Chapter 7

Digital Representation of Analog

Signals

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 Contents

◊ 7.1 Introduction
◊ 7.2 Why Digitize Analog Sources ?
◊ 7.3 The Sampling Process
◊ 7.4 Pulse-Amplitude Modulation
◊ 7.5 Time-Division Modulation
◊ 7.6 Pulse-Position Modulation
◊ 7.8 The Quantization Process
◊ 7.9 Pulse-Code Modulation
◊ 7.10 Delta Modulation

2
 Chapter 7.1

Introduction
Wireless Information Transmission System Lab.
Institute of Communications Engineering National Sun
Yat-sen University
 7.1 Introduction

◊ In the first step from analog to digital, an analog source

is sampled at discrete times. The resulting analog
samples are then transmitted by means of analog pulse
modulation.
◊ Pulse-Amplitude Modulation (PAM), the simplest form of
analog pulse modulation.
◊ Pulse-Position Modulation (PPM)

◊ In the second step from analog to digital, an analog

source is not only sampled at discrete times but the
samples themselves are also quantized to discrete levels.
◊ Pulse-code Modulation (PCM)
◊ Delta Modulation (DM)
4
 Chapter 7.2
Why Digitize Analog Source ?

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.2 Why Digitize Analog Source ?
◊ Advantages of digital transmission over analog transmission:
◊ Digital systems are less sensitive to noise than analog. For long transmission
lengths, the signal may be regenerated effectively error-free at different point
along the path and the original signal transmitted over the remaining length.

◊ With digital systems, it is easier to integrate different services, for example,

video and the accompanying soundtrack, into the same transmission scheme.

◊ The transmission scheme can be relatively independent of the source. For

example, a digital transmission scheme that transmits voice at 10 kbps could
also be used to transmit computer data at 10 kbps.

◊ Circuitry for handling digital signals is easier to repeat and digital circuits are
less sensitive to physical effect such as vibration and temperature.

◊ Digital signals are simpler to characterize and typically do not have the same
amplitude range and variability as analog signals. This makes the associated
hardware easier to design. 6
 7.2 Why Digitize Analog Source ?
◊ Digital techniques offer strategies for more efficient use of media,
e.g. cable, radio wave, and optical fibers.
◊ Various media sharing strategies, known as multiplexing techniques, are
more easily implemented with digital transmission strategies.

◊ There are techniques for removing redundancy from a digital transmission,

so as to minimize the amount of information that has to be transmitted. These
techniques fall under the broad classification of source coding and we
discuss some of these techniques in Chapter 10.

◊ There are techniques for adding controlled redundancy to digital transmission,

such that errors occur during transmission may be corrected at the receiver
without any additional information. These techniques fall under the general
category of channel coding, which is described in Chapter 10.

7
 7.2 Why Digitize Analog Source ?
◊ Digital techniques make it easier to specify complex standards that may be
shared on a worldwide basis. This allows the development of communication
components with many different features (e.g., a cellular handset) and their
interoperation with a different component (e.g., a base station) produced by a
different manufacturer.

◊ Other channel compensations techniques, such as equalization, especially

adaptive versions, are easier to implement with digital transmission techniques.

◊ It should be emphasized that the majority of these advantages for

digital transmission rely on availability of low-cost microelectronics.

8
 Chapter 7.3
The Sampling Process

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.3 The Sampling Process
◊ Through use of the sampling process, an analog signal is converted
into a corresponding sequence of samples that are usually spaced
uniformly in time.
◊ It is necessary that we choose the sampling rate properly, so that
the sequence of samples uniquely defines the original analog
◊ signal.
Let gδ(t) denote the ideal sampled signal


7.1
 g nTs  t  nTs
g t   n
◊ We refer to Ts as the sampling period, fs = 1/Ts as the
sampling rate.  

