You are on page 1of 41

Introduction to Analog And Digital

Communications

Telecomunicações

Based on the slides produced by: Simon Haykin, Michael Moher


Chapter 5 Pulse Modulation :
Transition from Analog to Digital Communications

5.1 Sampling Process


5.2 Pulse-Amplitude Modulation
5.4 Completing the Transition from
Analog and Digital
5.5 Quantization Process
5.6 Pulse-Code Modulation
5.9 Line Codes
5.11 Summary and Discussion
3

• Some parameter of a pulse train is varied in accordance with the message signal
• Analog pulse modulation
• A periodic pulse train is used as the carrier wave
• Some characteristic feature of each pulse is varied in a continuous manner in accordance with
the corresponding sample value of the message signal
• Digital pulse modulation
• The message signal is represented in a form that is discrete in both time and amplitude
• Its transmission in digital form as a sequence of coded pulse
• Lesson1 : Given a strictly band-limited message signal, the sampling theorem embodies the
conditions for a uniformly sampled version of the signal to preserve its information content
• Lesson2 : Analog pulse-modulation systems rely on the sampling process to maintain continuous
amplitude representation of the message signal. In contrast, digital pulse-modulation system use not
only the sampling process but also the quantization process. Digital modulation makes it possible to
exploit the full power of digital signal-processing techniques.
4

5.1 Sampling Process

• Instantaneous Sampling and Frequency-Domain Consequences


• Sample the signal g(t) instantaneously and at a uniform rate,
• Instantaneously (ideal) sampled signal
• The signal obtained by individually weighting the elements of a periodic sequence
of Dirac delta functions :
¥

gd (t ) = å g (nTs )d (t - nTs ) (5.1)


n = -¥

• Reproduce the relationships listed at the bottom of the right-hand side of the table 5.1
• The process of uniformly sampling a continuous time signal of finite energy results
in a periodic spectrum with a repetition frequency equal to the sampling rate.

¥ ¥ ¥

å g (nT )d (t - nT ) Û f å G( f - mf ) = å g (nT ) exp(- j 2pnT f ) = G ( f )


n = -¥
s s s
n = -¥
s
n = -¥
s s d (5.2)
5

Fig.5.1
6

table.5.1
7

• Sampling Theorem
• A discrete-time Fourier transform of the sequence

æ jpnf ö
¥
æ n ö
Gd ( f ) = å g ç ÷ expç - ÷ (5.3)
n = -¥ è 2W ø è W ø

Gd ( f ) = f s G ( f ) + f s å G( f - mf )
m = -¥
s

m¹0

• For a strictly band-limited signal, under the two conditions


1.G ( f ) = 0 for f ³ W
2. f s = 2W
1
G( f ) = Gd ( f ), - W < f < W (5.4)
2W
1 ¥ æ n ö æ jpnf ö
G( f ) = å g ç
2W n = -¥ è 2W ø
÷ expç-
è W ø
÷, - W < f < W (5.5)
8

Fig.5.2
9

• The sampling theorem for strictly band-limited signals of finite energy in two equivalent parts
• Analysis : A band-limited signal of finite energy that has no frequency components higher
than W hertz is completely described by specifying the values of the signal at instants of time
separated by 1/2W seconds.
• Synthesis : A band-limited signal of finite energy that has no frequency components higher
than W hertz is completely recovered form knowledge of its samples taken at the rate of 2W
samples per second.

• Nyquist rate
• The sampling rate of 2W samples per second for a signal bandwidth of W hertz
• Nyquist interval
• 1/2W (measured in seconds)
10

• Aliasing Phenomenon
• The phenomenon of a high-frequency component in the spectrum of the signal seemingly taking on
the identify of a lower frequency in the spectrum of its sampled version.
• To combat the effects of aliasing in practices
• Prior to sampling : a low-pass anti-alias filter is used to attenuate those high-frequency
components of a message 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.

