ES335 Communication Systems: Part One

Contents

1. DIGITIZATION __________________________________________________________1
Introduction__________________________________________________________________ 1
Overview ____________________________________________________________________ 1
Review of the Sampling Theorem ________________________________________________ 2
Signal Quantization____________________________________________________________ 5
Pulse-Code Modulation ________________________________________________________ 7
Noise in PCM Systems ________________________________________________________ 11
Summary of PCM features, strengths and weaknesses ______________________________ 13
Differential Pulse Code Modulation (DPCM) _____________________________________ 13
Coding Speech at Low Bit Rates ________________________________________________ 14
2. BASEBAND PULSE TRANSMISSION ______________________________________16
Introduction_________________________________________________________________ 16
The Matched Filter ___________________________________________________________ 16
Effects of noise on PCM error probability ________________________________________ 18
Intersymbol interference ______________________________________________________ 21
Nyquist’s Criterion for Distortionless Transmission ________________________________ 23
Baseband M-ary Pulse Transmission ____________________________________________ 26
Equalization_________________________________________________________________ 27
Eye Diagram ________________________________________________________________ 30
Synchronization______________________________________________________________ 30

1. DIGITIZATION

Introduction
The remainder of this module is concerned with digital telecommunications. Thus we need to
look at ways of digitizing analogue information carrying signals. The first process is sampling
followed by quantization. By digitizing signals, we will find that the time domain plays a very
important part of the discussion. Remember that in analogue communications, the frequency
domain was important.

Overview
• Sampling

ES335 COMMUNICATION SYSTEMS 1 Dr S. Lawson

• Pulse Modulation
• Time-Division Multiplexing(TDM)
• Quantization
• Pulse-Code Modulation(PCM)
• Differential PCM

Review of the Sampling Theorem

An analogue signal which is bandlimited to W Hz must be sampled at >2W in order that it
may be successfully reconstructed using a low-pass filter. The Nyquist Rate is normally held
to be the value 2W Hz.

g(t)

Ts

Figure 19: The sampling process

The ideal sampling process can be represented mathematically as

∞
gδ ( t ) = g ( nTs )δ(t − nTs )
n = −∞

or in frequency terms as
∞
Gδ ( f ) = f s G( f − mf s )
m= −∞

More practically, we sample by finite pulses, in which case we find that a sin x / x envelope
will be present.
We can view sampling in a symbolic way as in the next illustration. Here the signal is being
sampled at the Nyquist rate, i.e f s = 2W . To recover the original signal, an ideal lowpass
filter would be required. In practice, f s > 2W to allow for the filter roll-off.

ES335 COMMUNICATION SYSTEMS 2 Dr S. Lawson

 G(f)

G(0)

f
-W 0 W

Gδ (f)

2WG(0)

f
-2fs -fs -W 0 W fs 2f s

Figure 20: Sampling at the Nyquist rate

Using pulses, various types of modulation can be defined, e.g. Pulse Amplitude
Modulation(PAM), Pulse Position Modulation(PPM) and Pulse Duration Modulation(PDM).
g(kTs)

t
T
T
s

Figure 21: PAM signal

ES335 COMMUNICATION SYSTEMS 3 Dr S. Lawson

 g(kTs)

t
T
s

Figure 22: PDM signal

g(kTs)

t
T
s

Figure 23: PPM Signal

We are not taking the discussion on these forms of modulation any further. However, the
discussion on PAM leads to an important property, that of Time-Division
Multiplexing(TDM). This can be most explained by a diagram (see Fig.24). The spaces
between the pulsed amplitudes for one signal can be used for pulses from other multiplexed
signals. In practice, T << Ts .

ES335 COMMUNICATION SYSTEMS 4 Dr S. Lawson

 Anti-alias Reconstruction

LPF LPF

Commutator Decommutator
LPF LPF
Pulse Pulse
Mod Channel Demod

pulses pulses
LPF LPF

Figure 24: Block diagram of a TDM system

If all inputs have same message bandwidth W, the commutator should rotate at a rate
f s > 2W so that samples from any one input are spaced apart by Ts = 1 / f s < 1 / 2W . The time
interval Ts containing one sample from each input is called a frame. If there are M such inputs
then, then the pulse spacing in a frame is Ts / M = 1 / Mf s .

Thus the total number of pulse per second is r = Mf s > 2 MW , r represents the pulse or
signalling rate of the TDM system.

Signal Quantization
Assume we have a finite number of amplitude levels. So after sampling a signal at time kT,
we will need to determine the level from the fixed set of such levels nearest to x( kT ) . We
will call this quantity xq ( kT ) . The difference x( kT ) − xq ( kT ) is called the Quantization
Error. This error is of course random and we can determine some of its statistical parameters
quite easily.