10
 7.3 The Sampling Process
 

◊ Applying Eq. (2.88), we get the result  g t  mT  :

0 f0 
G nf   f  nf 
0 0

2.88 
(7.2)
g t  : fs
m n

m

where G( f ) is the Fourier transform of the original

signal g(t) 
andGfs f  mfs 
is the sampling rate.
◊ Eq. (7.2) state that the process of uniformly sampling a continuous-
times signal of finite energy results in a periodic spectrum with a
period equal to the sampling rate.

◊ Taking the discrete-time Fourier transform of both sides of Eq.


(7.1), we get
 g nTs exp  j2 nfTs
G  f   n 7.3
11


 7.3 The Sampling Process
◊ Hence, under the following two conditions
1. G f 0 for f (Band-Limited Signal)
W 1
⎛
2. fs  2W⎜ ⎝ or Ts 
⎞ ⎟
2W
we can get (from
⎠ Eq. (7.3))

G  f   g ⎛⎜⎝ n ⎞
⎟
⎠
exp
⎛ jπnf ⎞
⎜
⎝
 W 7.4 
 2W
n
⎠
⎟
◊ From Eq. (7.2), we readily see that the Fourier transform of gδ(t)
may also be expressed as


 G  f  mfs 
G  f   f s G  f   f sm 7.5 
m0
12
 7.3 The Sampling Process

◊ We find from Eq. (7.5) that

1
G  f   2 G  f , W  f  7.6 
W W

13
 7.3 The Sampling Process

◊ Substituting Eq. (7.4) in Eq. (7.6), we may also write


1
Gf  ⎛ n ⎞
g⎜ exp
⎛ jπnf ⎞ W  f 
 , 7.7 
2 n ⎝ 2W ⎠⎟ ⎝⎜ W W
W ⎠
(Physical
⎟ meaning?)
◊ Therefore, if the sample values of a signal g(t) are
specified for all time, then the Fourier transform G( f ) of
the signal is uniquely determined by using the discrete-
time Fourier transform of Eq. (7.7) . In the other words,
the sequence {g(n/2W)} has all the information contained
in g(t).

14
 7.3 The Sampling Process
◊ Reconstructing the signal of g(t)
◊ Substituting Eq. (7.7) in the formula for the inverse Fourier
transform g(t) in terms of G( f ), we get

g t   G  f  exp   ft df

  
j2
1

W
 ⎛ n ⎞ ⎛ j nf ⎞
W
n
g ⎜
⎝ 2W ⎠⎟ exp ⎝⎜  W exp j2 ft  df
◊ Interchanging the order ⎠
⎟ of summation and integration
2W

W
g t   ⎛ n ⎞ 1
 ⎡ ⎛ n ⎞⎤
g ⎜ 2W W exp ⎢ j2 f ⎜ t 
n ⎝ ⎣ ⎝ 2W ⎟⎥ df 7.8

 ⎟
⎠ 2W
⎛ n ⎞ sin  2Wt  n  ⎠⎦
g t   g  n
⎛ ⎞ -t
⎜⎝ 2W ⎟ g ⎜⎝ 2W ⎟sinc 2Wt  
n
 2Wt  n n
n(Physical meaning?)
⎠ 15 ⎠ 7.9

 7.3 The Sampling Process

◊ Reconstructing the signal of g(t)

◊ Eq. (7.9) provides an interpolation formula for reconstructing the

original signalfrom the sequence of sample values {g(n/2W)},
with the sinc(2Wt) playing the role of an interpolation function.

◊ Eq. (7.9) can be looked in another way: it represents the

convolution (or filtering) of the impulse train gδ(t) given by Eq.
(7.1) with the impulse response sinc(2Wt).

◊ Any impulse response that plays the same roles as sinc(2Wt) is

also referred to as a reconstruction filter.

16
 7.3 The Sampling Process

◊ The sampling theorem for strictly band-limited signals of

finite energy may be stated in two equivalent parts
◊A band-limited signal of finite energy, which only has frequency components
less than W Hz, is completely described by specifying the values of the signal
at instants of time separated by 1/2W seconds.
◊ A band-limited signal of finite energy, which only has frequency components

less than W Hz, may be completely recovered from a knowledge of its samples
taken at the rate of 2W samples per second.

