Beruflich Dokumente
Kultur Dokumente
Preface
1
1.1
1.2
2
2
3
4
7
1.3
1.4
1.5
xi
Introduction
Non-electrical telecommunication
1.2.1 Verbal non-electrical telecommunication
1.2.2 Visual non-electrical telecommunication
Modern telecommunication
1.3.1 Developments in binary codes for data
transmission
1.3.2 Historical systems and developments in
modern telecommunication
Communication system elements
1.4.1 Information source
1.4.2 Information sink
1.4.3 Transmitter
1.4.4 Receiver
1.4.5 Transmission medium
Classification of modern communication systems
1.5.1 Simplex or duplex?
1.5.2 Analogue or digital?
1.5.3 Baseband or modulated?
7
12
16
17
20
26
28
29
46
46
47
53
Telecommunication signals
74
2.1
2.2
74
Introduction
Forms and classification of telecommunication
signals
2.2.1 Subjective classification
2.2.2 Objective classification
75
77
83
vi
Contents
2.3
2.4
2.5
2.6
2.7
2.8
2.9
Sinusoidal signals
2.3.1 Qualitative introduction
2.3.2 Parameters of a sinusoidal signal
2.3.3 Addition of sinusoids
2.3.4 Multiplication of sinusoids
Frequency content of signals
2.4.1 Fourier theorem
2.4.2 Types of spectrum
2.4.3 Fourier transform examples
2.4.4 Bandwidth and frequency response
2.4.5 Inverse relationship between time and
frequency
Power content of signals
Logarithmic measures of signal and power
2.6.1 Logarithmic units for system gain
2.6.2 Logarithmic units for voltage and power
2.6.3 Logarithmic unit for noise power
Calibration of a signal transmission path
Transmission through linear systems
Signal distortion
2.9.1 Distortionless transmission
2.9.2 Attenuation and delay distortions
2.9.3 Non-linear distortions
88
88
90
95
97
100
100
107
111
113
116
118
123
125
126
127
128
130
134
134
136
138
Amplitude modulation
149
3.1
3.2
149
150
151
152
154
3.3
3.4
3.5
3.6
Introduction
AM signals time domain description
3.2.1 AM waveform
3.2.2 Sketching AM waveforms
3.2.3 Modulation factor
Spectrum and power of amplitude-modulated
signals
3.3.1 Sinusoidal modulating signal
3.3.2 Arbitrary message signal
3.3.3 Power
AM modulators
3.4.1 Generation of AM signals
3.4.2 AM transmitters
AM demodulators
3.5.1 Diode demodulator
3.5.2 Coherent demodulator
3.5.3 AM receivers
Merits, demerits and applications of AM
158
158
162
165
168
168
172
174
174
179
181
185
Contents
3.7
vii
Variants of AM
3.7.1 DSB
3.7.2 SSB
3.7.3 ISB
3.7.4 VSB
186
186
196
204
207
Angle modulation
217
4.1
4.2
218
218
221
224
227
234
234
236
241
242
252
264
264
267
268
273
273
276
279
279
283
283
285
286
287
288
288
289
289
4.3
4.4
4.5
4.6
4.7
4.8
Introduction
Basic concepts of FM and PM
4.2.1 Frequency modulation concepts
4.2.2 Phase modulation concepts
4.2.3 Relationship between FM and PM
FM and PM waveforms
4.3.1 Sketching simple waveforms
4.3.2 General waveform
Spectrum and power of FM and PM
4.4.1 Narrowband FM and PM
4.4.2 Wideband FM and PM
FM and PM modulators
4.5.1 Narrowband modulators
4.5.2 Indirect wideband modulators
4.5.3 Direct wideband modulators
FM and PM demodulators
4.6.1 Direct demodulator
4.6.2 Indirect demodulator
4.6.3 Phase demodulator
4.6.4 Frequency discriminators
FM transmitter and receiver
4.7.1 Transmitter
4.7.2 SNR and bandwidth trade-off
4.7.3 Pre-emphasis and de-emphasis
4.7.4 Receiver
Overview of FM and PM features
4.8.1 Merits
4.8.2 Demerits
4.8.3 Applications
Sampling
294
5.1
5.2
5.3
294
295
296
297
298
300
Introduction
Sampling theorem
Proof of sampling theorem
5.3.1 Low-pass signals
5.3.2 Bandpass signals
5.3.3 Sampling at Nyquist rate
viii
Contents
5.4
5.5
5.6
Aliasing
Anti-alias filter
Non-instantaneous sampling
5.6.1 Natural sampling
5.6.2 Flat-top sampling
5.6.3 Aperture effect
301
307
310
310
313
317
323
6.1
6.2
323
324
325
333
356
6.3
6.4
6.5
6.6
6.7
Introduction
Quantisation and encoding
6.2.1 Uniform quantisation
6.2.2 Non-uniform quantisation
Differential PCM (DPCM)
6.3.1 Adaptive differential pulse code modulation
(ADPCM)
6.3.2 Delta modulation
Low bit rate speech coding
6.4.1 Waveform coders
