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ECM410

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DIGITAL COMMUNICATION
SYSTEMS

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Digital transmission is the transmission of digital signals
between two or more points in a communications system.

The original source information may be in digital form which
could be binary or any other form of discrete-level digital pulses,
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.

Digital transmission systems use both metallic and optical fiber
cables for their transmission medium. Digital pulses cannot be
propagated through a wireless transmission system such as
Earths 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.
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1. Noise immunity

2. Inherently less susceptible to interference because it is not necessary to
evaluate the precise amplitude, frequency or phase to ascertain its logic
condition.

3. Better suited for processing and combining using a technique called
multiplexing.

4. 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.

5. Much simpler to store and the transmission rate can be easily changed to
adapt different environments and to interface with different types of
equipment.

6. More resistant to additive noise

7. They use signal regeneration rather than signal amplification.

8. Can be transported longer distances than analog signals.

9. Simpler to measure and evaluate

10. Easier to compare the error performance of one digital system to another
digital system

11. Transmission errors can be detected and corrected more easily and
more accurate.
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1. The transmission of digitally encoded analog signals requires
significantly more bandwidth than simply transmitting the
original analog signal.

2. Bandwidth is one of the most important aspects of any
communications system because it is costly and limited.

3. Analog signals must be converted to digital pulses
transmission and converted back.

4. Requires precise time synchronization between the clocks in
the transmission and receivers

5. Incompatible with older analog transmission systems
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1. Low-speed voice band data communications modems, such
as those found in most personal computers.

2. High-speed data transmission systems, such as broadband
digital subscriber lines (DSL).

3. Digital microwave and satellite communication systems.

4. Cellular telephone Personal Communication Systems (PCS).
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Information capacity refer to Hartleys 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).
Bit rate refers to the rate of the change of a digital information signal, which is
usually binary.

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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 = log
2
M
where N = number of bits necessary
M = number of conditions, levels, or combinations possible with N bits.

Simplified: 2
N
= M
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Nyquist bandwidth defines as the minimum theoretical bandwidth
necessary to propagate a signal . Nyquist bandwidth also sometimes known
as minimum Nyquist frequency.

Binary digital signals can be propagated through an ideal noiseless
transmission medium at a rate equal to two times the bandwidth of the
medium.
f
b
= 2 B where f
b
= bit rate in bps,
B = ideal Nyquist bandwidth.

The actual bandwidth necessary to propagate a given bit rate depends on
several factors which is type of encoding, system noise, types of modulation
used, desired error performance and the types of filter used.
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There are several digital modulation techniques used to modulate
digital signal or data, depending on the applications, the rate of
transmission required, allocated bandwidth and cost.


By referring to the equation above, 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 information signal is digital and the frequency (f) of the carrier is
varied proportional to the information signal FSK is produced.

If the information signal is digital and the phase (u) of the carrier is
varied proportional to the information signal PSK is produced.

If the information signal is digital and both the amplitude (V) and the
phase (u) of the carrier is varied proportional to the information signal
QAM is produced.

ASK, FSK, PSK and QAM are all forms of digital modulation.
) 2 sin( ) ( u t v + = t f V t
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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



Where:
u
(ask)(
t) = amplitude- shift keying wave

m
(t) = digital information (modulating) signals (volts)
A/2 = unmodulated carrier amplitude (volts)

c
= analog carrier radian frequency
| |
(

+ = ) cos(
2
) ( 1 ) (
) (
t
A
t v t v
c m ask
e
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ASK waveform:








Applications of ASK:
It is used in multichannel telegraph system. Simple ASK is no longer
used in digital communication system due to noise problems.
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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:

fsk
(t) =
c
cos { 2 [ fc +
m
(t)f ] }

Where
fsk
(t) = binary FSK waveform
v
c
= peak analog carrier amplitude (volts)
f
c
= analog carrier center frequency (volts)
f = peak change (shift) in the analog carrier frequency (hertz)

m
(t) = binary input (modulating) signal (volts)
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From the general equation of FSK, it can be seen that the peak shift in the
carrier frequency (f ) is proportional to the amplitude of binary input signal
Vm(t), and the direction of the shift is determined by the polarity.

The modulating signal is a normalized binary waveform where a
logic 1= +1V and logic 0 = -1V.

Hence, for logic 1 or mark input :

fsk
(t) = V
c
cos [ 2t ( fc + f )]
For logic 0 or space input :

fsk
(t) = V
c
cos [ 2t ( fc - f )]

The mark frequency is the higher frequency ( fc + f ) and the space
frequency is the lower frequency ( fc - f ).




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FSK waveform:

data


carrier

Modulated
signal

Applications of FSK:
FSK signaling schemes are used mainly for low-speed digital data
transmissions.
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The simplest form of PSK is binary phase-shift keying (BPSK), where N=1 and
M=2. Binary system encodes single bits and produces two output conditions
possible. One output represents a logic 1, and the other output 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.
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2
N
= M
If N=2, M=4 (4 possible combinations, known as Quarternary PSK (QPSK) or
Quadrature PSK). Bits are combined into groups of two bits called dibits.