◊ The sampling rate of 2W samples per second, for a signal

bandwidth of W Hz, is called the Nyquist rate; its
reciprocal 1/2W (measured in seconds) is called Nyquist
interval.

17
 7.3 The Sampling Process

◊ In practice, however, an information-bearing signal is not

strictly band-limited, with the result that some degree of
undersampling is encountered. Consequently, some
aliasing is produced by the sampling process.

18
 7.3 The Sampling Process

◊ To combat the effects of aliasing in practice, we may use

two corrective measures, as described here
◊ Prior to sampling, a low-pass pre-alias filter is used to attenuate those high-
frequency components of the signal that are not essential to the information
being conveyed by the signal.
◊ The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

◊ The use of a sampling rate higher than Nyquist rate also

has beneficial effect of easing the design of the
reconstruction filter used to recover the original signal
from its sampled version.

19
7.3 The Sampling Process

◊ The reconstruction
filter is low-pass with
a passband
extending from –W
to W.
◊ The reconstruction
filter has a transition
band extending
from W to fs-W.
◊ The fact that the
reconstruction filter
has a well defined
transition band means
that it is physically
20 realizable.
Nyquist Sampling Theorem

S  1
X ( f )  X ( f )  X ( f ) T  X ( f  nf )
S
S n

21
Spectra for Various Sampling Rates

22
Natural Sampling

23
 Chapter 7.4
Pulse-Amplitude Modulation

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.4 Pulse-Amplitude Modulation
◊ In pulse-amplitude modulation (PAM), the amplitudes of regularly
spaced pulses are varied in proportion to the corresponding
sample values of a continuous message signal; the pulses can be
of a rectangular form or some other shape.
◊ PAM is somewhat similar to natural sampling, where the message
signal is multiplied by a periodic train of rectangular pulses.
◊ In natural sampling the top of each modulated rectangular pulse
varies with the message signal. In PAM it is maintained flat.

Fig. 7.5 Flat-top samples.

25
 7.4 Pulse-Amplitude Modulation

◊ Two operations are involved in the generation of the PAM signal

◊ Instantaneous sampling of the message signal m(t) every Ts

seconds, where the sampling rate fs = 1/Ts is chosen in

accordance with the sampling theorem.
◊ Lengthening the duration of each sample so obtained to

some constant value T.

◊ In digital circuit technology, these two operations are jointly referred

to as “sample and hold.”

◊ One important reason for intentionally lengthening the duration of

each sample is to avoid the use of an excessive channel
bandwidth, since bandwidth is inversely proportional to pulse
duration. 26
 Chapter 7.5
Time-Division Multiplexing

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.5 Time-Division Multiplexing
◊ The sampling theorem provides the basis for transmitting the
information contained in a band-limited message signal m(t) as a
sequence of samples of m(t) taken uniformly at a rate that is slightly
higher than the Nyquist rate.

◊ An important feature of the sampling process is a conservation of

time. That is, the transmission of the message samples engages the
communication channel for only a fraction of the sampling
interval on a periodic basis, and in this way some of the time
interval between adjacent samples is cleared for use by other
independent message sources on a time-shared basis.

◊ We thereby obtain a time-division multiplex (TDM) system, which

enables the joint utilization of a common communication channel
by a plurality of independent message sources without mutual
interference among them.
28
 7.5 Time-Division Multiplexing
Figure 7.9 Block diagram of TDM System

◊ Low-pass pre-alias filter:

◊ To remove the frequencies that are nonessential to an adequate signal
representation.
◊ Commutator:
◊ To take a narrow sample of each of the N input messages at a rate fs that is
slightly higher than 2W, where W is the cut-off frequency of the pre-alias filter.
◊ To sequentially interleave these N samples inside the sampling interval Ts.

29
 7.5 Time-Division Multiplexing

◊ Pulse modulator:
◊ To transform the multiplexed signal into a form suitable for transmission over
the common channel.
◊ The use of TDM introduces a bandwidth expansion factor N, because the
scheme must squeeze N samples derived from N independent message sources
into a time slot equal to one sampling interval.

