoding technique where M=8
• The amplitude is not constant
8-QAM Transmitter
• The inverter is omitted between C channel
and Q product modulator
• Incoming data are divided into groups of
three bits (tribits)
• I, Q, and C bit streams each with a bit rate
equal to 1/3 of the incoming bit rate
8 – QAM Transmitter
• I and Q bits determine the polarity of the
PAM signal at the output of 2-to-4 level
converters
• The C channel determines the magnitude
• Magnitudes of the I and Q PAM signals
are always equal because C bit is fed
uninverted to both the I and Q channel 2-
to-4 level converters
8-QAM Transmitter
Problem
1. For a tribit input of Q = 1, I = 1, C = 1
(111), determine the output amplitude
and phase for the 8-QAM transmitter
2. For a tribit input of Q = 1, I = 0, C = 1
(101), determine the output amplitude
and phase for the 8-QAM transmitter
3. For a tribit input of Q = 0, I = 1, C = 1
(011), determine the output amplitude
and phase for the 8-QAM transmitter
Bandwidth Considerations of 8-
QAM
• Minimum bandwidth required is the same
as that of 8-PSK (fb/3)
16-QAM
• M-ary system where M = 16
• Input are in groups of 4 (quadbits)
• Four channels I, I’, Q and Q’
• Bit rate are each (fb/4)
• Four bits are serially clocked into the bit
splitter
• Then outputted in parallel with I, I’, Q and
Q’ channels
16-QAM Transmitter
• The I and Q bits determine the polarityat
the output of 2-to-4 level converters (logic
1 = positive and logic 0 = negative)
• The I’ and Q’ bits determine the magnitude
(logic 1= 0.821V and logic 0 = 0.22V)
• The 2-to-4 level converters generate a 4-
level PAM signal
• Two polarities are possible; ±0.22V and
±0.821 V
16-QAM Transmitter
• The PAM signals modulate the in-phase
and quadrature carriers in the product
modulators
16-QAM Transmitter
Truth Table
Truth Table
Constellation Diagram
Problem
1. For a quadbit input of I = 0, I’ = 1, Q = 0,
Q’ = 1 (0101), determine the output
amplitude and phase for the 16-QAM
modulator
2. For a quadbit input of I = 1, I’ = 1, Q = 0,
Q’ = 1 (1101), determine the output
amplitude and phase for the 16-QAM
modulator
Bandwidth Considerations of
16- QAM
• Channels are I, I’, Q, Q’
• Each channel has a bit rate of fb/4

fb
fN =
4
fb
fa =
8
Bandwidth Considerations of
16- QAM
Bandwidth Considerations of
16- QAM
Bandwidth Efficiency
• Often used to compare the performance of
one digital modulation technique to
another
• Is generally normalized to a 1-Hz
bandwidth