6.4.2 Vocoders
6.4.3 Hybrid coders
Line codes
6.5.1 NRZ codes
6.5.2 RZ codes
6.5.3 Biphase codes
6.5.4 RLL codes
6.5.5 Block codes
Pulse shaping
6.6.1 Ideal filtering
6.6.2 Raised cosine filtering
6.6.3 Information capacity theorem
Digital baseband receiver
6.7.1 Adaptive equalisation
6.7.2 Matched filter
6.7.3 Root raised cosine filter
6.7.4 Clock extraction
6.7.5 Eye diagrams
6.7.6 Matched filter worked examples
359
359
365
368
369
371
372
373
374
375
375
377
381
381
383
387
389
389
390
392
393
394
395
407
7.1
7.2
7.3
7.4
408
411
421
423
Introduction
Signal space
Digital transmission model
Noise effects
Contents
7.5
ix
425
428
432
437
437
439
441
444
445
446
447
448
451
454
456
457
460
460
462
468
474
477
482
485
Multiplexing strategies
495
8.1
8.2
495
500
500
503
505
515
517
518
518
521
531
539
547
548
550
553
558
8.3
8.4
Introduction
Frequency division multiplexing
8.2.1 General concepts
8.2.2 Demerits of flat-level FDM
8.2.3 FDM hierarchies
8.2.4 Wavelength division multiplexing
8.2.5 The future of FDM
Time division multiplexing
8.3.1 General concepts
8.3.2 Plesiochronous digital hierarchy
8.3.3. Synchronous digital hierarchy
8.3.4 ATM
Code division multiplexing
8.4.1 Types of spread spectrum modulation
8.4.2 CDM transmitter
8.4.3 CDM receiver
8.4.4 Crucial features of CDM
Contents
569
9.1
9.2
569
571
571
572
573
575
575
575
576
579
579
579
579
582
584
587
588
590
593
594
594
596
597
600
604
606
607
608
609
610
611
614
9.3
9.4
9.5
9.6
9.7
9.8
9.9
Introduction
Signal attenuation
9.2.1 Metallic lines
9.2.2 Optical fibre
9.2.3 Radio
Physical sources of random noise
9.3.1 Thermal or Johnson noise
9.3.2 Quantisation noise
9.3.3 Radio or sky noise
9.3.4 Shot noise
9.3.5 Partition noise
9.3.6 Quantum noise
9.3.7 Flicker or 1/f noise
Gaussian noise
White noise
Narrowband noise
9.6.1 Noise equivalent bandwidth
9.6.2 Canonical representation
9.6.3 Envelope representation
System noise calculations
9.7.1 Available noise power
9.7.2 Equivalent noise temperature
9.7.3 Noise figure of a single system
9.7.4 Noise figure of cascaded systems
9.7.5 Overall system noise temperature
9.7.6 Signal-to-noise ratio
Noise effects in analogue communication systems
9.8.1 DSB
9.8.2 SSB
9.8.3 AM
9.8.4 FM
Noise effects in digital communication systems
Appendices
A
B
C
Index
Trigonometric identities
Transmission of ASCII-coded data
Further reading and tables
620
622
624
633
Preface
Minimal mathematics
The essential approach taken is to impart a thorough grounding in the
fundamental concepts and design issues involved in communication engineering using minimal mathematics. Most of the existing textbooks on
communication engineering take a mathematical systems approach that
requires the student to jump over a mathematical high-hurdle before being
able to enjoy what is a thoroughly exciting subject and a highly rewarding
career. Such a needlessly mathematical approach may deter potentially
successful engineers or cloud their insight into the underlying engineering
concepts. At the other end of the scale are those textbooks that are unsuitable to the intended readership of Communication Engineering Principles
because they are written to educate non-technical and non-mathematical
readers on the terminology and basic concepts of communication
engineering.
xii
Preface
Lecturer support
Lecturer support is available in the form of a Solutions Manual, available
from the publisher. In addition, all the figures from the book are available
as PowerPoint slides, accessible from the Publishers Web site.
Preface l
xiii
modulation techniques, the design considerations, and circuits for modulation and demodulation.
Chapter 5 presents a lucid and comprehensive discussion of sampling and
lays an important foundation for the introduction of digital transmission
techniques. Topics treated include the sampling of low pass and bandpass
signals, anti-alias filter design, natural and flat-top sampling, and aperture
distortion.