• Physically realizable reconstruction filter


• The reconstruction filter is of a low-pass kind with a passband extending from –W to W
• The filter has a non-zero transition band extending form W to fs-W
11
12

Fig.5.4
13

5.2 Pulse-Amplitude Modulation

• Pulse-Amplitude Modulation (PAM)


• The amplitude of regularly spaced pulses are varied in proportion to the corresponding sample
values of a continuous message signal.
• Two operations involved in the generation of the PAM signal
• Instantaneous sampling of the message signal m(t) every Ts seconds,
• Lengthening the duration of each sample, so that it occupies some finite value T.
14

• Sample-and-Hold Filter : Analysis


• The PAM signal is ¥

s (t ) = å m(nTs )h(t - nTs ) (5.8)


n = -¥

• The h(t) is a standard rectangular pulse of unit amplitude and duration

æ T ö ì1, 0 < t < T


ç t - ÷ ïï 1
h(t ) = rect ç 2 ÷ = í , t = 0, t = T (5.9)
çç T ÷÷ ï 2
è ø ïî0, otherwise

• The instantaneously sampled version of m(t) is


¥

md (t ) = å m(nTs )d (t - nTs ) (5.10)


n = -¥
15

Fig.5.5
16

• To modify mδ(t) so as to assume the same form as the PAM signal


¥
md (t ) * h(t ) = ò md (t )h(t - t )dt

¥

å m(nT )d (t - nT )h(t - t )dt


¥
=ò s s

n = -¥
¥

= å m(nTs ) ò d (t - nTs )h(t - t )dt


¥
(5.11)

n = -¥
¥

ò-¥
d (t - nTs )h(t - t )dt = h(t - nTs )
• The PAM signal s(t) is mathematically equivalent to the convolution of mδ(t) , the instantaneously sampled
version of m(t), and the pulse h(t)
¥

md (t ) * h(t ) = å m(nTs )h(t - nTs ) (5.12)


n = -¥
¥
s (t ) = md (t ) * h(t ) (5.13) M d ( f ) = f s å M ( f - kf s ) (5.15)
¥k = -¥
S ( f ) = M d ( f ) H ( f ) (5.14) S ( f ) = f s å M ( f - kf s ) H ( f ) (5.16)
k = -¥
17

Fig.5.6
18

5.4 Completing the Transition from Analog to Digital


• The advantages offered by digital pulse modulation
• Performance
• Digital pulse modulation permits the use of regenerative repeaters, when placed along the transmission
path at short enough distances, can practically eliminate the degrading effects of channel noise and signal
distortion.
• Reliability
• Can be made highly reliable by exploiting powerful error-control coding techniques.
• Efficiency
• Inherently more efficient than analog communication system in the tradeoff between transmission
bandwidth and signal-to-noise ratio
• System integration
• To integrate digitized analog signals with digital computer data
19

5.5 Quantization Process

• Amplitude quantization
• The process of transforming the sample amplitude m(nTs) of a baseband signal m(t) at time
t=nTs into a discrete amplitude v(nTs) taken from a finite set of possible levels.

I k : {mk < m £ mk +1}, k = 1,2,..., L (5.21)

• Representation level (or Reconstruction level)


• The amplitudes vk , k=1,2,3,……,L
• Quantum (or step-size)
• The spacing between two adjacent representation levels
v = g (m) (5.22)
20

Fig.5.9
21

Fig.5.10
22

5.6 Pulse-Code Modulation


• PCM (Pulse-Code Modulation)
• 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
• The basic operation
• Transmitter : sampling, quantization, encoding
• Receiver : regeneration, decoding, reconstruction

• Operation in the Transmitter


1. Sampling
1. The incoming message signal is sampled with a train of rectangular pulses
2. The reduction of the continuously varying message signal to a limited number of discrete values per
second
2. Nonuniform Quantization
1. The step size increases as the separation from the origin of the input-output amplitude characteristic is
increased, the large end-step of the quantizer can take care of possible excursions of the voice signal
into the large amplitude ranges that occur relatively infrequently.
23

• Compressor
• A particular form of compression law : μ-law

log(1 + µ m )
v= (5.23)
log(1 + µ )
d m log(1 + µ )
= (1 + µ m ) (5.24)
dv µ
• μ-law is neither strictly linear nor strictly logarithmic
ì Am 1
ï1 + log A , 0 £ m £
• A-law : ï A
v =í (5.25)
ï 1 + log( A m ) 1
, £ m £1
ïî 1 + log A A
ì1 + log A 1
, 0 £ m £
d m ïï A A
=í (5.26)
dv ï 1
ïî(1 + log A) m , A £ m £ 1
24