fb
BWeff =
fn
• Where: BWeff = Bandwidth efficiency in
bps/Hz or bits/cycle
• fb = transmission rate in bps
• fn = minimum bandwidth in Hz
•
Problem
• Determine the bandwidth efficiencies for
the following modulation schemes with a
transmission rate of 100 Mbps
a) BPSK
b) QPSK
c) 8 – PSK
d) 16 – QAM
Which of modulation scheme is the most
efficient?
Digital Modulation Summary
Carrier Recovery
• It is sometimes called Phase Referencing
• It is the process of extracting a phase-
coherent reference carrier from a receiver
signal
Techniques in Carrier Recovery
1.Squaring Loop
2.Costas Loop
3.Remodulator
Absolute Phase Encoding
• The binary data is encoded as a precise
phase of transmitted carrier
• To correctly demodulate the data, a phase
coherent carrier is recovered and
compared with the received carrier in the
product detector
• To determine the absolute phase of the
received carrier, it is necessary to produce
a carrier at the receiver that is phase
coherent with the transmit reference
oscillator
Suppressed Carrier Systems
• With PSK and QAM the carrier is
suppressed in the balanced modulator and
therefore is not transmitted
• With this system sophisticated methods of
carrier recovery are required.
• These carrier recovery include: squaring
loop, Costas loop, and remodulator
Squaring Loop
• It is a common method of achieving carrier
recovery for BPSK
• The received BPSK waveform is filtered
and then squared
• The filtering reduces the spectral width of
the received noise
• The squaring circuit removes the
modulation and generates the second
harmonic of the carrier frequency
•
Squaring Loop
• This harmonic is phase tracked by the PLL
(Phase Locked Loop)
• The VCO output frequency from the PLL
then is divided by 2 and used as the phase
reference for the product detectors
Costas Loop
• It is also called Quadrature Loop
• It produces the same result as the
squaring circuit
• It uses two parallel tracking loops (I and Q)
simultaneously to derive the product of the
I and Q components of the signal that
drives the VCO
• The in-phase (I) loop uses the VCO as in
PLL
• The quadrature (Q) loop uses a 90 0
shifted VCO
• Once the frequency of the VCO is equal to
the suppressed carrier frequency, the
product of I and Q will produce an error
voltage proportional to any phase error in
the VCO
• The error voltage controls the phase, and
thus the frequency of the VCO
Costas Loop
Remodulator
• The remodulator produces an error
voltage proportional to twice the phase
error between the incoming signal and the
VCO signal
• It has a faster acquisition time than the
previous methods
•
Remodulator
Differential Phase Shift Keying
• It is an alternative form of digital
modulation where the binary input is
contained in the difference between two
successive signal elements rather than the
absolute phase
• With DBPSK it is not necessary to recover
a phase coherent carrier
• Received signaling element is delayed by
one signaling element time slot and them
compared with the next signaling element
• The difference in the phase of the two
signaling elements determines the logic
condition of the data
Differential BPSK
Transmitter
• DBPSK transmitter is shown
• An incoming information bit is XORed with
preceding bit prior to entering the BPSK
modulator
• An initial bit reference is assumed
DBPSK Modulator
BPSK Demodulator
• The received signal is delayed by one bit
time, then compared with the next
signaling element in the balanced
modulator
• If they are the same a logic one (+ voltage)
is generated
• If the referenced phase is incorrectly
assumed, only the first demodulated bit is
in error
• It can be implemented with higher binary
coding
• It needs no carrier recover which is its
advantage
• The disadvantage is that DBPSK requires
between 1 dB and 3 dB more signal-to-
noise ratio to achieved the same bit error
rate as that of absolute PSK
•
Clock Recovery
Clock Recovery
• Digital system requires precise timing or
clock synchronization between the
transmit and received circuitry
• It is necessary to regenerate clocks at the
receiver that are synchronous at the
transmitter
• The recovered data are delayed by one
half bit time and then compared with the
original data in the XOR circuit
• The frequency of the clock that is
recovered with this method is equal to the
received data rate (fb)
• As long as the received data contains a
substantial number of transitions (1/0
sequences) the recovered clock is
maintained
• Extended 1’s or 0’s would loss the
recovered clock, to prevent this data are
scrambled at the transmit end and
descrambled at the received end
• Scrambling introduces transitions using
prescribed algorithm and same algorithm
is used to remove the transitions
Probability of Error and Bit
Error Rate
• Probability of error P(e) is a theoretical
(mathematical) expectation of the bit error
rate in the system
• Bit error rate (BER) is an empirical
(historical) record of the system’s actual bit
error performance.
• Probability of error is a function of the
carrier-to-noise power ratio (or more
specifically the average energy per bit-to-
noise power density ratio) and the number
of possible encoding used (M-ary)
Carrier-to-noise Power Ratio
• Where: C = is the carrier power
• N = thermal noise power
• T = temperature
• B = bandwidth

C C
=
N KTB
C C
(dB ) = 10 log
N N

C
(dB ) = CdBm − N dBm
N
Energy per Bit
Noise Power Density
Energy per Bit-to-noise Density
Ratio
Problem
PSK Error Performance
• The bit error performance for the various
multiphase digital modulation system is
directly related to the distance between
points on a signal state-space diagram
PSK Error Performance
PSK Error Performance
• d = maximum distance of separation
• D = power level
• Vn = noise vector
• Vs = signal vector
• VSE = signaling element
• One BPSK signal state is exact negative
of each other
PSK Threshold Points
• M = number of signal states

π
TP = ±
M
Antipodal Signaling
• Optimum signalling format
• Signals are in 180 0 out of phase
• Two binary levels are allowed
• One signal is exact negative of each other
• Antipodal performance is used often as
reference for comparison
PSK – Maxium Distance
Formula
• d =error distance
• M = number of phases
• D = Peak signal amplitude