◊ Pulse demodulator :
◊ Performs the reverse operation of the pulse modulator.

◊ Decommutator:
◊ The narrow samples produced at the pulse demodulator output are distributed
to the appropriate low-pass reconstruction filters.
◊ In synchronism with commutator in the transmitter.
30
 Chapter 7.6
Pulse-Position Modulation

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.6 Pulse-Position Modulation
◊ In a pulse modulation system, we may use the increased bandwidth
consumed by pulses to obtain an improvement in noise
performance be representing the sample values of the message
signal by some property of the pulse other than amplitude.

◊ Pulse-Duration Modulation (PDM): the samples of the message

signal are used to vary the duration of the individual pulses. This
form of modulation is also referred to as Pulse-Width Modulation
(PWM) or Pulse-Length Modulation (PLM).
◊ In PDM, long pulses expend considerable power during the pulse

while bearing no addition information.

◊ Pulse-Position Modulation (PPM): the position of a pulse relative

to its unmodulated time of occurrence is varied in accordance with
the message signal.
32
7.6 Pulse-Position Modulation

Modulating
Wave

Pulse Carrier

PDM Signal

PPM Signal

33
 7.6 Pulse-Position Modulation

◊ Mathematical Representation of PPM Signal

◊ Using the sample m(nTs) of a message signal m(t) to modulate the
position of nth pulse, we obtain the PPM signal

s t   g t  nT  k m nT
s
p s 7.20 

n

where kp is the sensitivity of the pulse-position modulator and g(t)

 of interest.
denotes a standard pulse
◊ The different pulses constituting the PPM signal s(t) must be
strictly non-overlapping.
◊ A sufficient condition is given by: T /2 s Kp|m(t)| max
Ts/2
T
g t   t  2s  kp m t max 7.21
0, kp
 m34t max
2
7.22 
which in turn requires
 7.6 Pulse-Position Modulation
◊ Generation of PPM waves

◊ The message signal m(t) is first converted in to a PAM signal

by means of a sample-and-hold circuit, generating a staircase
waveform u(t).

◊ Next, the signal u(t) is added to a sawtooth wave, yielding the

combined signal v(t).

35
 7.6 Pulse-Position Modulation
(a) Message signal
◊ The v(t) is applied to
(b) Staircase a threshold detector that
approximation of
the message signal
produces a very narrow pulse
(approximating an impulse)
each time v(t) crosses zero
(c) Sawtooth wave in the negative-going
direction.
◊ Finally, the PPM signal s(t) is
generated by using this
(d) Composite sequence of impulses to
wave excite a filter whose
(e) Sequence of impulse response is defined
Impulses used to by the standard pulse g(t).
generate the PPM
signal
36
 Chapter 7.8
The Quantization Process

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.8 The Quantization Process

◊ A continuous signal has infinite number of amplitude levels.

◊ It is not necessary in fact to transmit the exact amplitudes of the
samples.
◊ Any human sense can detect only finite intensity differences.
◊ The original continuous signal may be approximated by a signal
constructed of discrete amplitudes selected on a minimum error
basis from an available set.
◊ Amplitude quantization is defined as the processes of transforming
the sample amplitude m(nTs) of message signal m(t) at time t =
nTs into a discrete amplitude v(nTs) taken from a finite set of
possible amplitudes.
◊ We assume that the quantization process is memoryless and
instantaneous.
38
 7.8 The Quantization Process
◊ Since we are dealing with a memoryless quantizer, we may use the
symbol m in place of m(nTs).

◊ The signal amplitude m is specified by the index k if it lies inside

the interval, where L is the total number of amplitude levels used in
quantizer.
the
k : mk  m  mk 1, k  1, 2,…, L

7.32 

39
 7.8 The Quantization Process

◊ The mk are called decision levels or decision thresholds.

◊ The vk are called representation levels or reconstruction levels.

◊ The spacing between two adjacent representation levels is called a

quantum or step-size.

◊ The mapping v=g(m) is the quantizer characteristic, which is

a staircase function by definition.