-1
0 1 2 3 4 5
time(secs
0.1

-0.1
0 1 2 3 4 5
time(secs
Figure 25: Quantization Process for a Sine Wave

ES335 COMMUNICATION SYSTEMS 5 Dr S. Lawson

We can calculate the statistics of the quantization error, if we know the precise format.If we
assume a uniform quantizer of the midrise type (see Fig. 26) then the quantization stepsize
∆ = 2mmax / L where L is the total number of levels and the input signal has a range
[ −mmax , mmax ] . The quantization error will be such that q ≤ ∆ / 2 .
If we assume that Q is uniformly distributed and that its probability density function is given
by the following,

1 ∆ ∆
− <q≤
fQ ( q ) = ∆ 2 2
0 otherwise

∆2
we can show easily that the mean is zero and that the variance σ = . 2
Q
12
Output Level Output Level

4 4

2 2

-4 -2 2 4 Input -4 -2 2 4 Input
Level Level

-2 -2

-4 -4

(a) (b)

Figure 26: (a) Midtread quantization (b) Midrise Quantization

We can go further by finding the signal-to-noise ratio of a uniform quantizer. If the number of
bits per sample is R then the number of levels, L = 2 R and hence
2mmax
∆= and
2R
1 2 −2R
σQ2 = mmax 2
3
If P is the average power of the message signal m( t ) then the output signal-to-noise ratio is
P 3P 2 R
as follows: SNR = = 2
σQ
2 2
mmax

ES335 COMMUNICATION SYSTEMS 6 Dr S. Lawson

This result shows that output SNR increases exponentially with an increase in the number of
bits per sample. However the bandwidth requirement will also increase.

Pulse-Code Modulation
The simplest type of digital pulse modulation consists of anti-aliasing filtering, sampling,
quantizing and finally encoding. This process is known as Pulse Code Modulation(PCM) and
can be viewed as analogue to digital conversion. Figure 27 shows a PCM transmitter,
transmission link and receiver.

TRANSMITTER
Source
Sampler Quantizer Encoder
Analogue fs NQ ν
Signal
Anti-alias
TRANSMISSION PATH
Regenerative Regenerative
Repeater Repeater

RECEIVER
Regen Destination
Decoder
Circuit
Analogue Output
Reconstruction
Filter
Figure 27: PCM System

1. Sampling
Train of narrow rectangular pulses at a rate f s > 2W to allow for non-ideal reconstruction
filter.
2. Quantization
Uniform quantization already dealt with
Non-uniform quantization more useful since for voice signals, ratio of loud to soft can be as
much as 1000. Input signal is compressed then uniformly quantized. Received signal is then
expanded. Overall process is called Companding. Two main types: µ-law and A-law.
These laws are logarithmic. They tend to give extra weight to small level signals. In practice,
piecewise linear approximation to these curves is used.

ES335 COMMUNICATION SYSTEMS 7 Dr S. Lawson

3. Encoding
Each quantized sample is encoded into ν pulses. For binary coding and 256 possible
quantized levels, ν would be 8. These 8 bits are called a codeword. We can use ternary or
quaternary coding but binary offers the best performance because of its high noise immunity.
There are many possible ways of electrically representing the bit stream. Figure 29 illustrates
a few.

mu-law A-law
1 1

0.9 0.9

0.8 0.8

0.7 0.7

| |
v|, 0.6 v|, 0.6
t t
u
pt u
pt
u u
o 0.5 o 0.5
d d
e e
sil sil
a 0.4 a 0.4
mr mr
o o
N N
0.3 0.3

0.2 ------- u=0 0.2 ------- A=1

oooo u=5 oooo A=2
++++ u=100 ++++ A=100
0.1 0.1

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1
Normalised input,|m| Normalised input,|m|

Figure 28: Compression Laws

4. Regeneration
Reconstruction of PCM signal by merans of a chain of regenerative repeaters loacted at
sufficiently close spacing along the transmission path. The basic structure of such a
regenerator is shown in Figure 30.
5. Decoding
After regeneration, pulses are grouped into codewords and decoded into a quantized PAM
signal.
6. Filtering
This is the reconstruction filter whose job it is to produce an analogue signal with all aliases
removed. Its bandwidth will be W.
7. Multiplexing
If we wish to increase the number of independent messages transmitted on the same PCM
link, then pulse duration must be reduced. Clearly a limit will be reduced for any given time-
division group.

