Communication and network technology

Learning aim
• To study the transmission process and transmission principle
of voice signal in digital baseband and digital broadband
transmission system, the concept of digital communication
system is established.

• Be able to test the performance of digital baseband

transmission system and digital broadband transmission
system, and be able to open and maintain specific
communication equipment .
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 Learning aim
• Be able to draw the digital band transmission system and the digital
baseband transmission system communication model, and correctly
describe the functions of each component, and analyze and test the
transmission systems.

• According to the operational process of training project to finish a

summary report about the project.

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 Digital baseband communication system

• 2.1 digital baseband communication system

2.1.1 Pulse code modulation

2.1.2 The principle of time division multiplexing
2.1.3 Types of digital baseband signal

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 Learning aim
You should be able to

• 1. to understand components of digital baseband systems and

functions of each components.

• 2. to master purpose and coding procedure of PCM

• 3. to master sampling theorem

• 4. to understand features of quantization, uniform quantization

and non-uniform quantization.
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2.1 Digital baseband transmission system

modes of voice transmission the digital communication system :

digital base-band transmission and digital broadband transmission.

Digital baseband signals are original data signals from data

terminals, such as binary sequences produced by the computer,
code output by the teletype machine, or PCM code etc.

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2.1 Digital baseband transmission system

digital baseband transmission In some cable channels with low-

pass characteristics, the digital baseband signals can be transmitted
directly, especially when the transmission distance is not too long.

Digital broadband transmission Most channels are band-pass type

such as various wireless channels and optical channels, digital
baseband signals must be modulated to move the spectrum to
certain high carrier for be transmitted in the channel.

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 Phone Signal Signal
Interface 1 multiplexing demultiplexing

Baseband
transmission channel

line line
voice encoding decoding voice Phone
codec codec Interface 2
Fig 2-1 digital baseband communication system block diagram

As shown Fig 2-1 , a digital baseband voice communication system

consists of telephone interface module, voice codec module, frame
multiplexing/demultiplexing module and line coding module.

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Explain terms

• User circuit also called a subscriber line interface circuit , is used

to convert other signals into electrical signals.

• Voice encoder uses PCM to convert analog signals to digital

signals.

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Explain terms

• Signal multiplex circuit adopts time division multiplexing to

combine the multiplex signals into one signal.

• Line coding circuit uses HDB3 or CMI encoding mode to convert

the binary sequence digital signals into HDB3 or CMI electrical
signals which suitable for cable channel transmission.

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2.1.1 脉冲编码调制 PCM

Analog to digital conversion （A/D） takes three steps: sampling,

quantization, and coding.

• Sampling: discretize time

• Quantization : discretize amplitude
• Coding: convert to binary code

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 2.1.1 脉冲编码调制 PCM

Analog  T t  Digital
information Communication Low-pass
source system filter

模拟信 抽 量 编 数字通 译 低
息源 样 化 码 信系统 码 通
x̂t 
sampling quantization encoding decoding
x(t) xs t  xq t  s 
k ŝ 
k
xˆ q t 
模拟随机信号
Analog
random signal A/D模数转换 D/A数模转换
A/D conversion
图6-1 模拟信号的数字传输 D/A conversion

Fig 2-2 PCM procedure

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 Electrical level

time

Actual sampling value

Quantized sampling value
Quantization series
PCM code group

2020/2/27 Fig 2-3 waveform of PCM signal-channel sampling-quantization, and coding 13

1. sampling theorem

• A time continuous signal x(t) is limited in a range of frequency(0,

fH) , is same interval (fS≥2fH )sampled and the sampling interval is
not greater then (1/2•fH), and then the signal will be completely
determined by the sampling value.