◊ Quantizers can be uniform or nonuniform.

◊ In a uniform quantizer, the representation levels are uniformly

spaced, otherwise, the quantizer is non-uniform.

◊ The quantizer characteristic can be of midtread or midrise type.

40
7.8 The Quantization Process

Midtread type Midrise type

41
 7.8 The Quantization Process
◊ Quantization Noise
◊ The use of quantization introduces an error defined as the
difference between the input signal m and the output signal v.
◊ This error is called quantization noise.

42
 7.8 The Quantization Process
◊ For simplicity, let the quantizer input m be the sample value of a
zero-mean random variable M.
◊ A quantizer g( ) maps the input random variable M of continuous
amplitude into a discrete random variable V.
◊ Let the quantization error be denoted by the random variable Q of
sample value q. We may thus write
qm
or 7.34 
Q M V
7.35
◊ With the input M having zero mean, and the quantizer assumed to
be symmetric, it follows that the quantizer output V and therefore
the quantization error Q will also have zero mean.
◊ In order to find the output signal-to-noise ratio, we need to find the
mean-square value of the quantization error Q.
43
 7.8 The Quantization Process
◊ Consider an input m of continuous amplitude in the range (-mmax,
mmax). Assuming a uniform quantizer of the midrise type, we find the
step-size of the quantizer is given by
2mmax
Δ L 7.36
where L is the total number of representation levels.
◊ For a uniform quantizer, the quantization error Q will have its
sample values bounded by −Δ/2 ≤ q ≤ Δ/2 .
◊ If Δ is sufficiently small or L is sufficiently large, it is reasonable to
assume that the quantization error Q is a uniform distributed random
variable. The probability density function of the quantization error
is
Q Δ
⎧ 1   q 
, 2 7.37
⎨⎪
f Q q  Δ Δ
⎪⎩  44 2
0, otherwise
 7.8 The Quantization Process
◊ Its variance is Δ2
 
2
q 2 fQ q dq  E ⎡Q2
Q
7.38
Δ 2
 ⎤

◊ Substituting Eq. (7.37) in (7.38), we get

1 Δ2 2 7.39 
   q dq 
2 2
Δ
Q
Δ Δ 2 12
◊ Typically, the L-ary number k, denoting the kth representation level
of the quantizer, is transmitted to the receiver in binary form.
◊ Let R denote the number of bits per sample used in the construction
of the binary code. 7.40
L  2R

R  log2 L
7.41
45

 7.8 The Quantization Process
◊ Substituting Eq. (7.40) in (7.36)
7.42
Δ max
L max
2R
◊ Thus, the use of Eq. (7.42) in (7.39) yields

 Q2 
Δ2

1 mmax
2
R 2 2
7.43
◊ The output signal-to-noise 12 3 a uniform quantizer is given by
ratio of

⎛ 3P
P 2
2 7.44 
SNR O   Q  ⎜⎞⎝ mmax
2 2 R

⎟
⎠
◊ Eq. (7.44) shows that the output SNR of the quantizer increases
exponentially with increasing of bits per sample.

46
 7.8 The Quantization Process

◊ Example 7.2 : Sinusoidal Modulating Signal

◊ Consider a full-load sinusoidal modulation signal of amplitude
Am , which utilizes all the representation level provided. The
average signal power is m2
P A
2
◊ The range of quantizer input is between

◊ The quantization noise as−Am and Am.

 Q2  1 A2 2m2 R
◊ Output SNR is 3

2 2 3 2R
SNR m 
A
2
 2
7.45
O
Am2 22 R 3

 47
 7.8 The Quantization Process
◊ Expressing the SNR in decibels, we get
10 log10 SNR
7.46 
◊ For various values of L and
 1.8 6R R, the corresponding values of SNR
O
are as given in the table as follows.

Number of
Representation Number of Signal-to-Noise
Levels, L Bits/Sample, R Ratio, (dB)
32 5 31.8
64 6 37.8
128 7 43.8
256 8 49.8

48
 Chapter 7.9
Pulse-Code Modulation

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.9 Pulse-Code Modulation
◊ In pulse-code modulation (PCM) a message signal is represented by
a sequence of coded pulses, which is accomplished by representing
the signal in discrete form in both time and amplitude.

