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2

Digital Communication
System Concepts
Vijay K. G a r g 2.1 Digital Communication System .............................................................. 957
Department of Electrical and 2.2 Messages, Characters, and Symbols ......................................................... 957
Computer Engineering, 2.3 Sampling Process ................................................................................. 957
University of Illinois at Chicago,
Chicago, Illinois, USA 2.4 Miasing .............................................................................................. 959
2.5 Quantization ....................................................................................... 960
Yih-Chen Wang 2.6 Pulse Amplitude Modulation ................................................................. 960
Lucent Technologies, 2.7 Sources of Corruption .......................................................................... 961
Naperville, Illinois, USA 2.8 Voice Communication .......................................................................... 963
2.9 Encoding ............................................................................................ 964
2.9.1 EncodingSchemes

2.1 Digital Communication System 2.2 Messages, Characters, and Symbols


Figure 2.1 shows a block diagram of a typical digital c o m m u - When digitally transmitted, the characters are first encoded
nication system. We focus primarily on formatting and trans- into a sequence of bits, called a bit stream or baseband signal.
mission of baseband signal. Data already in a digital format Groups of n bits can be combined to form a finite symbol set
would bypass the formatting procedure. Textual information is or w o r d o f M = 2 ~ for such symbols. A system using a symbol
transformed into binary digits by use of a coder. Analog infor- set size of M is called an M-ary system. The value of n or M
mation is formatted using three separate processes: represents an important initial choice in the design of any
digital communication system. For n = 1, the system is re-
• Sampling
ferred to as binary, the size of symbol set is M = 2, and the
• Quantization
modulator uses two different waveforms to represent the
• Encoding
binary 1 and the binary O. In this case, the symbol rate and
In all cases, the formatting steps result in a sequence of the bit rate are the same. For n = 2, the system is called
binary digits. These digits are transmitted through a baseband quaternary or 4-ary (M = 4). At each symbol time, the
channel, such as a pair of wires or a coaxial cable. However, modulator uses one of the four different waveforms to repre-
before we transmit the digits, we must transform the digits sent the symbol (see Figure 2.2).
into waveforms that are compatible with the channel. For
baseband channels, Compatible waveforms are pulses. The
conversion from binary digits to pulse waveform takes place 2.3 Sampling Process
in a wave encoder also called a baseband modulator. The
output of the waveform encoder is typically a sequence of Analog information must be transformed into a digital format.
pulses with characteristics that correspond to the binary digits The process starts with sampling the waveform to produce a
being sent. After transmission through the channel, the re- discrete pulse-amplitude-modulated waveform (see Figure
ceived waveforms are detected to produce an estimate of the 2.3). The sampling process is usually described in a time
transmitted digits, and then the final step is (reverse) format- domain. This is an operation that is basic to digital signal
ting to recover an estimate of the source information. processing and digital communication. Using the sampling

Copyright © 2005 by Academic Press. 957


All rights of reproduction in any form reserved.
958 Vijay K. Garg and Yih-Chen Wang

Digital
information II
Textual ~ II
informationl - [ Sampler Quantizer Coder WavefOrmencoder
~[Transmitter
I
I I
I I
I_ J

Channel
Binarydigits
Pulsewaveform

Analoginformation Low-passI. I Waveform~ Receiver


q Filter Decoder [
decoder
Textual 9
[

Digitalinformation
FIGURE 2.1 BlockDiagram of a Typical Digital Communication System

1 where g~(t) is the ideal sampled signal and where 8(t - nT~) is
the delta function positioned at time t = nTs.
A delta function is closely approximated by a rectangular
pulse of duration At and amplitude g ( n Ts) / A t; the smaller we
make At, the better will be the approximation:
-1 oo