360 0 d / 2
sin θ = sin =
2M D
0
180
d = (2 sin )D
M
QPSK – Signal State-space
Diagram
• QPSK can tolerate only ±45 0
• 8 – PSK and 16 PSK maximum phase
shift is ±22.5 0 and ±11.5 0
• The higher the value of M the greater
energy per bit-to-noise density ratio is
required to reduce the effect of noise
interference
• The higher the level of modulation the
smaller the separation between the signal
points and the smaller the error distance
Trellis Encoding
• Data transmission in excess of 56 kbps
can be achieved over a standard
telephone network using an encoding
technique called Trellis Code Modulation
(TCM)
• It was developed by Dr. Ungerboeck at
IBM Zuerich Research Laboratory
• It combines encoding and modulation to
reduce the probability of error
Trellis Encoding
• The idea is to introduce controlled
redundancy which reduces the likelihood
of transmission errors
• The redundancy is done by doubling the
number of signal points in a given PSK or
QAM constellation
• It also defines the manner in which signal-
state transitions are allowed to occur,
transmission that do not follow the pattern
are interpreted as errors in the receiver
Trellis Code Modulation
Trellis Code Modulation
Trellis Coding Gain
32-Point QAM TCM
Constellation
One-fourth of a 960-Point QAM
TCM Constellation
TCM

f b = NB
N = Number of bits encoded
B = Bandwidth in Hertz
fb = transmission bit rate (bps)
Digital Transmission
• It is the transmittal of digital signals
between two or more points in a
communications system.
• Unlike Digital Radio, signals in Digital
Transmission should be binary or any
other form of discrete-level digital pulses.
• With these kind of systems, a physical
facility, such as a pair of wires, is required
to interconnect the various points within a
system.
Advantages
• Noise Immunity
• Better suited for processing and combining
using multiplexing
• More resistant to additive noise because
they use signal regeneration rather than
signal amplification
• Simpler to measure and evaluate than
analog signals.
• Transmission error can be detected and
corrected
Disadvantages
• Requires significantly more bandwidth
than simply transmitting the original analog
signal.
• Additional encoding and decoding circuitry
is needed.
• Requires precise time synchronization
between the clock in the transmitter and
receiver.
• Incompatible with older analog
transmission systems.
Pulse Modulation
• It consists essentially of sampling analog
information signals and then converting
those samples into discrete pulses and
transporting the pulse from a source to a
destination over a physical transmission
medium.
• There are four predominant methods of
pulse modulation: PAM, PWM, PPM and
PCM.
Pulse Amplitude Modulation
• The amplitude of a constant-width,
constant-position pulse is varied according
to the amplitude of the sample of the
analog signal.
• Intermediate form of modulation with PSK,
QAM and PCM
Pulse Width Modulation
• Also called pulse duration modulation
(PDM) or pulse length modulation (PLM).
• A pulse modulation system in which the
amplitude of the pulses is kept constant,
while the width of each pulse is made
proportional to the amplitude of the signal
at the instant
• Used in military systems
Pulse Position Modulation
(PPM)
• A pulse modulation system in which the
amplitude is kept constant, while the
position of each pulse, in relation to the
position of a recurrent reference pulse is
varied by each instantaneous sampled
value of the modulating wave.
• The highest amplitude sample produces a
pulse a pulse to the far right and the
lowest amplitude produces a pulse to the
far left.
Pulse Code Modulation
• Alex H. Reeves is credited with inventing
PCM in 1937 while working for AT & T at
its Paris laboratories.
• The term pulse code modulation is
somewhat a misnomer, as it is not really a
type of modulation but rather a form of
digitally coding analog signals.
PCM
• With PCM, the pulses are of fixed length
and fixed amplitude.
• PCM is a binary system where a pulse or
lack of a pulse with a prescribed time slot
represents either logic 1 or a logic 0
condition. PWM, PPM and PAM are
digitally but seldom binary, as a pulse
does not represent a single binary digit
(bit).
Pulse Modulation
Simplex PCM Transmission System
PCM Sampling
• 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 to be converted to binary
PCM code.
• For the ADC to accurately covert 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.
• 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.