Chapter 6 covers digital baseband transmission, and includes a wide range
of techniques for the digital representation of analogue signals, line coding,
pulse shaping to combat intersymbol interference, digital baseband receiver
operations and the matched filter. In particular, the chapter features a
detailed discussion of A-law and -law pulse code modulation (PCM) and a
review of low bit rate speech coding techniques.
Chapter 7 on digital modulated transmission continues the discussion of
digital communication with a detailed treatment of the major digital
modulation techniques. An important feature is the use of a simple geometrical approach to obtain the bit error rate (BER) of an arbitrary binary transmission system, which is then extended to various M-ary transmissions.
Crucial information is provided to guide the systems designer on the choice
of modulation technique, bit rate, transmission bandwidth, signal power and bit
error rate.
Chapter 8 deals with multiplexing strategies for multi-user communication
systems and includes up-to-date information on various international telecommunications standards. Topics discussed in detail include frequency
division multiplexing (FDM), time division multiplexing (TDM), the plesiochronous and synchronous digital hierarchies, SONET, ATM and code division
multiplexing (CDM).
Chapter 9 is the final chapter of the book and covers the evaluation of the
impact of noise in digital and analogue communication systems, with
several illustrative examples from satellite communications. The chapter also
includes a discussion of signal attenuation in various transmission media
and a lucid treatment of white and narrowband random noise.
Acknowledgements
I would like to thank Dr John Brush (University of Dundee), Mr Henry
Green (Manchester Metropolitan University) and Dr Noel Evans (University of Ulster) for painstakingly reviewing the entire book and making
many constructive suggestions for improvement. Their positive comments
were an important source of inspiration. I wish to express my gratitude to
my editor Christopher Glennie for his professional support throughout the
writing of the book, and to Jacqueline Harbor for her enthusiastic editorial
work and encouragement during the crucial early stages of the project.
I would also like to thank my wife Buchi for her unflinching and selfless
support. She patiently engaged in countless discussions of different renderings of some sections of the book and made useful suggestions from her
literary arsenal. My thanks also go to our children, Ifiok (12), Andikan (11),
Yama (11), Iniekem (9) and Sara (6) for their keen interest in seeing the
xiv
Preface
Eb
N0
389
2h 1
h 0
h
= Lim
B
= 2h ln 2
h =0
= ln 2 = 0.69315
In the above, the second line is obtained by taking the derivatives (with
respect to h) of the numerator and denominator of the expression
under the limit operator. This limiting value of Eb/N0 is referred to as
the Shannon limit.
4.
6.7
Finally, we must point out that the theorem merely tells us that it is
possible to have error-free transmission at the channel capacity, but
does not show us how to design such a system. In practice, all digital
transmission systems both modulated and baseband fall short of
achieving the specified maximum data rate for a given bandwidth and
S/N. The theorem is nevertheless a useful yardstick against which the
performance of practical systems can be measured.
Transmitter
Channel
Ht(f )
Data in
Line
coder
d(t)
Transmit
filter
Hc(f )
Transmission
line
Receiver
He(f )
Hr (f )
Equaliser
Receive
filter
P(f )
To decision
circuit
Pulse generator
Noise and
distortion
Figure 6.30
(6.75)
390 l
where He(f) is the transfer function of the equaliser. Note that Eq. (6.75)
specifically excludes propagation delay t, which can be accounted for by
inserting the factor exp(j2pft) in the RHS.
In most cases, as in the PSTN, the channel characteristic and hence Hc(f)
is time-varying in a non-deterministic way. The equalisation then has to be
adaptive, with He(f) constantly and automatically adjusting itself to satisfy
Eq. (6.75). One way of achieving this is by using a tapped-delay-line filter.
See Fig. 6.15. In this case, however, the tap gains are constantly adjusted to
minimise the mean square error between expected and received pulse
shapes averaged over a number of sampling instants. This can be done with
the help of a training sequence of bits transmitted prior to the information
bits. The optimum values computed for the tap gains are then maintained
for the duration of the call on the assumption that the channel does not
change significantly during this (short) period.
2.
The gain response |Hr(f)| of the filter should not necessarily be flat
within its passband. Rather, it should be such that the filter attenuates
the white noise significantly at those frequencies where G(f) is small
since these frequencies contribute little to the pulse energy. And the
filter should boost those frequencies at which G(f) is large in order to
maximise the output pulse energy. We see then that the filter should be
tailored to the incoming pulse, with a gain response that is small where
G(f) is small and large where G(f) is large. The perfect way of doing this
G(f)
Hr (f )
Figure 6.31
391
Filter action.
3.