FIGURE 2.2 Binary and Quaternary Systems gs(t) = f~ ~_, G(f - mr,), (2.2)
m = - ~x~

where G(f) is the Fourier transform of the original signal g(t)


process, we convert the analog signal in a corresponding se- and f~ is sampling rate.
quence of samples that are usually spaced uniformly in time. Equation 2.2 states that the process of uniformly sampling a
The sampling process can be implemented in several ways, the continuous-time signal of finite energy results in a periodic
most popular being the sample-and-hold operation. In this spectrum with a period equal to the sampling rate.
operation, a switch and storage mechanism (such as a transis- Taking the Fourier transform of both side, of Equation 2.1
tor and a capacitor, or shutter and a film strip) form a and noting that the Fourier transform of the delta function
sequence of samples of the continuous input waveform. The 8(t - nT,) is equal to e j2"rrnfFs:
output of the sampling process is called pulse amplitude
modulation (PAM) because the successive output intervals
can be described as a sequence of pulses with amplitudes G~(f) = ~ g(nTs)e j2~nfr,. (2.3)
derived from the input waveform samples. The analog wave- n~--OC

form can be approximately retrieved from a PAM waveform by


Equation 2.3 is called the discrete-time Fourier transform.
simple low-pass filtering, provided we choose the sampling
It is the complex Fourier series representation of the periodic
rate properly. The ideal form of sampling is called instantan-
frequency function C~(t), with the sequence of samples g(nT,)
eous sampling.
We sample the signal g(t) instantaneously at a uniform rate defining the coefficients of the expansion.
We consider any continuous-time signal g(t) of finite
off~ once every T~ sec. Thus, we can write:
energy and infinite duration. The signal is strictly band-limited
with no frequency component higher than W Hz. This implies
ga(t) = ~ g(nTs)8(t- nTs), (2.1) that the Fourier transform G(f) of the signal g(t) has the
11~--0C
property that G(f) is zero for ]fl-> W. If we choose the
2 Digital Communication System Concepts 959

Ts(t)
g(t)

T
FIGURE 2.3 Sampling Process

sampling period T, = 1/2W, then the corresponding spec- width W Hz is called the Nyquist rate and 1/2 W sec is called
trum is given as: the Nyquist interval.
We discuss the sampling theorem by assuming that signal
g(t) is strictly band-limited. In practice, however, an infor-
C~(f) = g (~w)e j~- = fsG(f) + f~ G(f - mf~) mation-bearing signal is not strictly band-limited, with the
m--~, me0 result that some degree of under sampling is encountered.
(2.4) Consequently, some aliasing is produced by the sampling pro-
cess. Aliasing refers to the phenomenon of a high-frequency
Consider the following two conditions: component in the spectrum of the signal seemingly taking on
the identity of a lower frequency in the spectrum of its sampled
(1) G(f) = 0 for If[ > W.
version.
(2) fs = 2W.
We find from equation 2.4 by applying these conditions,
2.4 Aliasing
G(f)=2-~G~(f) - W < f < W. Figure 2.4 shows the part of the spectrum that is aliased due to
(2.5) under sampling. The aliased spectral components represent
.'. G(f) = ~ g e-(L~) - W < f < W. ambiguous data that can be retrieved only under special con-
n~--CXD ditions. In general, the ambiguity is not resolved and ambigu-
ous data appear in the frequency band between (fs -fro) and
Thus, if the sample value g(n/2W) of a signal g(t) is fro.
specified for all n, then the Fourier transform G(f) of the In Figure 2.5, we show a higher sampling rate Jj to eliminate
signal is uniquely determined by using the discrete-time Four- the aliasing by separating the spectral replicas.
ier transform of equation 2.5. Because g(t) is related to G(f) Figures 2.6 and 2.7 show two ways to eliminate aliasing
by the inverse Fourier transform, it follows that the signal g(t) using antialiasing filters. The analog signal is preflltered so
is itself uniquely determined by the sample values g(n/2 W) for that the new maximum frequency fm is less than or equal to
-cxD < n < c~. In other words, the sequence {g(n/2W)} has f J 2 . Thus, there are no aliasing components seen in Figure 2.6
all the information contained in g(t). because f~ > 2f'm. Eliminating aliasing terms prior to sampling
We state the sampling theorem for band-limited signals of is a good engineering practice. When the signal structure is
finite energy in two parts that apply to the transmitter and
receiver of a pulse modulation system, respectively.
lx~(ol
(1) A band-limited signal of finite energy with no fre-
quency components higher than W Hz is completely
described by specifying the values of signals at instants
of time separated by 1/2 W sec.
(2) A band-limited signal of finite energy with no fre-
quency components higher than W Hz may be com-
pletely recovered from a knowledge of its samples
taken at the rate of 2 W samples/sec.
This is also known as the uniform sampling theorem. The
sampling rate of 2 W samples per second for a signal band- FIGURE 2.4 SampledSignal Spectrum
960 Vijay K. Garg and Yih-Chen Wang