If N=3, M=8 (8 possible combinations, known as 8-PSK ).
Bits are combined into groups of three bits called tribits.

If N=4, M=16 (16 possible combinations, known as 16-PSK ).
Bits are combined into groups of four bits called quadbits.


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Pulse modulation is 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.

Pulse modulation is divided broadly into two categories which is
Analog Pulse Modulation (APM) and Digital Pulse Modulation
(DPM).

Analog information such as continuous speech or digital data can be
transmitted using pulse modulation technique. However, for a
continuous waveform, it is first converted into pulses by the process of
sampling.

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Sampling is the process of taking a periodic sample of the continuous
waveform to be transmitted. If sufficient samples are sent, the waveform
can be reconstructed at the receiver.

The Nyquist sampling theorem is used to determine minimum sampling
rate for any signal so that the signal will be correctly restored at the
receiver.

Nyquist sampling theorem states that;
The original information signal can be reconstructed at the receiver with
minimal distortion if the sampling rate in the pulse modulation system is equal
to or greater than twice the maximum information signal frequency.

That is, sampling frequency , fs 2 fm(max)
where fs = sampling frequency
fm(max) = maximum frequency of the modulating signal

The minimum sampling frequency , fs = 2 fm(max)

Figure shows the frequency spectrum of a
modulating waveform such as a voice signal.



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The 3 basic conditions of sampling process are:

1. Sampling at fs = 2 fm(max)
- When the modulating signal is sampled at a minimum sampling frequency, the frequency spectrum is
shown as figure below.



2. Sampling at fs > 2 fm(max)
- This sampling rate creates a guard band between fm(max) and the lowest frequency component
(fs - fm(max) ) of the sampling harmonics.





3. Sampling at fs < 2 fm(max)
- When the sampling rate is less than the minimum value, distortion will occurs. This distortion is called
aliasing or folded over distortion.



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In APM, the carrier is in the form of pulse modulation and the
modulated signal is where one of the characteristics (either
amplitude, width or position) is changed according to the modulating
signal.

The predominant methods of APM are Pulse width modulation
(PWM), Pulse position modulation (PPM) and Pulse amplitude
modulation (PAM).

1. Pulse amplitude modulation (PAM)
-PAM is the simplest form of pulse modulation. It is very similar to
amplitude modulation (AM).

- With PAM, the amplitude of a constant-width and constant position
pulse is varied according to the amplitude of the sample of the analog
signal.

- PAM is not generally used for a complete system but is largely
employed as an intermediate form of modulation with PSK, QAM and
PCM.




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2. Pulse width modulation (PWM)
-PWM also is called as pulse duration modulation (PDM) or pulses
length modulation (PLM). PWM refer to the technique of varying the
width of the constant-amplitude pulse proportional to the amplitude
of the modulating signal.

- PWM gives a better signal to noise performance than PAM. PWM
has a disadvantage, when compared to PPM, that is its pulses are of
varying width and therefore of varying power content.

3. Pulse position modulation (PPM)

-PPM is when the position of a constant-width and constant-
amplitude pulse within prescribed time slot is varied according to the
amplitude of the modulating signal.

-With PPM, the higher the amplitude of the sample, the farther to the
right the pulse is positioned within a prescribed time slot.

- PWM and PPM are seldom used for commercial system. There are
used in special-purpose communication systems such as military
communication systems.
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Digital modulation is the process by which digital symbols are
transformed into waveforms that are compatible with the characteristics
of the channel. In digital modulation code (DPM), a code is used to
represent the amplitude of the samples that has been divided into
various levels.

Advantages of digital system:
1. It is relatively immune to channel noise and interference.
2. Signals and messages can be coded for error detection and correction.
3. It is easier and more efficient to multiplex several digital signals.
4. It can carry a combination of traffics (telephone signals, data, coded
video) if the medium has enough capacity.
5. More economical.

Disadvantages of digital system:
1. The disadvantages of digital system compared to analog system is the
large bandwidth requirement. However, it is no longer a serious problem
because of the advent of large-bandwidth fiber optic systems.

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Pulse code modulation (PCM)

-PCM is a digitally encoded modulation technique that commonly use for
digital transmission.

-With PCM, a group of coded pulses are used to represent the analog
signal. The analog signal is sampled and converted to a fixed-length
serial binary number for transmission.

- PCM is the preferred method of communications because with PCM, it
is easy to combine digitized voice and digital data into a single, high
speed digital signal and propagate it over either metallic or optical fiber
cables





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PCM transmission system
(i) BPF limit the freq of analog input signal to standard voice band freq (300 to 3400Hz)
(ii) Sample and hold samples the input and converts those samples to a multilevel PAM signal
(iii) ADC converts the PAM samples to parallel PCM codes which are converted to serial binary data
in the parallel to serial converter ( serial digital pulses eg 10110001010)
(iv) In the Rx, the serial to parallel converter converts serial pulses received from transmission line to
parallel PCM codes. The DAC converts the parallel PCM codes to multilevel PAM signals.
(v) The hold circuit is basically a low pas filter that converts the PAM signals back to its original
analog form.

