COMMUNICATION

THEORY 1
ECM 221
CHAPTER 4:

DIGITAL
COMMUNICATION
SYSTEMS
 Introduction of digital modulation

• ν (t ) = V sin( 2π f t +θ )
•
• If the information signal is digital and the amplitude (V) of
the carrier is varied proportional to the information signal
 ASK is produced.
• If the frequency (f) of the carrier is varied proportional to the
information signal  FSK is produced.
• If the phase () of the carrier is varied proportional to the
information signal  PSK is produced.
• If both the amplitude (V) and the phase ( ) of the carrier is
varied proportional to the information signal  QAM is
produced.
• ASK, FSK, PSK and QAM are all forms of digital modulation.
 Application of digital modulation

• Low-speed voice band data communications modems, such

as those found in most personal computers.
• High-speed data transmission systems, such as broadband
digital subscriber lines (DSL)
• Digital microwave and satellite communication systems
• Cellular telephone Personal Communication Systems (PCS).
 INTRODUCTION
Digital transmission


• The transmittal of digital signals between two or more

points in a communications system.
• Can be binary or any other form of discrete-level digital
pulses.
• The original source information may be in digital form or
it could be analog signals that have been converted
to digital pulses prior to transmission and converted
back to analog signals in the receiver.
• Physical facility: a pair of wires, coaxial cable or an
optical fiber cable is required to interconnect the
various points within the system.
• Digital pulses cannot be propagated through a wireless
transmission system such as Earth’s atmosphere or
free space (vacuum).
• Today, digital transmission systems are used to carry
not only digitally encoded voice and video signals but
also digital source information directly between
computers and computer networks.
• Digital transmission systems use both metallic and
optical fiber cables for their transmission medium
 ADVANTAGES OF DIGITAL TRANSMISSION
• Noise immunity
• Inherently less susceptible to interference because it is not
necessary to evaluate the precise amplitude, frequency
or phase to ascertain its logic condition.
• Better suited for processing and combining using a technique
called multiplexing.
• Digital signal processing (DSP) is the processing of analog
signals using digital methods and includes bandlimiting
the signal with filters, amplitude equalization and phase
shifting.
• Much simpler to store and the transmission rate can be easily
changed to adapt different environments and to interface
with different types of equipment.
• More resistant to additive noise
• They use signal regeneration rather than signal amplification.
• Can be transported longer distances than analog signals..
• Simpler to measure and evaluate
• Easier to compare the error performance of one digital system
to another digital system
• Transmission errors can be detected and corrected more easily
and more accurate.
 DISADVANTAGES OF DIGITAL SIGNAL

• The transmission of digitally encoded analog signals

requires significantly more bandwidth than simply
transmitting the original analog signal.
• Bandwidth is one of the most important aspects of any
communications system because it is costly and
limited.
• Analog signals must be converted to digital pulses
transmission and converted back.
• Requires precise time synchronization between the
clocks in the transmission and receivers
• Incompatible with older analog transmission systems
 Information capacity, bits, bit rate, baud and
M-ary encoding

• Information capacity refer to Hartley’s Law (Already

covered in Chapter 1).
• Binary digit or bits: the most basic digital symbol used to
represent information.
• Bit rate is simply the number of bits transmitted during
one second and is expressed in bits per second (bps).
• Shannon law: (refresh)

 S
I = B log 2 1 + 
 N
or
 S
I = 3.32 B log 10 1 + 
 N
 M-ary encoding

• M-ary is a term derived from the word binary.