ES335 COMMUNICATION SYSTEMS 8 Dr S. Lawson

8. Synchronization
Clock synchronization. Frame in which synchronizing pulse is inserted into a group of
codewords representing one sample from each multiplexed signal.
9. PCM Bandwidth
For unique coding, number of quantization levels, N Q ≤ µ v . For a transmission rate of r
bauds, the bandwidth required is as follows,
r vf s
BTBB ≥ = ≥ vW
2 2
For an 8-bit PCM system with uniform quantization and telephone quality speech,
N Q = 256 ,µ = 2 ,v = 8,W = 4000 Hz

so that BTBB ≥ 32kHz .

ES335 COMMUNICATION SYSTEMS 9 Dr S. Lawson

 0 1 1 0 1 0 0 1

On-off s ignalling

NRZ

RTZ

Biploar

Manchester Code
Ref Bit
Time

Differential
Encoding
Figure 29: Electrical representations of binary data

ES335 COMMUNICATION SYSTEMS 10 Dr S. Lawson

 Distorted Amplitude Decision Regenerated
PCM Equalizer Making PCM
Device

Timing Circuit

Figure 30: Block diagram of regenerative repeater

Noise in PCM Systems

For binary PCM, we can derive an expression for signal-to-noise ratio at the destination.

Consider a binary codeword of the form bv −1bv − 2 b1b0 . An error in the mth bit shifts the
2 m
decoded level by the amount ε m = ± 2 where 2 / q is the step height.
q

The mean square error is found from the following:

2
1 v−1 2 m 4 v −1
ε =
2
m 2 = 4m
v m=0 q vq 2 m=0

4 4 −1 4 q −1 4
v 2
= = ≈
vq 2 3 3v q 2 3v
The decoding noise power is therefore
4
σ d2 = vPe ε m2 ≈ P e
3
since erroneous words occur with probability vPe .

The total destination noise power consists of decoding noise and quantization noise
σ 2q = 1 / 3q 2 . Therefore

1 + 4q 2 Pe
ND = σ + σ =
2
q
2
d
3q 2

S 3q 2
Thus = x2
N D 1 + 4q Pe
2

Quantization noise dominates when Pe is small but decoding noise dominates and reduces
destination SNR when Pe is large compared with 1 / 4q 2 .

ES335 COMMUNICATION SYSTEMS 11 Dr S. Lawson

As the input SNR increases, the destination SNR increases rapidly but only up to a point
called the error threshold after which there is little improvement.

50
q=128

40
q=32
30
(S/N)D dB
20

10

0
10 20 30 40 50
(S/N)R dB
Figure 31: Noise performance of PCM

The PCM Error Threshold is defined as the point at which decoding noise reduces by 1dB.
This definition precludes analytical investigation. More usefully, consider the situation in
which decoding errors have negligible effect if Pe ≤ 10 −5 . We can find that

≈ 6(M 2 − 1) for M-ary signalling. For any value less than this, the PCM output will be
S
N Rth
badly effected by decoding noise.

A useful parameter in analogue transmission is

SR B S
γ= = T
ηW W N R

r
The PCM transmission bandwidth is BT ≥ ≥ vW so that
2

γ th =
BT S
W N
≈6
BT
W
(M 2 − 1) ≥ 6v(M 2 − 1)
Rth

From this expression we can determine the minimum value of γ for PCM operation above
threshold given values for v and M. It also allows comparison with other transmission
methods.

ES335 COMMUNICATION SYSTEMS 12 Dr S. Lawson

 60
PCM b=8
50
PCM b=6
40
WBFM,
b=6
30
PCM b=4
(S/N)
D
dB 20 PPM b=6

10
Baseband
0
10 20 30 40 50
γ dB
Figure 32: Comparison between PCM and Analogue Modulation ( b = BT / W )

Summary of PCM features, strengths and weaknesses

• Robustness to channel noise and interference
• Efficient regeneration of the coded signal
• Channel bandwidth and signal-to-noise ratio are inversely related
• Uniform format for signals so easy integration with other forms of digital data
• Time-division multiplexing is relatively easy
• Secure communications possible with encryption of transmit data

All of these characteristics of PCM can be attributed to the use of coded pulses for the digital
representation of analogue signals. In this sense it is very different from all other forms of
analogue modulation.

Differential Pulse Code Modulation (DPCM)

When a speech signal is sampled at just higher than the Nyquist rate, it is found that there is
high correlation between successive samples. This suggests that the signal changes slowly and
therefore there is redundancy which can be exploited in a coding scheme. If we know a
sufficient part of a highly correlated signal then we will be able to predict the rest.