50
 7.9 Pulse-Code Modulation

◊ Sampling
◊ The incoming message signal is sampled with a train of narrow
rectangular pulses so as to close approximate the instantaneous
sampling process. Sampling rate must be greater than 2W.
◊ A pre-alias filter is used at the front end of the sampler in order to

exclude frequencies greater than W before sampling.

◊ Quantization
◊ The quantization process may follow a uniform law as described
in the previous section.
◊ Unacceptable signal-to-noise ratio for small signals.

◊ Solution: Increasing quantization levels – price is too high.

51
 7.9 Pulse-Code Modulation
◊ In certain applications, it is preferable to use a variable separation
between the representation levels.
◊ The use of a nonuniform quantizer is equivalent to pass the

baseband signal through a compressor and applying the

compressed signal to a uniform quantizer.
◊ In order to restore the signal samples to their correct relative level,

we must use a device in the receiver with a characteristic

complementary to the compressor. Such a device is called an
expander.
◊ The compression and expansion laws are exactly inverse so that,

except for the effect of quantization, the expander output is equal

to the compressor input.
◊ The combination of a compressor and an expander is called a

compander.
52
 7.9 Pulse-Code Modulation
◊ Two major types of compression law
◊ μ - law (usually μ = 255; used in US, Canada,
Japan)
log 1 μ m
  7.47
log 1
◊ A - law (usually A = 87.6; used in Europe)

⎧ A mμ
⎪ 1
1 log , 0  m  A
 ⎨ A 7.48
1 m1
⎪
⎪ 1 log
⎪ 1 log  
A m ,
A
A
◊ m and v are the normalized input and output voltages, respectively.
⎩
◊ The case of uniform quantization corresponds to μ=0 and A=1.
53
7.9 Pulse-Code Modulation

μ - law A - law

54
 7.9 Pulse-Code Modulation

◊ Encoding
◊ In combining the processes of sampling and quantizing, the
specification of a continuous message (baseband) signal becomes
limited to a discrete set of values, but not in the form best suited
to transmission over a line or radio path.

◊ To exploit the advantages of sampling and quantizing for the

purpose of making the transmitted signal more robust to noise,
interference, and other channel degradations, we require the use
of an encoding process to translate the discrete set of sample
values to a more appropriate form of signal.

◊ Any plan for representing each of this discrete events is

called a
code. 55
 7.9 Pulse-Code Modulation

◊ Line Codes
◊ Unipolar Nonreturn-to-Zero (NRZ)
signaling

◊ Polar Nonreturn-to-Zero (NRZ)

signaling

◊ Unipolar Return-to-Zero (RZ) signaling

◊ Bipolar Return-to-Zero (BRZ) signaling

◊ Alternate Mark Inversion
(AMI)

◊ Split-Phase (Manchester Code)

56
 7.9 Pulse-Code Modulation

◊ Unipolar Nonreturn-to-Zero (NRZ) signaling

◊ Symbol 1 is represented by transmitting a pulse of amplitude
A for the duration of the symbol, and symbol 0 is represented
by switching off the pulse.
◊ This line code is also referred to as on-off signaling.

◊ A disadvantage of on-off signaling is the waste of power due

to the transmitted DC level.

◊ Polar Nonreturn-to-Zero (NRZ) signaling

◊ Symbol 1 and 0 are represented by transmitting pulses of
amplitudes +A and –A.
◊ This line code is relatively easy to generate and is more power-
efficient than its unipolar counterpart.
57
 7.9 Pulse-Code Modulation

◊ Unipolar Return-to-Zero (RZ) signaling

◊ Symbol 1 is represented by a rectangular pulse of amplitude A
and half-symbol width, and symbol 0 is represented by
transmitting no pulse.
◊ An attractive feature is the presence of delta function at f
= 0,
±1/Tb in the power spectrum of the transmitted signal, which can
be used for bit-timing recovery at the receiver.
◊ It requires 3 dB more power than polar return-to-zero signaling

for the same probability of symbol error.