-A clock ( an oscillator circuit that outputs timing Pulses) is set to the
sampling frequency
-The sample command must be accurately clocked- cant have varying
sample times. Must be controlled by a clock designed with a highly
accurate crystal oscillator.
1 Sample-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
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.
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Three main processes in PCM transmission are sampling, quantization
and coding.

1. Sampling
- It is a process of taking samples of information signal at a rate of
Nyquists sampling frequency.

2. Quantization
- Quantization is a process of assigning the analog signal samples to
a pre-determined discrete levels.

3. Encoding
- This is the process where each quantized sample is digitally encoded
into n-bits codeword, where;
n = log
2
L where n = number of bits/sample
L = number of quantization levels
or L = 2
n
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010
101
111
101
011
001
0 1 0 1 0 1 1 1 1 1 0 1 0 1 1 0 0 1
Analog input signal
Sampling pulse
Sampled waveform
Quantized signal
PCM pulses
Figure below illustrates an analog waveform signal can be coded into 3 bits
code using single mode for transmission using PCM technique.
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Sampling process
- Multilevel PAM signal

Quantizing and encoding process
- Convert the PAM to parallel codes
Parallel codes is converted to serial data
in parallel to serial converter
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.


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Quantization is a process of assigning the analog samples to a pre-
determined discrete levels.

It also a process of rounding off the amplitude of flat-top samples to a
manageable number of level.

The number of quantization level, L, depends on the number of bits per
sample, n, used to code the signal where;
L = 2
n

The magnitude of the minimum stepsize of the quantization levels is
called resolution, AV. It is equal in magnitude to the voltage of the least
significant bit or the magnitude of the minimum stepsize of the digital to
analog converter (DAC).

The resolution, AV depends on the maximum and minimum voltage of
the information signal, where;
AV =

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Quantization error or quantization noise is the distortion introduced
during the quantization process when the modulating signal is not an
exact value of the quantized level.

Maximum quantization error Q
e(max)
= +

Quantization error can be reduced by increasing the number of
quantization levels, but this will increase the bandwidth required.

Quantization interval or quantum is the magnitude difference between
adjacent step.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.
- the magnitude of a quantum is also called resolution.




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Transmission bit rate, R
- is 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.
sec / bits f n R
s
=
Hz f n TB
s
=
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PCM Line Speed

Line speed is a data rate at which serial PCM bits are clocked
out of the PCM encoder into a transmission line.



Where,
Line speed = the transmission rate in bits per second
Samples/second = sample rate (f
s
)
Bit/sample = number of bits in the compressed PCM code.

Example 3
For a single channel PCM system with a sample rate f
s
= 6000
samples per second and a seven bit compressed of PCM code,
determine the line speed.

Solution

sample
bits
ond
samples
speed line =
sec
bps
sample
bits
ond
samples
speed line
000 42
7
sec
6000
=
=
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OKT 98/KEH215
A PCM transmitter system is transmitting a 6kHz audio signal. The
sampling frequency used is 10% higher than the minimum sampling
rate for the guard-band purpose and the quantization level used is
256.
Determine:

i) the number of bits for every sample,
ii) the sampling frequency rate and
iii) the transmission bit rate (bps).
If the maximum quantization error is 0.01V , what is the resolution
step of the quantization level . Hence determine the peak-to-peak
amplitude of the information signal.
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Example 2
An information signal in a form V
m
(t) = 20 cos(60t x 10
3
t)V to be transmitted through
a binary PCM modulation. The signal is sampled at a rate of 15% higher than the
minimum sampling frequency and the quantization level used is 512. Calculate :

i) The sampling frequency that can be used,
ii) The number of bit per sample,
iii) The transmission bandwidth,
iv) The resolution step and,
v) The quantization error.

If the maximum quantization error is 0.02V, determine the resolution step of the
quantization level and peak to peak amplitude of the information signal.
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Test 2 Oct 2010
Figure Q1e shows the analog waveform to be transmitted using 3 bit PCM
system. Using the ANSWER SHEET Q1e in appendix, shows how the analog
signal can be converted to:

i) Pulse Amplitude Modulation (PAM) signal,
ii) Quantized signal,
iii) A serial binary number (PCM code)
iv) PCM pulses.
(8 marks)

t
3
2
1
-1
0
-2
V(t)
V(t)
t
Analog
signal
Sampling
Pulse
Figure Q1e

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Serial PCM
Code
t
3
2
1
-1
0
-2
V(t)
V(t)
t
3
2
1
-1
0
-2
V(t)
t
3
2
1
-1
0
-2
V(t)
t
Analog
signal
Sampling
Pulse
PAM
Quantized
signal
PCM Pulses t
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