ES335 COMMUNICATION SYSTEMS 13 Dr S. Lawson

Suppose a baseband message signal m( t ) is sampled at a rate f s = 1 / Ts producing a
sequence {m( 0 ), m( Ts ), m( 2Ts ), , m( kTs ), }. We will define a differential quantization
scheme as shown in Figure 33 in which the input to the quantizer is as follows:
e( kTs ) = m( kTs ) − m̂( kTs )
This represents the error between the actual sample and its predicted value.

m (kT ) e(kT ) eq (kT ) DPC M

Quantizer Encoder
Out

^
m (kT )

P rediction
F ilter T ransmitter
mq (kT )

Decoder

Prediction
Filter
R eceiver
Figure 33: Block diagram of DPCM system
The prediction filter is most easily realized as a tapped delay line filter or FIR filter. We can
show that mq ( kTs ) = m( kTs ) + q( kTs ) where q( kTs ) is the quantization error, i.e.
eq ( kTs ) = e( kTs ) + q( kTs ) . By careful design, it is possible to produce quantized signals
whose quantization error will be lower than that obtained using PCM.
DPCM performs similarly to PCM and the SNR at the destination is given by the following:
S
= G p 3q 2 x 2 where q is the number of quantized levels and G p is the prediction gain.
N D
Prediction gains of 5-10dB for speech and 12dB for video have been obtained.

Coding Speech at Low Bit Rates

It is possible to transmit speech at rates well below the Nyquist rate by exploiting the
statistical behaviour of speech waveforms. Two important things need to be borne in mind.
• Removal of redundancies in speech signal

ES335 COMMUNICATION SYSTEMS 14 Dr S. Lawson

• Assigning available bits to code the non-redundant parts of the speech in an efficient way
Standard PCM used a sampling rate of 8kHz and 8 bits per sample non-uniform quantization
so the bit rate is 64kb/s. It is possible to reduce this to 32, 16, 8 and 4kb/s.
Adaptive Differential PCM (ADPCM) for 32kb/s
Adaptive Subband Coding for 16kb/s
Applications:
• Digital Multiplexers in which computer outputs, digitised voice signals, digitised fax and
tv signals are combined into a single data stream.
• Lightwave transmission for long distance and high speed links.

ES335 COMMUNICATION SYSTEMS 15 Dr S. Lawson

 2. BASEBAND PULSE TRANSMISSION

Introduction
• Transmission of digital data over a baseband channel
• Broad spectrum
• Low-frequency content
• Lowpass Channel
• Dispersive channel
• Inter-symbol interference(ISI)
• Pulse shaping
• Channel noise
• Pulse detection and matched filtering

The Matched Filter

The problem: Detection of a pulse signal buried in additive white noise.
If the received pulse signal is
x( t ) = g ( t ) + w( t ) 0 ≤ t ≤ T
where T is an arbitrary observation interval, g ( t ) is the pulse and w( t ) is the additive noise.
The pulse signal at particular time instants will represent binary 0 or 1. We will assume that
the noise is white with zero mean and PSD N 0 2 . We will also assume that the receiver
knows the basic pulse shape.
Sample at t=T
g(t) x(t) y(t) y(T)
LTI h(t)

w(t)
Figure 34: Linear receiver

In the receiver of Figure 34, the output of the Linear Time-Invariant(LTI) Circuit may be
expressed as follows:
y( t ) = g 0 ( t ) + n( t )
where g 0 ( t ), n( t ) are the outputs corresponding to the pulse and noise inputs respectively
(superposition principle for LTI circuits).
Can we design a circuit which will obtain the best estimate (in some sense) of the pulse?

ES335 COMMUNICATION SYSTEMS 16 Dr S. Lawson

 2
g0 ( T )
Consider the measure η=
E [ n 2 ( t )]
This represents the peak pulse signal-to-noise ratio. Can we find the impulse response of a
LTI system that maximises η?
Using the Fourier transform, we can show that
∞ 2

g0 ( T ) = H ( f )G( f ) exp( j 2πfT )df

2

−∞

and, using the properties already known about white noise, that
∞
[
E n2 ( t ) = ] N0
2
2
H ( f ) df
−∞

With these expressions , it is possible to show that

hopt ( t ) = kg( T − t )
The impulse response of the optimum LTI circuit (filter) is a scalar times the time-reversed
and delayed version of the pulse signal being transmitted. In this way we can see that the filter
is matched to the pulse and is henceforth called a matched filter.
The matched filter transfer function is obtained by applying the Fourier transform to the
expression above, thus
H opt ( f ) = kG* ( f ) exp(− j 2πfT )
Consider the output of a matched filter,
G0 ( f ) = H opt ( f )G( f )
= kG* ( f )G( f ) exp(− j 2πfT )
= k G( f ) exp (− j 2πfT )
2