trade-off is needed between the cost of a small transition


bandwidth and costs of the higher sampling rate, which are
those of more storage and higher transition rates.
In many systems, the answer has been to make the transition
bandwidth 10 and 20% of the signal bandwidth. If we account
for the 20% transition bandwidth of the antialiasing filter,
we have an engineering version of Nyquist sampling rate:
~f f,>_ZZfm.
fm fs'-fm
Example 3
FIGURE 2.5 Higher Sampling Rate to Eliminate Aliasing
We want to produce a high-quality digitalization of a 20-kHz
bandwidth music signal. The sampling rate of greater than or
equal to 22 ksps should be used.
The sampling rate for compact disc digital audio player is
44.1 ksps, and the standard sampling rate for studio-quality
audio player is 48 ksps.

2.5 Quantization
,f
f'm fs-f'm t~
In Figure 2.8, each pulse is expressed as a level from a finite
number of predetermined levels; each such level can be repre-
FIGURE 2.6 Prefiltering to Eliminate Aliasing
sented by a symbol from a finite alphabet. The pulses in Figure
2.8 are called quantized samples. When the sample values are
quantized to a finite set, this format can interface with a digital
IX~(f)l
system. After quantization, the analog waveform can still be
recovered but not precisely; improved reconstruction fidelity
of the analog waveform can be achieved by increasing the
i
number of quantization levels (requiring increased system
1 bandwidth).

I,

fn,'
2.6 Pulse Amplitude Modulation
td2
There are two operations involved in the generation of the
FIGURE 2.7 Postfiltering to Eliminate Aliasing Portion of the
pulse amplitude modulation (PAM) signal:
Spectrum
(1) Instantaneous sampling of the message signal m(t)
well known, the aliased terms can be eliminated after sampling every Ts sec, where fs = 1/Ts is selected according to
with a linear pass filter (LPF) operating on the sampled data. the sampling theorem
In this case, the aliased components are removed by postfilter- (2) Lengthening the duration of each sample obtained to
ing after sampling. The filter cutoff frequency f'm removes the some constant value T
aliased components; i'm needs to be less than (f~-fm). It
should be noted that filtering techniques for eliminating the
aliased portion of the spectrum will result in a loss of some g(t)
signal information. For this reason, the sample rate, cutoff
bandwidth, and filter type selected for a particular signal
bandwidth are all interrelated.
Realizable filters require a nonzero bandwidth for the tran-
sition between the passband and the required out-of-band
attenuation. This is called the transition bandwidth. To min- t
imize the system sample rate, we desire that the antialiasing -*IT
filter has a small transition bandwidth. Filter complexity and
cost rise sharply with narrower transition bandwidth, so a FIGURE 2.8 Flattop Quantization
2 Digital Communication System Concepts 961

h(t)
1.0 f)l
ctrummagnitude

0 -3/T -1/T 0 1/T 3/T


Pulse arg[H(f)]

3/T~~_ _~I_/T., j Spectrumphase


- 3/T

FIGURE 2.9 Rectangular Pulse and Its Spectrum

These two operations are jointly referred to as sample and (1) Assuming a sampling rate of 8 kHz, calculate the
hold. One important reason for intentionally lengthening the spacing between successive pulses of the multiplexed
duration of each sample is to avoid the use of an excessive signal.
channel bandwidth because bandwidth is inversely propor- (2) Repeat your calculations using Nyquist rate sampling.
tional to pulse duration.
The Fourier transform of the rectangular pulse h(t) is given 106
as (see Figure 2.9): T~ -- - - -- 125 Ixs.
8000