• M represents a digit that corresponds to the number of
conditions, levels and combinations possible for a given
number of binary variables.
• The number of bits necessary to produce a given number of
conditions is expressed mathematically as:
 N = log2 M
 where N = number of bits necessary
 M = number of conditions, levels, or
combinations possible with N bits.
• Simplified: 2N = M
 Baud and minimum bandwidth

• Baud is a term that is often misunderstood and commonly

confused with bit rate (bps).
• Bit rate refers to the rate of the change of a digital
information signal, which usually binary.
• Baud is also a rate of change, however baud refers to the
rate of change of a signal on the transmission medium
after encoding and modulation have occurred.
• Baud = 1/ts where baud = symbol rate (baud per second)

ts= time of one signaling
element (seconds)
• Nyquist bandwidth:
– Binary digital signals can be propagated thru an ideal
noiseless transmission medium at a rate equal to two
times the bandwidth of the medium.
 Baud and minimum bandwidth

• Nyquist bandwidth:
– Binary digital signals can be propagated thru an ideal
noiseless transmission medium at a rate equal to two
times the bandwidth of the medium.
– The minimum theoretical bandwidth necessary to
propagate a signal is called Nyquist bandwidth or
sometimes the minimum Nyquist frequency.
– Thus:
• fb= 2 B where fb= bit rate in bps, B = ideal
Nyquist bandwidth.
• The actual bandwidth necessary to propagate a given bit
rate depends on several factors:
– Type of encoding -- system noise
– Modulation used -- desired error performance
– The types of filter used
–
 Baud and minimum bandwidth

• If multilevel signaling is used, the Nyquist formulation for

channel capacity is:
 fb = 2 B log 2 M
 where: fb = channel capacity (bps)
 B = minimum Nyquist bandwidth (hertz)
 M = number of discrete signal or voltage
levels.
 Amplitude Shift Keying (ASK)

• The simplest digital modulation technique is amplitude- shift

keying (ASK), where a binary information signal directly
modulates the amplitude of an analog carrier.
• In ASK, a carrier wave is switched ON and OFF by the input
data or binary signals. During a ‘mark’ (binary ‘1’), a
carrier wave is transmitted and during a ‘space’ (Binary
‘0’), the carrier is suppressed.
• Hence, it also known as ON- OFF keying (OOK).
• Mathematically, amplitude- shift keying is
A 
v( ask ) (t ) = [1 + vm (t )]  cos(ωc t )



2 
 Where:
– (ask)( t) = amplitude- shift keying wave
– νm(t) = digital information (modulating) signals (volts)
– A/2 = unmodulated carrier amplitude (volts)
– ωc = analog carrier radian frequency
Amplitude Shift Keying (ASK)
 Frequency Shift Keying (FSK)

• Frequency-shift keying (FSK) is another relatively simple, low-

performance type of digital modulation.
• FSK is a form of constant-amplitude angle modulation similar to
standard frequency modulation (FM) except the modulating
signal is a binary signal that varies between two discrete
voltage levels rather than a continuously changing analog
waveform.
• The general expression for FSK is:
 νfsk (t) = νccos { 2π [ fc + νm (t)∆f ] }
• Where νfsk (t) = binary FSK waveform
 vc = peak analog carrier amplitude (volts)
 fc = analog carrier center frequency (volts)
 ∆f = peak change (shift) in the analog carrier
frequency (hertz)
 νm(t) = binary input (modulating) signal (volts)
 Frequency Shift Keying (FSK)

• The modulating signal is a normalized binary waveform

where a logic 1= +1V and logic 0 = -1V.
• For logic 1 or mark input :
 νfsk (t) = Vccos [ 2 ( fc + ∆f )]
• For logic 0 or space input :
 νfsk (t) = Vccos [ 2 ( fc - ∆f )]
• The mark frequency is the higher frequency ( fc + ∆f ).
• The space frequency is the lower frequency ( fc - ∆f ).
•
•
•
•
Frequency Shift Keying (FSK)
 Phase Shift Keying (PSK)

• The simplest form of PSK is binary phase-shift keying

(BPSK), where N=1 and M=2.
• Therefore, BPSK  2 phase (21=2) are possible for the
carrier.
• One phase represents a logic 1, and the other phase
represents a logic 0.
• As the input digital signal changes state (ie, from a 1 to a 0
or from 0 to 1), the phase will switch normally 0 to
180.
Phase Shift Keying (PSK)
 Phase Shift Keying (PSK)

error
 PULSE MODULATION
A process of sampling analog information signals and then
converting
those samples into discrete pulses and transporting the pulses from

a source to a
destination over a physical transmission medium.