• With flat-top sampling, the input voltage is

sampled with a narrow pulse and then held
relatively constant until the next sample is
taken. 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. flat-top sampling is
accomplished in a sample-and-hold circuit.
• Aperture Error happens when the
amplitude of the sampled signal changes
during the sample pulse time
• This prevents the recovery circuit in the
PCM receiver from exactly reproducing the
original analog signal voltage. The
magnitude of error depends on how much
the analog signal voltage changes while
the sample is being taken and the width
(duration) of the sample pulse.
• Flat-top sampling, however, introduces
less aperture distortion than natural
sampling and can operate with a slower
analog-to-digital converter.
Sample-and-Hold Circuit
Sampling Rate
The Nyquist sampling theorem establishes
the minimum sampling rate (fs) that can be
used for a given PCM system. For a sample to
be produced 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 ≥2 fa
 Quantization and 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.
Thus, converting an analog signal to a PCM
code with a limited number of combinations
requires quantization. In essence, quantization
is the process of rounding off amplitudes of
flat-top samples to a manageable number of
levels.
Dynamic Range
The number of PCM bits transmitted per sample is
determined by several variables, including maximum
allowable input amplitude, resolution and dynamic
range. Dynamic range (DR) is the ratio of the
largest possible magnitude to the smallest possible
magnitude (other than 0V) that can be decoded by
the digital-to-analog converter in the receiver.
Mathematically, dynamic range is
DR= Vmax/Vmin
Where:
DR = dynamic range (unitless ratio)
Vmin = the quantum value resolution
Vmax = the maximum voltage magnitude that can be
discerned by the DACs in the receiver
• Proceed to the next file Oslide 69
PCM
• The analog signal is sampled and
converted to a fixed length serial binary
number for transmission.
• Binary number varies according to the
amplitude of the analog signal
Applications
• PAM – used as an intermediate form of
modulation with PSK, QAM, and PCM and
usually not used by itself.
• PWM & PPM – are used in special
purpose communication systems (usually
for military) but are seldom used for
commercial systems.
• PCM – the most prevalent method of pulse
modulation.
Line Encoding
• Involves converting standard digital logic
levels (TTL or CMOS) to form suitable to
telephone transmission.
• It is a form of serial transmission of binary
signals.
Types of Transmission Voltage
Levels
• Unipolar (UP) Transmission: + V = logic
1, 0 V = logic 0
• Bipolar (BP) Transmission: + V = logic 1,
- V = logic 0
Two General Types of Formats
• Non – Return to Zero (NRZ) – the binary
pulse is maintained for the entire time.
One bit signal does not return to zero
midway between the bit period.
• Return – to Zero (RZ) – if the active time
of the binary pulse is less than 100% of
the bit time. Signal of one bit returns to
zero between the entire bit period.
Line Encoding Formats
Format Types
• UPNRZ (Unipolar Non-Return to Zero) –
a change in state only occurs when there
is a 1 to 0 or 0 to 1 transition. A string of
1’s is a continuous pulse or “ON”
condition, and a string of 0’s is a
continuous “OFF” condition. A single pulse
completely occupies the designated
interval.
Format Types
• BPNRZ (Bipolar Non-Return to Zero) –
similar to UPNRZ except that there are
now two nonzero voltages (+V for logic 1
and –V for logic 0)
• UPRZ (Unipolar Return to Zero) – there
is a transition for every bit transmitted
whether a 1 or 0, the signal always returns
to zero and this results to a pulse width
less than the bit interval.
Format Types
• BPRZ (Bipolar Return to Zero) – Similar
to UPRZ except that there are two voltage
levels.
• BPRZ – AMI (BPRZ Alternate Mark
Inversion) – the two polarity voltage levels
(+V and – V) represents logic 1 and 0 V
represents logic 0. Each successive logic
1’s is inverted in polarity from the previous
logic 1.
Format Types
• Manchester Encoded Forms – the binary
information is carried in the transition that
occurs at the mid pulse. It has a built-in
clock capability at the center of each data
bit period.
Types of Manchester Encoded
Forms
• Manchester – the first half of the data bit
period is inverted and change in state
follows.
• Differential Manchester – moves the
detection of data level to the leading edge
of a data bit time. A change in state at the
beginning of the data bit period represents
logic 0, if there is no change in state
means logic 1.
•
Problems
1. Show the Manchester and Differential
Manchester of the bit stream
111100101101.
2. Show the UPNRZ, BPNRZ, and BPNRZ-
AMI of the bit stream 111001101010011.
Show also the Manchester and
Differential Manchester of the bit stream.
Line Encoding Comparison