◊ Bipolar Return-to-Zero (BRZ) signaling

◊ This line code uses three amplitude levels. (0, ±A)
◊ No pulse is always used for
58 symbol 0.
 7.9 Pulse-Code Modulation
◊ Positive and negative pulses of equal amplitude (+A and –A) are
used alternately for symbol 1, with each pulse having a half-
symbol width.
◊ The transmitted signal has no DC components.
◊ Also called alternate mark inversion (AMI) signaling.

◊ Split-Phase (Manchester Code)

◊ Symbol 1 is represented by a positive pulse of amplitude A
followed by a negative pulse of amplitude −A, with both pulses
being a half-symbol wide.
◊ For symbol 0, the polarities of these two pulses are
reversed.
◊ The Manchester code suppress the DC component and has

relatively insignificant low-frequency components, regardless of

59
the signal statistics.
 7.9 Pulse-Code Modulation
◊ Differential Encoding
◊ This method is used to encode information in terms of signal transitions.
◊ A transition is used to designate symbol 0 in the incoming binary data stream,
while no transition is used to designate symbol 1.
◊ The waveform of the differentially encoded data assumes the use of unipolar
nonreturn-to-zero signaling.
◊ The original binary information is recovered by comparing the polarity of
adjacent binary symbols to establish whether or not a transition has occurred.
◊ Notice that differential encoding requires the use of a reference bit before
initiating process.

60
 7.9 Pulse-Code Modulation

1
0 1 1 0 0 0 1 1 0 1
+E

-E

◊ Polar - RZ
◊ “One” and “Zero” are represented by opposite
level polar pulses that are one half-bit in width.

61
 7.9 Pulse-Code Modulation

1
0 1 1 0 0 0 1 1 0 1
+E

-E

◊ Bi-φ-M ( Biphase Mark or Manchester 1)

◊ A transition occurs at the beginning of every bit
period.
◊ “One” is represented by a second transition one half bit

period later.
◊ “Zero” is represented
62
by no second transition.
 7.9 Pulse-Code Modulation

1
0 1 1 0 0 0 1 1 0 1
+E

-E

◊ Dicode Non-Return-to-Zero
◊ A “One” to “Zero” or “Zero” to “One” changes
polarity.
◊ Otherwise, a “Zero” is sent.

63
 7.9 Pulse-Code Modulation

1
0 1 1 0 0 0 1 1 0 1
+E

-E

◊ Dicode Return-to-Zero
◊ A “One” to “Zero” or “Zero” to “One” transition produces
a half duration polarity change.
◊ Otherwise, a “Zero” is sent.

64
 7.9 Pulse-Code Modulation

1
0 1 1 0 0 0 1 1 0 1
+E

-E

◊ Dicode Non-Return-to-Zero
◊ A “One” is represented by a transition at the midpoint of
the bit interval.
◊ A “Zero” is represented by a no transition unless it is
followed by another zero. In this case, a transition is
placed at the end of bit period of the first zero.
65
7.9 Pulse-Code Modulation

O -
-

66
 7.9 Pulse-Code Modulation

◊ 4B3T Line Code

◊ Ternary words in the middle column are balanced in their DC
content.
◊ Code words from the first and third columns are selected
alternately to maintain DC balance.
◊ If more positive pulses than negative pulses have been
transmitted, column 1 is selected.
◊ Notice that the all-zeros code word is not used.

67
Spectral Densities of Various PCM Waveforms

68
 7.9 Pulse-Code Modulation

◊ Criteria for Selecting PCM Waveform

◊ DC component: eliminating the dc energy from the signal’s
power spectrum.
◊ Self-Clocking: Symbol or bit synchronization is required for
any digital communication system.
◊ Error detection: some schemes provide error detection without
introducing additional error-detection bits.
◊ Bandwidth compression: some schemes increase bandwidth
utilization by allowing a reduction in required bandwidth for a
given data rate.
◊ Noise immunity.
◊ Cost and complexity.