The predominant methods of pulse modulation

1) Pulse width modulation (PWM)

• Called pulse duration modulation (PDM) or pulses length

modulation (PLM)
• The technique of varying the width of the constant-amplitude
pulse proportional to the amplitude of the modulating signal.
2) Pulse position modulation (PPM)

 The higher the amplitude of the sample, the farther to the

right the pulse is positioned within a prescribed time slot.
3) Pulse amplitude modulation (PAM)

• The amplitude of a constant width

• Constant position pulse is varied according to the amplitude of the
sample of the analog signal.
• Waveform resemble the original analog signal more than the
waveforms for PWM or PPM
4) Pulse code modulation (PCM)

• Sampled and converted to a serial n-bit binary code for

transmission.
• Each code has the same number of bits
• Same length of the time for transmission
 PULSE CODE MODULATION (PCM)

• PCM is only digitally encoded

modulation technique that
commonly use for digital
transmission.
• With PCM, the pulses are of fixed
length and fixed amplitude.
• It is a binary system where a pulse
or lack of a pulse within a
prescribed time slot represents
either a logic 1 or a logic 0
Figure 10.1: Pulse Modulation: (a) analog signal (b) sample pulse (c) PWM
(d) PPM (e) PAM (f) PCM
re 1 0 . 2 : S im p lifie d b lo ck d ia g ra m o f a sin g le -ch a n n e l, sim p lex PC M tra n sm issio n
 From The Block Diagram
1 The Simple-and-hold circuit;
• It periodically samples the analog signal and converts
those samples to a
multilevel PAM signal
2 Analog-to-digital converter (ADC);
• Convert the PAM samples parallel PCM codes which are
converted to serial binary data in the parallel-to-
serial converter. After that, the outputted onto the
transmission line as serial digital pulse.
3 Repeaters;
• are placed at prescribed distance to regenerate the
digital pulse.
• In receiver a serial-to-parallel converter converts the
serial pulses to to parallel PCM code.
4 Digital-to-analog converter (DAC);
• To converts the parallel PCM codes to multilevel PAM
signals
5 The hold circuit is basically a low-pass filter that
converts the PAM signals back to the original analog
form.
 PCM Sampling
• Function of sampling circuit in PCM transmitter is to
periodically sample the continually changing analog
input voltage and convert those samples to a series
of constant-amplitude pulse that can more easily be
converted to binary PCM code.
•  
• ü      Natural sampling
• When tops o the sample pulses retain the natural shape
during the sample interval, making it difficult for ADC
to convert the sample to PCM code.
• With natural sampling, the frequency spectrum of the
sampled output is different from that of an ideal
sample.
•  
• ü      Flat-top sampling
• The most common method use for sampling voice
signals in PCM, which is accomplished in a sample-
and-hold circuit.
 (a)

(b)

(c)

Figure 10.3: Natural sampling Figure 10.4: Flat-top sampling

(a) input analog signal (b) (a) input analog signal (b) sample
sample pulse (c) sampled pulse (c) sampled output
output
 Example 10.1
• For the sample and hold circuit shown in figure 10.5a,
determine the largest value capacitor that can be used. Use
an output impedance for Z1 of 10 , an on resistance for Q1
of 10  , an acquisition time of 10  s, a maximum peak to
peak input of 10 V, a maximum output current from Z1 of
10 mA, and an accuracy of 1%.
•  Solution
• Current through a capacitor
•
•   i =C
dv
•   dt
• Rearranged and solve for C
•  
•   dt
•   C = i
• Where; dv
• C = maximum capacitance
• i = maximum output current form Z1, 10 mA
• dv = maximum change in voltage across C1, which equals 10
V
• dt = change time, which equals to aperture time, 10 s
•
• Therefore, (10mA)(10 µs )
•   Cmax = =10nF
10V
•
• The charge time constant for C1 when Q1 is on is
•  
•   τ = RC
• Where
•  
•  = one charge time constant (s)
• R= output impedance of Z1 plus the on resistance of Q1 ()
• C= capacitance value of C1 (F)
•  
• Rearranging and solving for C gives us
•    τ
•   Cmax =
R
•  
• The charge time of capacitor C1 is also dependent on the
accuracy desired from the device. The percent accuracy
and its required RC time constant are summarized as
follows:
•
 Accuracy (%) Charge time