(6.76)
We complete the filter specification by considering phase requirements. The instantaneous signal power at the decision instant t = Ts
(equal to the pulse duration) will be maximum if all frequency components in the pulse have been delayed by the same amount. This means
that the only complex term in the spectrum of the filter output P(f)
must be the factor exp(j2pfTs), since this ensures the same delay Ts for
all frequencies. See our discussion of distortionless transmission in
Section 2.9.1 if in doubt. Therefore, in view of Fig. 6.31, we may write
P(f ) = G(f )Hr (f ) R(f )exp(j2p fTs )
The left-hand side (LHS) of this equation is real, and so must be the
RHS, which is the product of two complex functions one of them in
square brackets. Since the product of two complex functions is real if
and only if the two functions are related as complex conjugates, it
follows that
Hr (f ) = G (f )exp(j2p fTs )
(6.77)
(6.78)
That is, the impulse response of the matched filter is a time-reversed and
delayed version of the pulse g(t).
This is a good point to study Section 6.7.6 where we employ several
worked examples to introduce important concepts, including the convolution integral, the irrelevance of pulse shapes to a matched filter and the
correlation receiver. Wherever possible, the simplicity of a graphical
392 l
(6.79)
where Hr(f) is the transfer function of the receive filter and we require P(f) to
be a raised cosine spectrum. Equation (6.79) is an important result, which
shows that to minimise ISI the transmit and receive filters must be designed
so that the product of their transfer functions is a raised cosine spectrum.
To satisfy this requirement as well as that of matched filtering for optimum
performance in the presence of white noise, we must have
| H t (f ) | = | Hr (f ) | = raised cosine spectrum
| f | f1
1,
| f | -f1
1
=
1 + cos p
, f1 f f2
f2 - f1
2
0
f > f2
(6.80)
where f1 and f2 are given as before by Eq. (6.67). Note that a scale factor
1 / Rs has been ignored in the transfer function. This means that using a
pair of root raised cosine filters, one located at the transmitter and the other
at the receiver, allows us to eliminate ISI and at the same time obtain
optimum performance in the presence of white noise. This is a notable
achievement!
One final observation that must be made concerns the effect of the
equaliser on our specification of the receive filter. In general, channel
attenuation increases with frequency. The equaliser must therefore have a
gain that increases with frequency in keeping with Eq. (6.75). The result is
that the noise reaching the receive filter, having passed through the
equaliser, is no longer white but coloured with an amplitude spectrum
that increases with frequency. Under this condition, SNR can be maximised
by arranging for the receiver to attenuate the higher frequencies in order to
reduce noise. This also unavoidably attenuates the desired pulse energy at
these frequencies. So we compensate for this by designing the transmit
filter to proportionately boost the high-frequency components of the
pulse. RZ codes serve the same aim by reducing the pulse widths by a factor
393
Received
symbol stream
Non-linear
device
fs/2 etc.
Figure 6.32
fs etc.
Clock extraction.
BPF or PLL
fs
Clock
Threshold signal
comparator T
fs only
s
394 l
The width of the eye opening gives the timing error that can be tolerated in the sampling instants at the receiver. The best sampling instant
is at the centre of the eye opening.
2.
3.
The height of the eye opening gives the noise margin of the system.
It is therefore obvious that the larger the eye opening the lower will be the
symbol error rate of the system. Figure 6.34 demonstrates the impact of
noise and timing error on the eye diagram of a binary system that uses
raised cosine-filtered pulses. A narrowing of the eye opening by these
effects clearly indicates an increased probability of error. The eye diagram is
indeed a very useful diagnostic tool for checking for the presence of timing
error, noise and pulse distortion in a digital baseband system.
(a)
t
(b)
Bit stream
Threshold level
Figure 6.33
diagram.
Noise only
Figure 6.34
395
WORKED EXAMPLE
6.7
Time-reversal and delay
Sketch the impulse response of a matched filter for receiving the pulse g(t)
shown in Fig. 6.35(a), where the pulse duration Ts = 5 s.
The required impulse response is given by Eq. (6.78) and can be obtained
in two steps. First, the pulse g(t) is time-reversed to give g(t). Then, g(t)
is delayed by Ts to give g(Ts t), which is the required impulse response.
The waveforms g(t) and g(Ts t) are sketched in Figs. 6.35(b) and (c). It is
important to understand how these two waveforms are obtained.
Observe that the waveform of g(t) may be obtained simply by flipping
396 l
1.0
(a)
0.5
t (s)
0
g(t)
1.0
(b)
0.5
t (s)
5
g(Ts t)
1.0
Ts = 5 s
(c)
0.5
t (s)
5
Figure 6.35
g(t) horizontally about t = 0, and that the waveform g(Ts t) results from
delaying g(t) by a time Ts. In this case, since g(t) starts at the time t =
5 s, it follows that g(Ts t) must start at a time Ts (= 5 s) later, which
is therefore the time t = 0.