H(f) = Tsinc(fT)e -j2"~fr. (2.6) For 25 channels (24 voice channels +1 sync), time allocated for
each channel is 125/25 = 5 b~s. Since the pulse duration is
We observe that by using flattop samples to generate a PAM 1 izs, the time between pulses is (5 - 1) = 4 b~s.
signal, we introduce amplitude distortion as well as a delay of The Nyquist rate is 7.48 Hz (2.2 x 3.4).
T/2. This effect is similar to the variation in transmission In addition:
frequency that is caused by the finite size of the scanning
aperture in television. The distortion caused by the use of 106
Ts -- -- 134 p~s.
PAM to transmit an analog signal is called the aperture affect. 7480
This distortion may be corrected by using an equalizer
134
(see Figure 2.10). The equalizer has the effect of decreasing Tc -- -- 5.36 Izs.
25
the in-band loss of the filter as the frequency increases in such a
manner to compensate for the aperture effect. For T/Ts < O.1,
The time between pulses is 4.36 Vs.
the amplitude distortion is less than 0.5%, in which case the
need of equalization may be omitted altogether.

Example 4 2.7 Sources of Corruption


Sampled uniformly and then time-division multiplexed are 24
The sources of corruption include sampling and quantization
voice signals. The sampling operation involved flattop samples
effects as well as channel effects, as described in the following
with 1 Ixs duration. The multiplexing operation includes pro-
bulleted list.
vision for synchronization by adding an extra pulse of suffi-
cient amplitude and also 1 Ixs duration. The highest frequency • Quantization noise: The distortion inherent in quantiza-
component of each voice signal is 3.4 kHz. tion is a roundoff or truncation error. The process of

PAMsignal ]
s(t) IlL Filter(LPF)I~ [
EqualizerJ . Messagesignalm(t)
FIGURE 2.10 An Equalizer Application
962 Vijay iv(. Garg and Yih-Chen Wang

encoding the PAlVl waveform into a quantized wave- symbol duration and cause signal pulses to overlap. This
form involves discarding some of the original analog overlapping is called inter-symbol interference ISI), ISI
information. This distortion is called quantization causes system degradation (higher error rates); it is a
noise; the amount of such noise is inversely propor- particularly insidious form of interference because raising
tional to the number of levels used in the quantization the signal power to overcome interference will not im-
process. prove the error performance:
• Quantizer saturation: The quantizer allocates L levels to
the task of approximating the continuous range of inputs q/2 q/2
with a finite set of outputs (see Figure 2.11). The range of 0-2= Ie2p(e)de= Ie21de=q2
inputs for which the difference between the input and q 12
output is small is called the operating range of the
-q/2 -q/2
converter. If the input exceeds this range, the difference = average quantization noise power.
between the input and output becomes large, and we say
that the converter is operating in saturation. Saturation
errors are more objectionable than quantizing noise. Gen-
Vd z (V_~)2 (~)2_ L2q2~ (2.7)
erally, saturation is avoided by use of automatic gain
control (AGC), which effectively extends the operating
range of the converter. ~ q q2/12 -- 3L2. (2.8)
• Timing jitter: If there is a slight jitter in the position of
the sample, the sampling is no longer uniform. The effect In the limit as L ---, oc, the signal approaches the PAM format
of the jitter is equivalent to frequency modulation (FM) (with no quantization error) and signal-to-quantization noise
of the baseband signal. If the jitter is random, a low-level ratio is infinite. In other words, with an infinite number of
wideband spectral contribution is induced whose proper- quantization levels, there is zero quantization error.
ties are very close to those of the quantizing noise. Timing Typically L = 2R, R = Log2L, and q = G- 2vp = (2Vp)/2 R.
jitter can be controlled with very good power supply
isolation and stable clock reference.
• Channel noise: Thermal noise, interference from other 1"2Vp\ 2 ---=_1V2 2_2R
users, and interference from circuit switching transients
can cause errors in detecting the pulses carrying the
Let P denote the average power of the message signal re(t),
digitized samples. Channel-induced errors can degrade
and then:
the reconstructed signal quality quite quickly. The rapid
degradation of the output signal quality with channel-
induced errors is called a threshold effect. (SNR)o -- ~ -- 22R. (2.9)
° Intersymbol interference: The channel is always band- \ P/
limited. A band-limited channel spreads a pulse wave-
form passing through it. When the channel bandwidth is The output SNR of the quantizer increases exponentially
much greater than pulse bandwidth, the spreading of the with increasing number of bits per sample, R. An increase
pulse will be slight. When the channel bandwidth is close in R requires a proportionate increase in the channel band-
to the signal bandwidth, the spreading will exceed a width.