10 2.3

1 4.6

0.1 6.9

0.01 9.2

For an accuracy of 1%

10µ s
C= = 108.7 nF
4.6 (20)
 Figure 10.6

• Figure 10.6a-
– The output spectrum includes two original inputs –
audio and the fundamental frequency of the
sampling pulse
– Shows the sum and difference frequencies (fs  fa) all
the harmonics of fs and fa and their associated cross
products.
– It is made up of a series of harmonically related sine
waves.
– None of the side frequencies form one harmonic will
spill into the sidebands of another harmonic and
aliasing does not occur.
• Figure 10.6b
• Shows the results when an analog input frequency greater
than fs/2 modulates fs.
• Aliasing or foldover distortion has occurred. It cannot be
removed through any technique.
• Example 10.2
•
• For a PCM system with a maximum audio input frequency of 4
kHz, determine the minimum sample rate and the alias
frequency produced if a 5 kHz audio signal were allowed to
enter the sample-and-hold circuit.
• Solution
• Using Nyquist’s sampling
• fs ≥ 2 fa
•
• Therefore
• f s ≥ 8 kHz
•
• 10.2

•
•
• If a 5 kHz audio frequency entered the sampled-and-hold
circuit, the alias frequency of 3 kHz has been introduced
into the original audio spectrum.
• With PCM, an analog input signal is sampled, then converted
to a serial binary code and transmitted to receiver where
it is converted back to the original analog signal. PCM
uses n-bit codes. Table 10.1 shows an n-bit PCM code
where n = 3. The most significant bit is used to represent
the sign of the sample
• Logic 1 = positive; logic 0 = negative.
• The two remaining bits represent the magnitude. There are
four codes possible for positive numbers and four codes
for negative numbers. The total is eight possible code L =
2n = 23 = 8
 QUANTIZATION AND THE FOLDED BINARY
CODE

• Quantization; is the process of converting an infinite

number of possibilities to a finite number of conditions.
• Converting an analog signal to a PCM code with a limited
number of combinations of requires quantization.
• It also a process of rounding off the amplitude of flat-top
samples to a manageable number of level.
• For example, a sine wave with a peak amplitude of 5V
varies between +5V and -5V.
 Table 10.2:


– -The total voltage range is subdivided into a smaller

number of sub-ranges.
– -For a three bit sign-magnitude code with eight
possible combinations – (four +ve and four –ve)
– -The leftmost bit is the sign bit
– -The two rightmost bits represent magnitude
– -It is called as a folded binary code – the code on the
bottom half of the table are a mirror image of the
codes on the top half.
– -The magnitude difference between adjacent step is
called the quantization interval or quantum
–
– -Resolution V is defined as the voltage of the
minimum step size, which is equal to the voltage
of the least significant bit f the PCM code. The
Vmax as
equation can be written
–
∆V =
L −1
–
–
– -The smaller the magnitude of a quantum, the better
(smaller) the resolution and the more accurately
the quantized signal will resemble the original
analog sample.
E xa m p le o n h o w to illu stra te a n
a n a lo g w a ve fo rm sig n a l ca n b e Analoginput signal

co d e d in to 3 b its co d e u sin g
sin g le m o d e fo r tra n sm issio n
u sin g PC M te ch n iq u e .

S o lu tio n :
Samplingpulse

Fo r a 3 b its co d e o f sin g le m o d e
tra n sm issio n syste m u sin g P C M Sampledwaveform

te ch n iq u e .
111

101 101

Quantizedsignal
011

010
001

0 1 0 1 0 1 1 1 1 1 0 1 0 1 1 0 0 1

PCMpulses
Figure 10 . 8 :
1) Each sample voltage is rounded
of to the closest available level
and then converted to its
corresponding PCM code.