Applying the inverse Fourier transform,

∞
g 0 ( T ) = k G0 ( f ) exp( j 2πfT )df
−∞
∞
= k G( f ) df
2

−∞

= kE
where E is the signal energy.
Looking at the noise, we can find that, using an earlier result,
∞
[
E n2 ( t ) = ] k 2 N0
2
G( f ) df
2

−∞

= k N0 E / 2
2

Therefore the maximum value of the peak pulse SNR is as follows,

ES335 COMMUNICATION SYSTEMS 17 Dr S. Lawson

 ηmax =
(kE )2 =
2E
2
( k N0E / 2 ) N0
This is an important result. It shows that the peak pulse SNR depends only on the ratio of the
signal energy and the PSD of white noise at the the filter input. The ratio is dimensionless as
E is measured in joules and N 0 in watts per hertz. The ratio E / N 0 is known as the signal
energy-to-noise spectral density ratio.
In the case of a rectangular pulse, the matched filter impulse response is, of course, identical
to the signal. The matched filter output which is shown in Figure 35(b) has a maximum at
t = T equal to kA2T . An integrate-and-dump circuit can be used to detect rectangular pulses.
The output of such a circuit is shown in Figure 35(c).

Effects of noise on PCM error probability

Assumptions: matched filter detector, NRZ signalling, AWGN with zero mean and PSD
N0 / 2 .
In the signalling interval 0 ≤ t ≤ Tb , received signal is

+ A + w( t ), symbol 1 transmitted
x( t ) =
− A + w( t ), symbol 0 transmitted
where Tb is the bit duration and A is the transmitted pulse amplitude. The receiver needs to
make a decision as to whether the received pulse is 1 or 0. A suitable receiver is shown in
Figure 36.
g(t)

A
(a)
t
T
g0(t)
2
kA T
(b)
t
T

AT
(c)
t
T
Figure 35: (a) Rectangular pulse signal (b) Output of matched filter (c) Integrator output.

ES335 COMMUNICATION SYSTEMS 18 Dr S. Lawson

 Sampler

s(t) y Decision 1 if y>L

Matched Filter
Device
0 if y<L

w(t) Threshold L
Figure 36:Block diagram of PCM receiver using a matched filter
Two types of error to consider:
Symbol 1 received when 0 was transmitted. Error of the first kind.
Symbol 0 received when 1 was transmitted. Error of the second kind.
If symbol 0 is sent, then the matched filter output, sampled at t = Tb , is
Tb

y = x( t )dt
0
T
1 b
= −A + w( t )dt
Tb 0
The variable y is a sample from a Gaussian random variable Y with mean − A . The variance
of the random variable Y is
[
σY2 = E (Y + A)
2
]
Tb Tb
1
= 2E w( t )w( u )dtdu
Tb 0 0
Tb Tb
1
= 2 Rw ( t ,u )dtdu
Tb 0 0

Recall that the noise is white so that

Tb Tb

δ(t − u )dtdu
1 N0
σ = 2
2
Y
Tb 0 0
2
N0
=
2Tb
The probability density function of the random variable Y (given that 0 was sent) is then

fY ( y | 0 ) =
1
exp −
( y + A) 2

πN 0 / Tb N 0 / Tb

We can sketch this function (Figure 37). From the graph, we can see that
Pe 0 = P( y > L | 0 is sent )
∞
= fY ( y | 0 )dy
L

ES335 COMMUNICATION SYSTEMS 19 Dr S. Lawson

 f(y|0)

Pe0

y
-A L

f(y|1)

Pe1

y
L A

Figure 37: Sketches showing error probabilities when (i) 0 and (ii) 1 are transmitted
respectively.
Can we find a value for L? Firstly let the a-priori probabilities be respectively p0 and p1 , that
is the probabilities of transmitting a 0 or 1. If we assume symbols are equally likely at the
source, then p0 = p1 = 12 . Since the received pulse amplitude is either + A or − A , the logical
choice for L is zero.
Thus

Pe 0 =
1
∞
exp −
( y + A) dy 2

πN 0 / Tb 0 N 0 / Tb

y+ A
Let z = then
N 0 / Tb
∞
1
Pe 0 = exp( − z 2 )dz
π Eb / N 0

where Eb is transmitted signal energy per bit defined as follows:

Eb = A2Tb
If we now define the complementary error function, erfc, as follows:
∞
erfc(u ) =
2
exp − z 2 dz ( )
πu
then

ES335 COMMUNICATION SYSTEMS 20 Dr S. Lawson

 1 Eb
Pe 0 = erfc
2 N0

We can show that Pe1 = Pe 0 .