Table 6.10 provides verification of the above procedures. Noting that
g(ti) is the value of the pulse g(t) at t = ti, it follows by definition of g(t) in
Fig. 6.35(a) that g(10) = 0, g(5) = 0, g(1) = 1, g(4) = 0.5, g(10) = 0 and so
on, where t is in s. Table 6.10 gives values of the waveforms g(t), g(t)
and g(Ts t) at various values of t. For example, at t = 4, g(t) = g(4) = 0.5 (by
definition); g(t) = g(4) = 0 (by definition); and g(Ts t) = g(5-4) = g(1) = 1
(by definition). Plotting the entries of this table leads to Fig. 6.35, with
column 3 plotted against column 1 to give Fig. 6.35(b) and column 4
plotted against column 1 to give Fig. 6.35(c).
Table 6.10
WORKED EXAMPLE
397
Worked example 6.7. Entries plotted in Fig. 6.35. Note that Ts = 5 ms.
t (ms)
g(t)
g(t)
hr(t) = g(Ts t)
10
= g(10) = 0
= g(5) = 0
= g(5 5) = g(10) = 0
= g(4) = 0.5
= g(5 4) = g(9) = 0
= g(3) = 1
= g(5 3) = g(8) = 0
= g(2) = 1
= g(5 2) = g(7) = 0
= g(1) = 1
= g(5 1) = g(6) = 0
= g(0) = 0
= g(5 0) = g(5) = 0
= g(1) = 0
= g(2) = 0
= g(5 2) = g(3) = 1
= g(3) = 0
= g(5 3) = g(2) = 1
0.5
= g(4) = 0
= g(5 4) = g(1) = 1
= g(5) = 0
= g(5 5) = g(0) = 0
10
= g(10) = 0
6.8
Convolution integral
Determine the output pulse g0(t) that is obtained at the receiver when the
transmitted pulse g(t) in the previous Example (Fig. 6.35a) is detected using
a matched filter.
We know (from Section 2.8 and Fig. 6.31) that a filter of transfer function
H(f) and corresponding impulse response h(t) processes an input signal
g(t) with spectrum G(f) to give an output signal g0(t) whose spectrum is given by
Go (f ) = G(f )H (f )
g (t )h(t t )dt
(6.81)
Eq. (6.81) defines the convolution integral, which states that g0(t) is given
at each time instant t by the total area under the function g(t)h(t t),
398 l
h(t ) g (t t )dt
2.
3.
The value of the output pulse g0(t) at a time t is the area under the
curve of g(t)h(t t). For example, it can be seen in Fig. 6.36(d) that
the area under the curve of g(t)h(7 t) is 1.5, which means that g0(t)
= 1.5 at t = 7 s.
4.
Note that the matched filter has distorted the transmitted pulse g(t) in
such a way that the maximum value of the output pulse g0(t) occurs at
the decision instant t = Ts. It can be seen in Fig. 6.36(c) that h(Ts t) =
g(t). Since the pulse g(t) is a real signal, it follows from Eq. (6.81) that,
g 0 (Ts ) =
=
g (t )h(Ts t )dt =
| g (t ) |2 dt
g (t ) g (t )dt
(6.82)
E Energy of signal g (t )
Thus g0(Ts) is the energy of the transmitted pulse g(t), assuming of course
that the gain of the matched filter is normalised to unity and the effect of
the transmission channel has been equalised according to Eq. (6.75). The
matched filter in this example can be approximated using an integrate-
399
g(t)
1.0
(a)
0.5
-5
-4
-3
-2
-1
t (ms)
h(t)
1.0
(b)
0.5
-5
-4
-3
-2
-1
t (ms)
(c)
h(-2 -t)
h(-t)
h(2-t)
h(5-t)
1.0
h(10-t)
h(7-t)
0.5
-5
-7
-4
-3
-2
-1
t (ms) 7
g(t)h(t-t)
1.0
(d)
g(t)h(5-t)
g(t)h(7-t)
0.5
g(t)h(2-t)
-5
-4
-3
-2
-1
t (ms)
Figure 6.36 Worked example 6.8. (a) Input pulse; (b) impulse response of matched
filter; (c) time-reversed and delayed versions of h(t); (d) Curves of g(t)h(t t), for t = 2, 5
and 7 s.
and-dump filter, which consists of an integrator followed by a sampleand-hold (S/H) circuit. The input pulse is integrated over its duration Ts,
and the output is sampled and the integrator reset at the end of each integration period. This technique yields an exact matched filter realisation
if the transmitted pulse is rectangular.
400 l
Decision instant
3
go(t)
2.5
1.5
0.5
t (ms)
1
10
Ts
Figure 6.37
WORKED EXAMPLE
6.9
Irrelevance of pulse shapes
Show that the signal-to-noise ratio (S/N)o at the output of a matched filter
depends only on the ratio between input pulse energy E and noise power
density, and not on the particular shape of the pulse.
It is clear from the last worked example, and more specifically Eq. (6.82),
that the signal at the output of a matched filter has a maximum value E at
the decision instant t = Ts, where E is the transmitted pulse energy.