Encoding Minimum Average DC Clock Error

Format Bandwidth Recovery Detection

UPNRZ fb/2 +V /2 Poor No

BPNRZ fb/2 0V Poor No

UPRZ fb +V/2 Good No

BPRZ fb 0V Best No

BPRZ-AMI fb/2 0V Good Yes

Manchester fb 0V Best No
PCM
• It was invented by Alex H. Reeves. It was
developed by AT&T in 1937 at their Paris
Laboratory.
• PCM is the preferred method of
communication within the Public Switch
Telephone Network (PSTN).
• It is a form of source coding.
PCM Simplex Diagram
PCM
• Bandpass filter limits the frequency of the
input analog signal to the voice band
frequency range (300 – 3000 Hz).
• The sample-and-hold circuit periodically
samples the analog input and converts
those samples to a multilevel PAM signal.
•
PCM
• Analog-to-digital converter (ADC) converts
the PAM samples to parallel PCM codes
which are converted to serial data in the
parallel-to-serial converter then outputted
onto the transmission lines.
• At the receiver the reverse process is
done.
• An integrated circuit that performs PCM
encoding and decoding functions is called
CODEC (Coder/Decoder).
PCM Sampling
• There are two types of sampling: Natural
Sampling and Flat-top Sampling
• Natural Sampling – the tops of the
sampled analog waveform retain their
natural shape.
• Flat-Top-Sampling – it is the most
common method of PCM systems. It
introduced aperture error which prevents
exact reproduction of the original signal.
PCM Sampling
• When Q1 is turned on a low impedance
path is established to deposit the analog
voltage across the capacitor C1.
• The time Q1 is on is called the Aperture
Acquisition Time.
• C1 is the hold circuit.
• When Q1 is off C1 does not have the
complete path to discharge. (Storage
time).
• The storage time is called A/D
Conversion Time. Because at this time
ADC converts the sample voltage to a
PCM code.
• The acquisition time should be very short
to minimize aperture error.
• Flat-top sampling introduces less aperture
error that the natural sampling and
requires slower analog-to-digital converter.
• To avoid droop (cause by capacitor
discharging through its own leakage
resistance and input impedance Z2), Z2
and the leakage resistance of C1 should
be as high as possible.
For the Capacitor

dV dt
i=C C =i
dt dV
• Where: C = maximum capacitance
i = maximum output current from Z1
dV = maximum change in voltage across C1
dt = charge time which equals the aperture
time.
 τ
C=
R
• Where τ = one charge time constant
R = output impedance of Z1 plus the on
resistance of Q1
C = capacitance value in farad
Accuracy (%) Charge Time
10 2.3τ
1 4.6τ
0.1 6.9τ
0.01 9.2τ
Problem
1. For the sample-and-hold circuit,
determine the largest-value capacitor that
can be used. Use the output impedance
for Z1 of 10 Ω, and on resistance of Q1 of
10 Ω, an acquisition time of 10 µs, a
maximum peak-to-peak voltage of 10 V, a
maximum output current from Z1 of 10
mA, and an accuracy of 1%.
Sampling Rate

fs = 2 fa
• Nyquist Sampling Theorem establishes
the minimum sampling rate (fs)
• Where fs = minimum Nyquist sampling
rate
fa = highest frequency to be sampled
• If fs is less than twice fa a distortion called
Aliasing or Foldover Distortion occurs.
(Harmonics)
• Input bandpass filter at the PCM block
diagram is called Antialiasing or
Antifoldover Filter.
• Cut off frequency of the filter is calculated
as follows:
fs
fc =
2
• Where fc = is the cutoff frequency
• fs = sampling frequency
Problem
1. For a PCM system with a maximum audio
input of 4 kHz; determine the minimum
sample rate and the alias frequency
produced if 5 kHz audio signal were
allowed to enter the sample-and-hold
circuit.
PCM Quantization
• Quantizing or Quantization – assigning
PCM codes to absolute magnitudes
(samples).
• If the magnitude of the sample exceeds
the highest quantization interval Overload
Distortion occurs (also called peak
limiting).
• Resolution – it is the magnitude of the
minimum step size. This is equal to the
magnitude to the voltage of the least
significant bit.
• The smaller the step size the better the
resolution
• Quantization range should be equal to
one half the resolution.
• Quantization Error – it is the difference
between the quantized and unquantized
samples (also called Quantization Noise)
1
Qe = ∆
2
• Where Qe = Quantization Error
∆ = resolution or Step size
• Dynamic Range – the ratio of the largest
possible magnitude to the smallest
possible magnitude that can be decoded
by DAC.