We are now in a position to find the probability of a symbol error,
Pe = p0 Pe 0 + p1 Pe1
= Pe 0
1 Eb
= erfc
2 N0

This is an important result. It has been shown that the probability of a symbol error in a binary
encoded PCM receiver depends only on the ratio of the transmitted signal energy per bit and
the noise spectral density. Figure 38 shows the variation of probability with this ratio.

0
10
-2
10
-4
10
-6
10
Pe 10-8
-10
10
-12
10
-14
10
-16
10
0 5 10 15
Eb/No,dB
Figure 38: Probability of error in a PCM receiver using matched filter detection.

Intersymbol interference
To consider the effects of the channel on pulse transmission, we need to have a model of a
suitable baseband binary data transmission system. Such a system is shown in Figure 39.

ES335 COMMUNICATION SYSTEMS 21 Dr S. Lawson

 bk ak s(t)
PAM Transmit Channel
Filter

Clock

x(t) y(t) y(t i) Binary

Receive Decision
Output
Filter Sample Device
t i= iT b
AWGN w(t) Threshold, L
Figure 39: Block diagram of baseband binary data transmission system
The Pulse Amplitude Modulator takes the input binary data and converts them into a sequence
of short pulses whose amplitude is as follows:
+ 1 if symbol bk is 1
ak =
− 1 if symbol bk is 0

This sequence is then passed through a filter with impulse response g ( t ) producing an
output,
s( t ) = ak g (t − kTb )
k

which is then modified by the channel whose impulse response is h( t ) . At the receiver, the
signal plus noise is passed through a filter with impulse response c( t ) . The output signal is
sampled at the data rate and the amplitude at each sampling time is thresholded to determine
whether the received symbol is 1 or 0.
The receive filter output is as follows,
y( t ) = µ ak p( t − kTb ) + n( t )
k

where µ is a scaling factor and p( t ) is the pulse which has yet to be defined. The expression
above should also show a delay through the channel which, for the sake of simplicity, has
been set to zero.
The filter output is then sampled at ti = iTb producing
∞
y( ti ) = µ ak p[(i − k )Tb ] + n( ti )
k = −∞
∞
= µai + µ ak p[(i − k )Tb ] + n( ti )
k = −∞
k ≠i

assuming the pulse p( t ) is defined so that p( 0 ) = 1 .

ES335 COMMUNICATION SYSTEMS 22 Dr S. Lawson

The result above is important as it shows that the received signal consists of the desired
symbol ai plus a term which introduces interference from other symbols and hence is called
intersymbol interference or ISI. The third term is the noise sampled at time t = ti .
ISI and noise are unavoidable problems in digital transmission systems. The question is: can
we design the system to minimize their effects?
When the SNR is high, as in telephone systems, the performance is limited by ISI. In the next
section, we will look at the problem of ISI and how we can mitigate its effects.

Nyquist’s Criterion for Distortionless Transmission

The expression we found for the ith receiver output suggests that zero ISI is possible if we can
find a pulse shape such that
1 i=k
p( iTb − kTb ) =
0 i≠k
We now state Nyquist’s criterion for distortionless baseband transmission in the absence of
noise:
The frequency function P( f ) eliminates ISI for samples at intervals of Tb
provided that the following equation is satisfied
P( f − nRb ) = Tb
∞
n = −∞

where µP( f ) = G( f )H ( f )C( f ) refers to the whole system. Rb = 1 / Tb is bit

rate in bits per second.
The rectangular function defined below satisfies Nyquist’s criterion:
1
f <W
P( f ) = 2W
0 f >W
1 f
= rect
2W 2W
Rb 1
where the system bandwidth W = = .
2 2Tb
The sinc function satisfies Nyquist’s criterion since its form p( t ) = sin(2πWt ) / 2πWt )
ensures that at sampling times other than zero, the pulse has a null.
The problem with the Nyquist pulse is, of course, that it is not finite-duration although the ISI
is zero.
If we define the frequency function P( f ) in a different way, it is possible to overcome the
problems associated with the Nyquist pulse.
Recall the following,
P( f − nRb ) = Tb
∞
n = −∞

For the Nyquist pulse, we used only one term on the LHS. If, instead, we use three terms, i.e.