Therefore, the instantaneous output signal power at the decision instant
is
Ps = [ g 0 (Ts )]2 = E2
(6.83)
(6.84)
401
where G(f) is the spectrum of the transmitted pulse, and we have used the
matched filter transfer function given by Eq. (6.77). The square of |C(f)|
gives the output noise power spectral density, and hence the average
output noise power as
Pn =
| C(f ) |2 df =
N0
2
| G(f ) |2 df
N
= 0E
2
(6.85)
| G(f ) |2 df
(6.86)
Thus,
(S/N )o =
=
Ps
E2
=
Pn ( N0/2)E
2E
N0
(6.87)
Eq. (6.87) is the desired result. Note that N0/2 is the power spectral
density of the input (white) noise. It is interesting that the shape or
waveform of the transmitted pulse g(t) does not feature in the achievable
signal-to-noise ratio. All that matters is the pulse energy, which may be
increased to improve (S/N)o by increasing the amplitude and/or duration
of the pulse. The latter option however would reduce the symbol rate. In
summary then, provided a matched filter is used at the receiver, all pulses of
the same energy are equally detected in the presence of white noise irrespective of the pulse shapes. We must therefore emphasise that pulse
shaping (studied in Section 6.6) is required for ISI minimisation and has
no bearing whatsoever on the impact of white noise.
WORKED EXAMPLE
6.10
Correlation receiver
402 l
(a)
Correlation receiver
Received
pulse
(b)
Matched
filter
Received
pulse
h(t)
= g(Ts t) g (t)
go(Ts)
g (t )
o
sample at
t = Ts
Ts
g (t )
go(Ts)
g(t)
(known pulse)
Coherent demodulator
Modulated
signal
LPF
(c)
Message
signal
Carrier
Figure 6.38 Worked example 6.9: Equivalence between (a) correlation receiver,
(b) matched filter and (c) coherent demodulator.
Ts
g 0 (Ts ) =
g (t ) g (t )dt
(6.88)
g 0 (t ) =
=
g (t )h(t t )dt
Ts
(6.89)
g (t ) g (Ts t + t )dt
where we have used our knowledge that the pulse has a finite duration
Ts, and also substituted the expression for the impulse response of the
matched filter h(t) = g(Ts t), which, when t is replaced by t t on both
sides of the expression, leads to the above result. Sampling g0(t) in Eq.
(6.89) at t = Ts gives
g 0 (Ts ) =
=
Ts
g (t ) g (Ts Ts + t )dt
g (t ) g (t )dt
0
Ts
0
which is identical to Eq. (6.88). Thus the correlation receiver and the
matched filter give identical results, provided of course that the matched
filter output is sampled at t = Ts.
403
We will see in the next chapter that a digital modulated system may
transmit orthogonal bandpass symbols to represent groups of 0s and 1s.
Therefore, a matched filter may be implemented in such a system as a
bank of correlators. Each correlator is fed with one of the symbols. When
the received symbol is applied to all the correlators, the largest output is
obtained at the correlator corresponding to the transmitted symbol.
Note that real symbols or waveforms g1(t), g2(t), g3(t), ..., gN(t), each of
duration Ts, are said to be orthogonal with respect to each other if
Ts
0,
g i (t ) g j (t )dt =
Ei ,
i j
i=j
(6.90)
Ts
g 2 (t )dt
0
Ts
[ g1(t ) + g 2 (t )]2 dt
0
Ts
g12 (t )dt +
Ts
(6.91)
Ts
= E1 + E2 + 0
SUMMARY
This now completes our study of digital baseband transmission. We have
acquired a thorough grounding in the principles of uniform and nonuniform quantisation and encoding of analogue signals for digital transmission, and a basic understanding of low bit rate speech coding. We also
studied how such digitised signals, and indeed all types of digital data, are
Index
1% bandwidth 25961
16-QAM 47881
average energy per symbol
BER 4812
detector 4801
modulator 4801
2B1Q line code 64
3-dB bandwidth 113
566 VCO 2723
479
SQNR of 3515
versus -law 355
alias frequency 301, 3045, 306
aliasing
distortion 3016
filter design 3079
steps to minimise 306
alignment channel 524
alternate mark inversion see AMI
AM see amplitude modulation
AMI 62, 373, 3756
amplification 13, 17, 54, 181
amplifier 140, 1724, 1845
see also operational amplifier
amplitude and phase keying 69,
4778, 489
amplitude modulation 149216
demodulators 174
features and applications of 185,
212
modulation factor 1546, 166, 168
modulation sensitivity 150
modulators 168
noise effects 61011
power 1657
receivers 181
spectrum 15865
transmitters 172
variants 186
waveforms 1514
see also ASK
amplitude response 114, 133, 136
amplitude shift keying see ASK
AMPS 498
analogue communication system
baseband 5360
compared to digital 513
modulated 67, 149293
noise effects 60714
analogue signal 4850, 83
634 l
Index
Index
635
13,
636 l
Index
Index
637
DSB 192
FM 274, 276
ISB 206
PM 279
SSB 203
VSB 210, 211
see also Receivers
de-multiplexers
analogue TDM 59
FDM 195, 502
optical (WDM) 51617
design considerations 1412, 3312,
3612, 4825, 61415
detectors
ASK 446, 454, 464
DPSK 458
FSK 448, 456, 474
PSK 447, 470
QAM 480
deviation ratio 223
see also modulation index
dibit 64