DR =
V max V max
= DRdB = 20 log DR
V min ∆
log( DR + 1)
n n=
DR ≤ 2 − 1 log 2
• Where n = no. of PCM bits excluding the
sign bit
DR = absolute value of Dynamic Range
• Coding Efficiency – represents how
efficiently a PCM code is utilized
Bit min
%CodingEff = 100
Bit Actual
• Where: Bitmin = minimum number of bits
(Including sign Bit)
BitActual = actual number of bits (Including
sign Bit)
•
Problem
1. A PCM system has the following
parameters: maximum analog frequency
of 4 kHz; maximum decode voltage at the
receiver of 2.55 Vp; and a minimum
dynamic range of 46 dB. Determine the
following: minimum sample rate, minum
number of bits used in PCM code;
resolution; quantization error and coding
efficiency.
PCM Coding
1. Linear
2. Nonlinear
Linear Coding
• The magnitude change between any two
successive steps is uniform.
• In linear encoding the accuracy
(resolution) for higher amplitude signals is
the same as the lower amplitude signals
Nonlinear Coding
• The step size increases with the amplitude
of the input signal
• There are more codes at the bottom of the
scale than there are at the top thus
increasing the accuracy for smaller
signals.
• The SQR is sacrificed for high amplitude
signals
• It is difficult to manufacture nonlinear
ADC’s.
Idle Channel Noise
• It is the thermal noise which is present
when there is no analog input signal to
PAM sampler.
• The input noise is quantized by the ADC.
• Midthread Quantization – reduces idle
channel noise. In this method the first
quantization interval is made larger in
amplitude than the rest of the steps. As a
result noise is suppressed during encoding
process.
•
• Midrise Quantization – is used to decode
idle channel noise. In this process the
lowest magnitude positive and negative
codes have the same voltage range.
Coding Methods
1. Level-at-a-time Coding
2. Digit-at-at-time Coding
3. Word-at-at-time Coding
Level-at-time Coding
• Compares the PAM signal to a ramp
waveform while a binary counter is being
advanced at a uniform rate.
• When the ramp waveform equals or
exceeds the PAM sample, the counter
contains the PCM code.
• It requires a very fast clock and 2 n
sequential decisions be made for each
PCM code.
• It is generally limited to low speed
applications
Digit-at-a-time Coding
• It determines each digit of the PCM code
sequentially.
• It is analogous to a balance where a
known reference weights are used to
determine the unknown weight.
• It is a compromise between speed and
complexity
Word-at-a-time Coding
• These are flash encoders and more
complex usually suitable for high speed
applications.
• It uses multiple threshold circuits and is
impractical for large number of bits.
Companding
• It is a process of compressing and
expanding signals.
• Compression is done at the transmitter
and expansion at the receiver
Types of Companding
• Analog
• Digital
Analog Companding
• It uses specially designed diodes.
• Two types of analog companding: µ - Law
and A – Law
µ - Law Companding
It is the companding used in US and Japan
The formula is:
• Voice transmission requires a minimum
DR of 40 dB and a 7 bit PCM code at µ =
100 or higher
• Most recent digital transmission systems
use 80 bit PCM system at µ = 255.
Problem
1. For a compressor with a µ = 255,
determine
a) The voltage gain for the following relative
values of Vin: Vmax, 0.75 Vmax, 0.5 Vmax,
and 0.25 Vmax.
b) The compressed output voltage for a
maximum input voltage of 4V.
c) Input and output dynamic ranges and
compression.
A – Law Companding
• It is used in Europe and was established
by CCITT.
• It is used to approximate logarithmic
companding.
• It is inferior to µ- Law in terms of small
signal-quality (Idle Channel Noise)
 Vin
A Vin 1
Vmax 0≤ ≤
Vout = Vmax Vmax A
1 + ln A