ES335 COMMUNICATION SYSTEMS 23 Dr S. Lawson

 1
P( f ) + P( f − 2W ) + P( f + 2W ) = Tb = − W ≤ f ≤ W then pulses with realisable
2W
properties can be found.

0.8

0.6

0.4
amplitude

0.2

-0.2

-0.4

-0.6

-0.8

-1
-5 0 5 10 15
time(secs) x 10
-4

Figure 40: Sinc pulses corresponding to the binary sequence 1011010

In particular, we can define what is known as the Raised Cosine Spectrum.
1
0 ≤ f ≤ f1
2W
1 π( f − W )
P( f ) = 1 − sin f1 ≤ f < 2W − f1
4W 2W − 2 f1
0 f ≥ 2W − f1

f1
where the rolloff factor, α = 1− .
W
The transmission bandwidth BT = 2W − f1 = W (1 + α ) .

ES335 COMMUNICATION SYSTEMS 24 Dr S. Lawson

 1
Legend
--- alpha=0
- - alpha=0.5
0.8
-.- alpha=1

0.6
P(f)

0.4

0.2

0
-2 -1.5 -1 -0.5 0 0.5 1 1.5 2
f/W

Figure 41: Frequency characteristics of the raised cosine pulse for different rolloff factors

The time domain response can be derived by applying the inverse Fourier transform to P( f )
to give
cos(2παWt )
p( t ) = sinc(2Wt )
1 − 16α 2W 2t 2
This expression is seen as the product of two factors. The first is the ideal Nyquist channel
response which gives zeros at the desired sampling times of t = iT and the second term
reduces the sidelobes considerably.

ES335 COMMUNICATION SYSTEMS 25 Dr S. Lawson

 1

0.8 Legend
--- alpha=0
- - alpha=0.5
0.6 -.- alpha=1

0.4
p(t)

0.2

-0.2

-0.4
-3 -2 -1 0 1 2 3
t/Tb

Figure 42: Time response characteristics of the raised cosine pulse for different rolloff factors

Example:
T1 carrier system used in US, Canada and Japan
24 channel 8 bit PCM
Companding: 15 linear segments, µ-law, µ=255
Voice signal bandwidth: 300-3100Hz
Sampling frequency: 8kHz
Frame: 24 8-bit words + 1 synch. bit = 193 bits
Bit duration of 0.647µs and transmission rate of 1.544 Mbits/s
1
BT = W = = 772kHz with α = 0
2Tb
More realistic value using α = 1,
1
BT = W ( 1 + α ) = 2W = = 1.544MHz
Tb
For Frequency Division Multiplexing, the transmission bandwidth is 24×4=96kHz.

Baseband M-ary Pulse Transmission

We can extend the earlier discussion on error probabilities to the case where we have M
levels. It can be shown that in the case where the amplitude levels are

ES335 COMMUNICATION SYSTEMS 26 Dr S. Lawson

 {ak } = ± A , ± 3 A , ,±
(M − 1)A
2 2 2
with decision thresholds at
M −2
0 , ± A, ± 2 A , ,± A
2
that
1 A
Pe = 2 1 − Q
M 2σ
where
∞
Q( k ) =
1
(
exp − λ2 / 2 dλ )
2π k
and
erfc(k ) = 2Q 2 k( )
We can show that the mean square amplitude is
M 2 − 1 A2
A2 = .
3 4
So
SR S M 2 − 1 A2
= R =
N R N0 B 3 4σ 2
and thus

1 SR 3
Pe = 2 1 − Q
M M −1 NR
2

SR
If we let ρ = and r = 2 B then
N0r

1 6ρ
Pe = 2 1 − Q
M M 2 −1

( )
which simplifies to Pe = Q 2ρ in the binary case.

Equalization
Irrespective of pulse shape, some residual ISI remains in the output as a result of imperfect
filter design and incomplete knowledge of the channel. An Equalizer is used between the
receiving filter and the decision device. The most popular configuration for an equalizer is a
FIR filter structure as shown in Figure 46.

ES335 COMMUNICATION SYSTEMS 27 Dr S. Lawson

x(t)
T T T T

C C -N+1 C -N+2 C -N+3 C N-1 CN

-N
x eq(t)

Figure 46: Block diagram of FIR equalizer

The output of the equalizer is as follows:
N
xeq ( t ) = Ck x(t − kT − NT )
k =− N

The distorted input x( t ) has a peak at t = 0 and ISI on each side.