69
 7.9 Pulse-Code Modulation
◊ Regeneration
◊ The most important feature of any digital system lies in the
ability to control the effects of distortion and noise produced by
transmitting a digital signal through a channel.
◊ This capability is accomplished by reconstructing the signal by
means of a chain of regenerative repeaters, which perform three
basic functions: equalization, timing, and decision making.

70
 7.9 Pulse-Code Modulation
◊ Equalizer: shapes the received pulses so as to compensate for the
effects of amplitude and phase distortions produced by the channel.
◊ Timing Circuitry: provides a periodic pulse train, derived from the
received pulses, for sampling the equalized pulses at the instants of
time where the signal-to-noise ratio is a maximum.
◊ Decision-making Device: the sample extracted is compared to a
predetermined threshold.

◊ The regenerated signal may depart from the original signal:

◊ The unavoidable presence of channel noise and interference

causes the repeater to make wrong decisions occasionally,

thereby introducing bit error in to regenerated signal.
◊ If the spacing between received pulses deviates from its

assigned value, a jitter is introduced into the regenerated pulse

position, thereby causing distortion.
71
 7.9 Pulse-Code Modulation
◊ Decoding
◊ The first operation in the receiver is to regenerate the received
pulse one last time.
◊ These clean pulses are then regrouped into code words and
decoded in to a quantized PAM signal.
◊ The decoding process involves generating a pulse the amplitude
of which is the linear sum of all the pulses in the code word, with
each pulse being weighted by its place value (20,21,…,2R-1) ,
where R is the number of bits per sample.
◊ Filtering
◊ The final operation in the receiver is to recover the message
signal wave by passing the decoder output through a low-pass
reconstruction filter whose cutoff frequency is equal to the
message bandwidth W.
72
 Chapter 7.10
Delta Modulation

Wireless Information Transmission System Lab.

Institute of Communications Engineering National Sun
Yat-sen University
 7.10 Delta Modulation
◊ In some applications, the increased bandwidth requirement of PCM
is a major concern.
◊ In delta modulation (DM), an incoming message signal is
oversampled (at a rate much higher than the Nyquist rate) to
purposely increase the correlation between adjacent samples of the
signal.

74
 7.10 Delta Modulation
◊ The difference between the input and the approximation is
quantized into only two levels, namely, ± Δ, corresponding
to positive and negative differences, respectively.
◊ Provided that the signal dose not change too rapidly from sample to
sample, we find that the staircase approximation remains within ±
Δ of the input signal.

◊ The delta modulator using a fixed step-size is often referred to as a

linear delta modulator.

◊ The principle virtue of DM is its simplicity.

◊ DM is subject to two types of quantization error:

◊ Slope Overload Distortion
◊ Granular Noise
75
 7.10 Delta Modulation
◊ Slope-overload distortion occurs when the Δ is too small relative to
the local slope of m(t). In order for the sequence of samples to
increase as fast as the input sequence, we require that Δ dmt 
 max
Ts

dt
◊ Granular noise occurs when the Δ is too large relative to the local
slope of m(t).

To improve performance, we need to

Make the delta modulator adaptive, in
the sense that the step-size is made to
vary in accordance with the input signal.
76
 7.10 Delta Modulation
◊ Delta-Sigma Modulation
◊ A drawback of delta modulation in that transmission disturbances
such as noise result in an accumulative error in the modulated
signal.
◊ This drawback can be overcome by integrating the message

signal prior to delta modulation.

◊ A DM scheme that incorporates integration at its input
is called
delta-sigma modulation or sigma-delta modulation.
◊ The reason for investing delta modulation is its reduced
bandwidth requirements compared to PCM.
◊ For telephone applications, a typical PCM system may use an 8-kHz
sampling rate with an 8-bit representation for an overall binary symbol rate
of 64 kHz.
Typical oversampling rate for delta 77
modulation range from 16 to 32 kHz. Thus
DM may provide a net bandwidth saving of 50 to 75 percent over PCM.