Fig.5.11
25

3. Encoding
1. To translate the discrete set of sample vales to a more appropriate form of signal
2. A binary code
§ The maximum advantage over the effects of noise in a transmission medium is obtained by
using a binary code, because a binary symbol withstands a relatively high level of noise.
§ The binary code is easy to generate and regenerate
26

Fig.5.12
27

table.5.2
28

• Operations in the Receivers


1. Decoding and expanding
1. Decoding : regenerating a pulse whose amplitude is the linear sum of all the pulses in
the code word
2. Expander : a subsystem in the receiver with a characteristic complementary to the
compressor
1. The combination of a compressor and an expander is a compander

2. Reconstruction
1. Recover the message signal : passing the expander output through a low-pass
reconstruction filter
29

5.9 Line Codes (Baseband Digital Transmission) – Electrical Representation


of a Binary Sequency

• Several line codes (PAM formats)

1. On-off signaling

2. Nonreturn-to-zero (NRZ)

3. Return-to-zero

4. Bipolar return-to-zero (BRZ)

5. Split-phase (Manchester code)


30

Fig.5.20
31

Advantages and Disadvantages (1)

• A line code should have particular properties to be practical to use. Some of these properties
are:

1. Transmission Bandwidth: Bandwidth should be as small as possible.

2. Power Efficiency: For a data rate and error probability, transmitted power should be as
small as possible.

3. Error Detection Capability: Should be able to detect and possibly correct errors. (Bipolar
RZ, Bipolar NRZ)

4. Favorable Power Spectral Density: A zero PSD (power spectral density) at f=0 would be
best due to ac coupling. (heats up the wires.)
32

Advantages and Disadvantages (2)

5. Adequate Timing Content: The line code should have extractable clock information from
the signal/data. Frequent line voltage transitions, directly proportional to the clock rate,
helps clock recovery. (e.g. Manchester, Bipolar (RZ+NRZ) - guarantees transitions for
timing recovery with long runs of ones.

6. Transparency: Transparency is defined as a line code in which the bit pattern does not
affect the accuracy of the timing. A transmitted signal would not be transparent if there are
a long series of 0's which would cause an error in the timing information.
Power Spectral Density

• PSD: R(k) is the auto-correlation function, g(t) is the shape of the impulse

N
s (t ) = å a g ( t - nT )
n =- N
n b

G( f )
2

Ss ( f ) =
Tb
å R (k ) e
k =-¥
j 2p f kTb
• UNIPOLAR NRZ
• UNIPOLAR RZ
• POLAR NRZ

Espectro de potência [dB] fb

-3 f b -2 f b - fb 0 fb 2 fb 3 fb
• POLAR RZ
• BIPOLAR NRZ

Espectro de potência [dB] fb

-3 fb -2 fb - fb 0 fb 2 fb 3 fb
• BIPOLAR RZ (“false” 0 in "# ; 1st sinc null in 2" ) #

fb
Espectro de potência [dB]

-3 f b -2 f b - fb 0 fb 2 fb 3 fb
• MANCHESTER

2 fb
Espectro de potência [dB]

-6 f b -4 f b -2 f b 0 2 fb 4 fb 6 fb
41

5.11 Summary and Discussion

• Sampling : which operates in the time domain ;


• The sampling process is the link between an analog waveform and its discrete-time representation
• quantization : which operates in the amplitude domain;
• The quantization process is the link between an analog waveform and its discrete-amplitude
representation
• Sampling theorem
• A strictly band-limited signal with no frequency components higher than W Hz is represented uniquely by a
2W samples per second.
• The sampling process is basic to the operation of all pulse modulation systems
• Analog pulse modulation results from varying some parameter of the transmitted pulses
• Digital pulse modulation systems transmit analog message signals as a sequence of coded pulses
• Source-encoding strategy (PCM)
• Whose purpose is to convert analog signals into digital form

You might also like