Ideally,
1 k =0
xeq (t k ) =
0 k≠0
In practice, we will choose the tap gains {Ci } such that

1 k =0
xeq (t k ) =
0 k = ±1,± 2 , ,± N
thereby forcing N zero values on each side of the peak of xeq ( t ) . If we apply this constraint, a
system of linear equations is created which can be solved to find {Ci } .
For N = 2 , the linear system of equations is,
x0 x−1 x− 2 x− 3 x− 4 C− 2 0
x1 x0 x−1 x− 2 x− 3 C−1 0
x2 x1 x0 x−1 x− 2 C 0 = 1
x3 x2 x1 x0 x−1 C1 0
x4 x3 x2 x1 x0 C2 0
This defines the Zero-Forcing Equalizer. The solution of this linear equation set gives the
values of the 2 N + 1 = 5 taps. The technique represents an optimum strategy in the sense that
it minimizes the peak ISI. Various other types of equalizers are used in telecommunication
systems. It is possible to train the equalizer by first transmitting a pseudo-random sequence
which is used to compute an estimate of the channel characteristic and hence the FIR filter
coefficients. Once set, the system is then used to receive data. For time-varying channel
characteristics, an adaptive equalizer may be used whose coefficients (taps) are varied in real-
time according to some algorithm such as the Least Mean Square(LMS).
Example: Design a three tap zero-forcing equalizer for the pulse shown in Figure 47.

ES335 COMMUNICATION SYSTEMS 28 Dr S. Lawson

 1.0

0.6

0.2

-3T -2T -T T 2T 3T
-0.2

Figure 47: Received pulse shape

1.0 0.1 0.0 C−1 0
− 0.2 1.0 0.1 C0 = 1
0.1 − 0.2 1.0 C1 0

Solving gives C−1 = −0.096,C0 = 0.96,C1 = 0.2 . Using these values in

1
xeq ( t ) = Ck x(t − kT − T )
k =−1

gives a pulse shape as shown in Figure 48.

1.0

0.6

0.2

-3T -2T -T T 2T 3T 4T
-0.2

Figure 48: Equalized pulse shape

ES335 COMMUNICATION SYSTEMS 29 Dr S. Lawson

Eye Diagram
To study the ISI in a data transmission system, it is useful to construct what is known as the
Eye Diagram or Eye Pattern. The diagram can be obtained using a long persistance
oscilloscope with appropriate synchronization and sweep-time setting. Figure 49 shows a
general eye pattern with various features labelled.
Properties
(i) Width of eye opening defines the time interval over which the received wave can
be sampled without ISI error.
(ii) Sensitivity of system to timing errors is determined by the rate at which the eye
closes as sampling time is varied.
(iii) Height of eye opening, at a specified sampling time defines the noise margin.
ISI

Timing sensitivity
Noise
Margin

Zero-crossing
Jitter

Optimum Sampling
Time
Figure 49: Generalized Binary Eye Diagram

Synchronization
The receiver and transmitter have separate clocks so that transmission delay will cause
problems with synchronization.
We could use a separate sync signal in addition to information-bearing signal. More efficient
to extract timing from received signal.
Frame synchronization
Identify start of message or subdivision within message sequence.

ES335 COMMUNICATION SYSTEMS 30 Dr S. Lawson

 y(t) Output
Regenerator
Message

Bit Frame Frame

Sync. Clk Sync.
Indicator
Prefix Start
Message Bits

N-bit Sync Word

Figure 50: Bit and Frame Synchronization
Bit synchronization
Zero crossing detector generates a rectangular pulse with half bit duration Tb / 2 starting at
each zero crossing in y( t ) .
Pulsed waveform z( t ) then multiplies square-wave coming from voltage controlled clock
(VCC). The control voltage to the VCC, v( t ) , is the result of integrating and lowpass filtering
z( t ) × c( t ) .
The feedback loop reaches equilibrium when the edges of c( t ) and z( t ) are synchronized
and offset by Tb / 4 so product has zero area and v( t ) remains constant.
Many other approaches available including randomizing data so that long sequences of 1’s or
0’s are avoided, thus improving synchronization capabilities (see Carlson, Communication
Systems, McGraw-Hill, 3rd edition).

ES335 COMMUNICATION SYSTEMS 31 Dr S. Lawson

 z(t)
y(t) Z e ro Crossing v(t) c(t)
D e te c tor Inte gra tor VCC

c(t)

(a)

y(t)
Tb
Tb /2
z(t)

c(t)

z(t)c(t)

Tb /4

v(t)
(b)
Figure 51: Bit synchronization with a voltage controlled clock (a) Block diagram (b)
Waveforms

ES335 COMMUNICATION SYSTEMS 32 Dr S. Lawson