1 + ln( A
Vin
) 1 Vin
V max ≤ ≤1
Vout = Vmax A Vmax
1 + ln A
Digital Companding
• Involves compression at the transmit end
after the input sample has been converted
to linear PCM code.
• Expansion is done at the receiver prior to
PCM decoding.
• It uses 12-bit code and an 8 bit
compressed code.
Vocoders
• Also known as voice encoders (vocoders)
• Designed to reproduce only the short term
power spectrum and the decoded time
waveforms often vaguely resemble the
original input signal.
• It commonly reproduced unnaturally
sounding speech and therefore generally
used for recorded information such as
“wrong number messages”, encrypted
voice for transmission over analog
telephone circuits, computer output signals
and educational games.
Three Techniques in Vocoding:
1. Channel Vocoder
• Developed by Homer Dudley in 1928
• Compressed conventional speech
waveforms into an analog signal with a
bandwidth of 300 Hz and operates at less
than 2 kbps
2. Formant Encoder
• Takes advantage of the short-term
spectral density of typical speech which is
commonly located at three to four peak
frequency called formant.
3. Linear Predictive Coders
• Provides more natural sounding speech
PCM Line Speed
• It is the rate at which serial PCM bits are
clocked out of the transmitter onto the
transmission line or from the transmission
line to the receiver.

Samples Bits
LineSpeed = ( )( )
Second Sample
Problem
1. Find the line bit of single channel PCM
system with a sample rate fs = 8000
samples/second and an 8-bit compressed
PCM.
Delta Modulation PCM
• It uses single-bit PCM code to achieve
digital transmission of analog signals.
• If the current sample is smaller than the
previous sample logic zero is transmitted.
• If the current sample is larger than the
previous sample, logic 1 is transmitted.
Problems in Delta Modulation
1. Slope Overload Distortion

The slope of the analog signal is greater
than the delta modulator can maintain.
2. Granular Noise
• The reconstructed signal has variations
that were not present in the original signal.
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 signal.
• When the output of the transmitter is a
string of consecutive 1’s or 0’s, this
indicates that the slope of the DAC output
is less than the slope of the analog signal
in either the positive or negative direction.
Differential Pulse Code
Modulation
• Designed specifically to take advantage of
the sample-to-sample redundancies in
typical speech waveforms.
• Only the difference in amplitude of two
successive samples is transmitted rather
than the actual sample as in standard
PCM.
• Since the range of the sample difference is
typically less than the range of the
individual samples, lesser bits are
required.
•
Pulse Transmission
• All digital carrier system systems involve
the transmission of pulses through a
system with finite bandwidth.
• Therefore, practical digital systems
generally utilized filters bandwidths that
are 30 % or more in excess of the ideal
Nyquist bandwidth.
•
• Bandlimiting pulse causes the energy from
the pulse to spread over significantly
longer time in the form of secondary lobes
called Ringing Tails.
• If the signaling rate is confined within the
limit of the Nyquist rate the signal can be
preserved without causing excessive
distortion.
 R = 2B
Where: R = signaling rate = 1/T or the
Nyquist rate
B = specified bandwidth
Intersymbol Interference
• For ideal filters transmitting an NRZ input
signal, the output signal reaches it full
value for each transmitted pulses precisely
the center of the sampling interval.
• For non-ideal filters; the output response
will resemble a signal that does not attain
the maximum value.
• The ringing tails of several pulses have
overlapped, thus interfering the major
lobe. This is called Intersymbol
Interference (ISI).
Why is ISI important in pulse
transmission?
• Circuits with limited bandwidth and non-
linear response are affected by it.
• The narrower the bandwidth, the more
rounded the pulses become. If the
distortion is excessive, the pulse may tilt
and can affect the next pulse.
• ISI can cause cross talk between channels
that occupy the adjacent slots in TDM
systems.
• Equalizers – a special filter inserted in the
transmission path to equalize the distortion
for all frequencies, creating uniform
transmission medium and reducing the
transmission impairments caused by ISI.
Causes of ISI
• Timing inaccuracies
• Insufficient bandwidth
• Amplitude distortion
• Phase distortion
Multiplexing
• Multiplexing is the process of
simultaneously transmitting two or more
distinct signals over a single
communications channel.
Types of Communications
Channel
1. Bounded

– Metallic wire cable

– Optical wire cable
2. Unbounded
• Microwave Radio Link
Examples of Mutiplexing
• Telephone Systems
• Radio Broadcasting
• Satellite Communication
•