Example 5
We consider a full-load sinusodial modulating signal of ampli-
tude A that uses all representation levels provided. The average
signal power is (assuming a load of 1 fl):

L levels
1 The equations are written and solved as follows:

P ~--.
A2
2

L 0-2 = fA22 -2R.

(SNR) o
3

m
2
A
(I/3A22_2R)
2 _~
(22R) = 1.8 + 6R dB.
FIGURE 2.11 Uniform Quantization
2 Digital Communication System Concepts 963

L R [bits] SNR [decibels] In equation 2.10, IX is constant, x and y are the input
32 5 31.8 and output voltages, ~ = 0 represents uniform quantization,
64 6 37.8 and ix = 255 is the standard value used in North America.
128 7 43.8
• A-Law, used in Europe, is as follows:
256 8 49.8
y _ A(lx[/Xm~)sgnx, 0 < Ixl < 1 (2.11a)
yma× 1 + In A Xma~- A
2.8 V o i c e C o m m u n i c a t i o n 1 + ln(lx[/Xmax) 1 Ixl
< 1. (2.11b)
= 1 + in A sgnx, A < Xma~
For most voice c o m m u n i c a t i o n , very low speech volumes
predominate; about 50% of the time, the voltage characteriz- The A is the positive constant, and A = 87.6 is the standard
ing detected speech energy is less than 1/4 of the root-mean- value used in Europe.
square (rms) value. Large amplitude values are relatively rare;
only 15% of the time does the voltage exceed the rms value. Example 6
The quantization noise depends on the step size. When the The information in an analog waveform with maximum fre-
steps are uniform in size, the quantization is called the uniform quency fm= 3 kHz is transmitted over an M-ary PCM system,
q u a n t i z a t i o n . Such a system would be wasteful for speech where the number of pulse levels is M = 32. The quantization
signals; many of the quantizing steps would rarely be used. In distortion is specified not to exceed 4- 1% of the peak-to-peak
a system that uses equally spaced quantization levels, the analog signal.
quantization noise is same for all signal magnitudes. Thus, (1) What is minimum number of bits/sample or bits/
with uniform quantization, the signal-to-noise ratio (SNR)
PCM word that should be used?
is worse for low-level signals than for high-level signals. Non- (2) What is minimum sampling rate, and what is the
uniform q u a n t i z a t i o n can provide fine quantization of the
resulting transmission rate?
weak signals and coarse quantization of the strong signals.
(3) What is the PCM pulse or symbol transmission rate?
Thus, in the case of nonuniform quantization, quantization
noise can be made proportional to signal size. Improving the Solutions:
overall SNR by reducing the noise for predominant weak
signals, at the expense of an increase in noise, can be done for lel _<pyre,
rarely occurring signals. The nonuniform quantization can
be used to make the SNR a constant for all signals within the where p is fraction of the peak-to-peak analog voltage•
input range. For voice, the signal dynamic range is 40 dB.
N o n u n i f o r m q u a n t i z a t i o n is achieved by first distorting the • le axl- Vpp
original signal with logarithmic compression characteristics 2L '
and then using a uniform quantizer. For small magnitude
signals, the compression characteristics have a much steeper •