differential NRZ code 374
differential phase shift keying see DPSK
differential pulse code modulation see
DPCM
differentiator 22930, 279
digital baseband receiver 645,
389404
digital communication system
baseband 605, 323406
compared to analogue 513
design considerations 4825
modulated 679, 40794
noise effects 61415
digital modulation techniques,
comparison 4845
digital signal 50, 83
digital signal processing see DSP
digital subscriber line see DSL
digital to analogue conversion (DAC)
29, 51
digital translator 334, 348
digital transmission model 4213
digital versatile disc see DVD
diode demodulator
circuit and operation 174
conditions 177
diode detector see Diode demodulator
diode ring modulator 188
dipole antenna 574
Dirac delta function 11112, 313, 630
direct sequence spread spectrum
processing gain 550
receiver 554
transmitter 551
discrete baseband communication
system 5560
dispersion 36, 59, 137, 572
display devices
638 l
Index
CRT 22
flat panel 23
printers 24
distortionless transmission 134
distortions
aliasing 301, 306
aperture effect 315, 317
attenuation 136
delay 136
harmonic 140
intermodulation 140
non-linear 138
overload 363
phase 136, 203, 477
quantisation 53
double sideband suppressed carrier
amplitude modulation (DSB)
18696
applications 194, 212
demodulation 192
effects
of noise 6089
of phase error 192
modulation 186
phase synchronisation 193
spectrum 186
waveform 187
double sided spectrum 107
downlink 578
DPCM 61, 35665
DPSK
BER 460
detection 458
generation 458
introduction 457
M-ary 473
droop 63
DSB see double sideband suppressed
carrier amplitude modulation
DSL 31
DSP 13, 367
DTE see data terminal equipment
duplex
full 468
half 468
duty cycle 85
DVD 26
dynamic range 51, 327, 333
Earth
radiated noise 577
radius 42
station, for satellite communications
43, 606
EBCDIC 11
effective isotropically radiated power
(EIRP) 5734
efficiency
antenna 617
bandwidth 388, 460
code 64
data transmission 521, 622, 623
electroluminescent display see plasmapanel display
electromagnetic spectrum 39
electromagnetic waves 4, 13, 28,
3840, 767, 570
encryption 27, 51
energy
of bit 388, 432, 436, 614
of pulse 400
Rayleighs theorem 401
of signal 398
of sinusoidal symbol 410
envelope detector 4545
see also diode demodulator
envelope distortion 156, 198
envelope form of narrowband noise
5934
equaliser 115, 137
equivalent noise temperature
of overall receiver station 6046
of single system 5967
error burst 546
error correction 47, 52, 489
error function see complementary error
function
Euclidean space 413
exclusive-OR gate 5503.
see also XNOR gate
extended binary coded decimal
interchange code see EBCDIC
extraterrestrial noise 576
eye diagram 3945
facsimile (fax) 823
Farnsworth, Philo 13
fast fading 137
FAW see frame alignment word
FDM signals
group 5058
hypergroup 50911
jumbogroup 51314
mastergroup 51214
pilot tones 514
supergroup 5058
supermastergroup 51214
FDM see frequency division
multiplexing
FEC see forward error correction
fibre see optical fibre
field effect transistor (FET) see
transistor
figure of merit 607
filters
anti-alias 61, 3079
defined (BPF, BSF, HPF, LPF)
11516
ideal (brickwall) 116, 300, 381
matched 390
Index
639
demodulators 273
merits and demerits 288
modulators 264
narrowband 242
noise performance 2837, 61114
phase effects 230, 246, 284
power 256
wideband 252
waveforms 234
see also FSK
frequency multiplication 268
frequency response
of human ear 21
of system 113, 116, 131
of tuned circuit 282
frequency reuse 499
frequency sensitivity 221
frequency shift keying see FSK
frequency swing 222
frequency translation of 99, 183, 268
frequency variations in PM 227
FSK
binary
BER 432
coherent detection 448
frequency spacing 442
generation 441
introduction 689, 40811
non-coherent detection 456
spectrum 442
M-ary
BER 475
generation and detection 474
noise-bandwidth trade-off 476
see also bandwidth
full cosine roll-off characteristic 385
full duplex see duplex
functions 11112, 630
fundamental frequency 84
future of FDM 51718
gain response see amplitude response
galactic noise 576
gaseous absorption 575
Gaussian distribution 425, 582
Gaussian noise 5824
see also AWGN
generic flow control 542, 545
geometric representation 411
geostationary 43
Gibbs phenomenon 102
glottis 77
gold sequence 560
granular noise 363, 364
Gray code converter 4623, 47980
Gray coding 421, 432, 4634
ground wave 42
group delay 136
group signal see FDM signals
640 l
Index
14
Index
logarithm 123
loudspeaker 20
low bit rate speech coding 36572
classes
hybrid coders 3712
vocoders 36971
waveform coders 3689
computational complexity 367
processing delay 367
quality 365
robustness to transmission errors
367
table of standards
trade-offs 366
transparency 366
low earth orbit 43
low-level AM transmitter 172
lowpass filter 116
lowpass signal see baseband signal.