2) The PAM signal in the

transmitter is essentially the
same PAM signal produced in the
receiver.

3) Any round-off errors

(quantization error Qe or
quantization noise Qn) in the
transmitter signal are reproduced
when the code is converted back to
analog in the receiver.
Figure 10.9 shows the same analog input
signal shown in figure 10.8 except the
signal is being sampled at a much higher
rate.

sample voltage
The folded PCM code is =
resolution

The quality of the PAM signal can be

improved by

a)Using a PCM code with more bits,

reducing the magnitude of a quantum
and improving the resolution.
b) Sampling the analog signal
at a faster rate
Figure 10.10: The quantized signal is a
staircase function
 Example 10.3:

• For a PCM coding scheme shown in Figure

10.8, determine the quantized voltage,
quantization error (Qe) and PCM code for
the analog sample voltage of 1.07 V.
• Solution:
• To determine the quantized level, divide the
sample voltage by resolution and then
round the answer off to the nearest
quantization level +1.07
=1.07 =1
• 1
•
• The quantization error is the difference
between the original sample voltage and
Qe =1.07
the quantized −1 =0.07
level,
 PCM transmission bit rate (R) and Transmission
bandwidth (TB)
• R – the rate of information transmission (bit/s).
It is depends on the sampling frequency and
the number of bit per sample used to
encode the signal. Transmission bandwidth
(TB) is equal to the transmission bit rate, but
the unit is hertz

R = n × f s bits / sec

TB = n × f s Hz
 Dynamic Range (DR)

• 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. V
• DR = max
Vmin
•
• Where;
• DR = dynamic range (unit less ratio)
• Vmin = the quantum value (resolution)
• Vmax = the maximum voltage magnitude that can be
discerned by the DACs in the receiver
Vmax
• So DR =
• resolution
•
• In dB, Vmax
DR = 20 log
• Vmin
•
• Example 10.4
• For a PCM system with the following parameters, determine
(a) minimum sample rate, (b) minimum number of bits
used in the PCM code, (c) resolution and (d) quantization
error.
• Maximum analog input frequency = 4 kHz
• Maximum decoded voltage at the receiver =  2.55 V
• Minimum dynamic range = 46 dB
•
• Solution fs = 2 f a = 2( 4 kHz ) =8 kHz
• The minimum sample rate is
•
• The absolute of dynamic range Vmax
•
46 dB = 20 log
Vmin
•
• DR =199 .5
•
• log(199 .5 +1)
n= = 7.63
• The minimum number of bit is log 2
• To choose the number of bit must be greater than minimum
value which is equal to 8. it is also require a bit for the sign
bit. Therefore the total number of bit is nine. The total
number of the PCM code is 29 = 512
• The actual dynamic range
DR ( dB ) = 20 log (2 n − 1)
•
= 48.13 dB
•
•
• Note: the actual dynamic range is not include the sign bit
• The resolution is determined by dividing the maximum positive
or maximum negative voltage by the number of positive or
negative nonzero PCM codes
Vmax Vmax
2.55
• ∆V = = n = 8 = 0.01V
• L −1 2 −1 2 −1
•
• The maximum quantization error is
resolution 0.01V
Qe = = = 0.005V
2 2
• PCM Line Speed
• Line speed is a data rate at which serial PCM
bits are clocked out of the PCM encoder
into a transmission samples line.
bits
line speed = ×
• Where, sec ond sample

•
• Line speed = the transmission rate in bits per
second
• Samples/second = sample rate (fs)
• Bit/sample = number of bits in the
compressed PCM code.
•
• Example 10.8
• For a single channel PCM system wit a
sample rate fs6000 = samples
6000 samples
7 bits per second
line speed = ×
and a seven bitseccompressedond sample of PCM code,
determine the line speed.
• Solution = 42 000 bps