\2t
< pv,,).
slope than the slope for large magnitude signals. Thus, a given -

signal change at small magnitudes will carry the uniform


1
quantizer through more steps than the same change at large • 2R:L>--.
-- 2p
magnitudes. The compression characteristic effectively changes
the distribution of the input signal magnitude so there is no I
preponderance of low-magnitude signals at the output of the 2R > -- 50,
- 2 × 0.01
compressor. After compression, the distorted signal is used as
an input to a uniform quantizer. At the receiver, an inverse .'.(R>5.64) useR=6.
compression characteristic, called expansion, is used so that
the overall transmission is not distorted. The whole process • f~ = 2fro = 6000 samples/sec.
(compression and expansion) is called c o m p a n d i n g .
fs = 6 x 6000 = 36 kbps.
• The ~-Law, used in North America, is as follows:
y In [1 + Ixl/xmax] • M = 2/, = 32.
ym~x in [1 + ix] sgnx. (2•10)
b = 5 bits/symbol.
sgnx = 1, x _ 0.
36000
sgnx = - 1 , x < 0. Rs-- 5 -- 7200 symbols/sec.
964 Vijay K. Garg and Yih-Chen Wang

2.9 Encoding 2.9.1 Encoding Schemes


The following encoding schemes are often used.
Codeword t i m e s l o t s are shown in Figure 2.12 in which the
codeword is 4-bit representation of each quantized sample. In (1) N o n r e t u r n to Zero-Level (NRZ-L)
the bit duration p o r t i o n o f Figure 2.12, each binary 1 is repre- • 1 = high level
sented by a pulse, and each binary 0 is represented by the • 0 = low level
absence of a pulse. (2) N o n r e t u r n to Z e r o - M a r k (NRZ-M)
If we increase the pulse width to the m a x i m u m possible • 1 = transition at the beginning of interval
(equal to bit duration, t), we have the waveform shown in • 0 = no transition
the + V and - V b o t t o m p o r t i o n of Figure 2.12. Rather than (3) N o n r e t u r n to Zero-Space (NRZ-S)
describe this waveform as a sequence of present or absent • 1 = no transition
pulses, we can describe it as a sequence of transitions between • 0 = transition at the beginning of the interval
two levels. W h e n the waveform occupies the upper voltage (4) R e t u r n to Zero (RZ)
level, it represents a b i n a r y 1; when it occupies the lower • 1 = pulse in first half of bit interval
voltage, it represents a binary 0. • 0 = no pulse
We need an e n c o d i n g process to translate the discrete sets of (5) Biphase-level (Manchester)
sample value to a more appropriate form of signal. Any plan to • 1 = transition from high to low in middle of interval
represent each of the discrete sets of value as a particular • 0 = transition from low to high in middle of interval
arrangement of discrete events is called a code. One of the (6) B i p h a s e - M a r k
discrete events in a code is called a code s y m b o l or symbol. A • Always a transition at the beginning of interval
particular arrangement of symbols used in a code to represent a • 1 = transition in middle of interval
single value of the discrete set is called a c o d e w o r d or character. • 0 = no transition in middle of interval
Most c o m m o n l y used pulse code modulation (PCM) wave- (7) B i p h a s e - s p a c e
forms are classified into the following groups: • Always a transition at the beginning of interval
• 1 = no transition in middle of interval
• Nonreturn to zero (NRZ) • 0 = transition in middle of interval
• Return to zero (RZ) (8) Differential Manchester
• Phase-encoded • Always a transition in middle of interval
• Multilevel binary • 1 = no transition at the beginning of interval
The reason for the large selection relates to the differences in • 0 = transition at beginning of interval
performance that characterize each waveform. In selecting a (9) Delay m o d u l a t i o n (Miller)
coding scheme for a particular application, some of the par- • 1 = transition in middle of interval
ameters worth examining are: • 0 = no transition if followed by 1, or transition at
end of interval if followed by 0
• The dc c o m p o n e n t (10) Bipolar
• Self-clocking • 1 = pulse in first half of interval, alternating polarity
• Error detection from pulse to pulse
• Bandwidth compression • 0 = no pulse
• Noise i m m u n i t y
• Biphase level (Manchester code)

Code word time slot


I. 1 0 1 1 "1
I 1 0 0
Bit duration I

N" I
I
H I
I
+v

-V

FIGURE 2.12 Bit Sequence and Waveform

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