LPC see linear predictive vocoder
LSB see sideband
LSF see side frequency
LTI see linear time-invariant system
luminance signal 81, 207
magnetic
disks 25
readers 19
tape 25
MAN see metropolitan area network
Manchester code 373, 375, 378, 379
Marconi, Guglielmo 12
mark (binary 1) 89, 374
M-ary modulation see multilevel
modulation
masking effect 3, 21
mastergroup signal see FDM signals
mastergroup translating equipment see
MTE
matched filter
derivation 3902
worked examples 395404
MATLAB code 2378
maximum length sequence 553
maximum likelihood rule 424
maximum power transfer 595
Maxwell, James 13
mean opinion score see MOS
mean square quantisation error 327,
360
mean square value 119
medium earth orbit (MEO) 43
MELP see mixed excitation linear
prediction vocoder
mesosphere 41
metropolitan area network 35, 37
microphones
carbon 18
condenser 18
dynamic 17
641
electret 18
electromagnetic 17
piezoelectric 17
variable-resistance 18
midrise quantisation 325
midstep quantisation 325
minimum shift keying see MSK
mixed excitation linear prediction
vocoder (MELP) 371
mobile communication 43
modem 32, 489, 515, 518
modulation depth 154
modulation factor 1546, 166, 168
modulation index 154, 223
modulation sensitivity 150
modulation
defined 65
roles 657
types 679
modulators
AM 168, 170, 171
ASK 438, 463
DPSK 458
DSB 186
FM 265, 267, 268, 270, 272
FSK 441, 474
ISB 205
narrowband PM 265
PSK 440, 470
QAM 480
SSB 200, 201
VSB 208
see also transmitter
monomode step index fibre 36
Morse code 7
Morse, Samuel 7
MOS 3656
MSK 444
MSQE see Mean square quantisation
error
MTE 512, 514
-law PCM
characteristic curve 33841
companding advantage 3545
companding gain 341, 3545
practical implementation 3447
speech quality 3656
SQNR 3515
versus A-law 355
muldex
add/drop 536, 539
E1 hierarchy 525
J1 hierarchy 529
NA PDH hierarchy 528
primary 522
SDH 532
multiframe 520, 523
multiframe alignment word 5234
multilevel modulation
bandwidth efficiency 460
642 l
Index
Index
643
644 l
Index
Index
of
of
of
of
645
ISB 205
PSK 440
raised cosine filter 383
sampled signals
bandpass 299
lowpass 298, 312, 316
of speech signal 79
of SSB 197
types of 102, 107
of VSB 208
speech signal
characteristics of 78
coding of 333, 365
production of 77
speech synthesis model 370
SPL see sound pressure level
splicing 35, 573
spread spectrum modulation 28, 547
spreading loss 574
Sputnik I 14
SQNR see signal to quantisation noise
ratio
square law demodulator 215
square law modulator 171
square law noise-bandwidth trade-off
285, 331, 613
square QAM constellation 478
SSB transmitter 201
SSB see single sideband suppressed
carrier amplitude modulation
star QAM constellation 478
STE 509, 512, 514
step size see quantiser step size
stereo 194
storage devices 25
STP see screened twisted pair
stratopause 41
stratosphere 41
Strowger, Almond 12
superframe 5289
supergroup signal see FDM signals
supergroup translating equipment see
STE
superheterodyne (superhet) receiver
182, 288
supermastergroup signal see FDM
signals
superposition principle 131
switching delay 540
switching impulse train 296, 298
switching modulator 170
symbol error rate 425, 432, 475
symbol rate 69, 382, 387
symbol slip 393
synchronisation
of bit 393, 519
of carrier phase 193
of frame 520
synchronous demodulation see
coherent demodulation
646 l
Index
Index
647
129