• f H is the highest frequency of signal x(t).

f s is the sampling frequency.
restoration condition of ensuring the signal can be undistorted :

fs  2 fH
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Sampling theorem

Analog signal x(t)

(low-pass type)

Sampling signal xs(t)

Ts

fs（=1/Ts） fs  2 fH
Analog signal by receiver
terminal rebuilt x′(t)

图2-7 抽样定理示意图
2020/2/27 Fig 2-7 sampling theorem diagram 15
Spectrum waveform when the signal is sampled by transmit terminal

Fig 2-4 spectrum charts when signal be sampled (send port)

Problem 1

fs  2 fH

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2.脉冲振幅调制(PAM)
• pulse amplitude modulation

xt 
基带信号
Baseband signal

0 t
PAM

PAM signal

0 t
PDM

Fig 2-5 PAM diagram

2020/2/27 0 t18
 x t  x s t 
(a)
s t  X  
xt 

t S  
s t 
Ts Principle of PAM :
PAM is a modulation
 t mode  2,  that
4 H
the
X   4
x s t 
S

amplitude of pulse
t carrier changes
 5 withthe
H H

baseband signal.
Fig 2-6 principle of PAM diagram
3. Quantization of analog signals

• Quantization ： The sampled values of analog signals are

represented by a finite number of levels. The amplitude of the
signal is quantified hierarchically.

• Quantization error: The error of the quantized signal and the

original signal.

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 Actual value of signal

quantization error
Quantized value of signal

Fig 2-7 quantization error

uniform quantization and hierarchical quantization

• Uniform quantization ：A quantization process that divides

the value range of the original signal at equal amplitude , that
is, the interval between adjacent layers is equal.

• Hierarchical quantization ： The range of signal sampled is

divided into several layers (quantization level), each sample using
methods of half adjust or rounding make each sample into the
closest quantized value.

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Basic terms

• Quantization range: the range of the signal sampling value, usually defined
by the difference between the maximum and minimum of the signal
amplitude.

• Quantized value : the values obtained by sampling and quantization, which

is limited to the discrete level value.

• Quantization level: the number of quantized values ("layers") that can be

represented by M.

• Quantization interval (scales): difference between adjacent quantization

levels, represented by ∆ .

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Problem 2

• The range of signal is 0～7V，if ∆=1V，8 quantization levels are

generated (M=8), they are 。According to half-
adjust method,<0.5V is quantified as 0V，0.5V-1.5V is quantified
as 1V，……，>6.5V quantified as 7V. In fig 2-7,sampling values
such as 2.2V 、 4.0V 、 5.0V 、 2.8V and 1.8V are separately
converted to quantized values, they are 、 、 、
and .

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The lack of uniform quantization

• Because ∆ is a constant, so the aforementioned quantization method

called "uniform quantization", then the encoding based on this
quantization is called linear coding.

• In the reality, the method is not suitable for small signal. Because
of the uniform quantization scale is fixed, it has nothing to do with
the input sample values , whether it is large or small.

• For large signals, the SNR is more than enough; while for small
signals, the SNR is inadequate, uniform quantization is detrimental
to small signals.
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The lack of uniform quantization

• In order to improve the SNR of small signals, ∆ can be

subdivided , certainly the SNR of large signals can also
improve, but this method lead to the digital rate increase, then
more broadband channel be needed for transmission, it also
reduce effectiveness.

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Non-uniform quantization

• ∆ change its size varies with the size of the signal, it is not a
constant. When input signal is small signal then ∆ is small ,
and input signal become large , it turn large . So almost the
same SNR is obtained and the total quantization levels if less
then of uniform quantization.

• This non-uniform quantization level arrangement is called

non-uniform quantization or nonlinear quantization, the
method shorten the length of code word and improve the
efficiency of communication.

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 Signal compressor

• The signal compressor is a nonlinear transform circuit , it amplify

the weak signal and compress the strong signal.

• Before uniform quantization is carried out, the signal is processed

one time to compress the large signal and amplify the small
signal.

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Signal compressor
non-uniform quantization
channel

uniform coding decoding

compress quantization expand
input
output

expand
compress

input output
The large
The small signal
signal
2020/2/27 Fig 2-8 non-uniform quantization diagram 29
A compression law 13 polygonal lines
• The signal is normalized at first.

• The X axis(input) and the y axis(output) are divided in two different ways.

• The X axis is unevenly divided into 8 segments within the 0~1 (normalized)
range. The rule: the next segment is 1/2 of the last segment.

NO. 1 2 3 4 5 6 7 8
range 0- 1/128- 1/64- 1/32- 1/16- 1/8- 1/4- 1/2-1
1/128 1/64 1/32 1/16 1/8 1/4 1/2
slope 16 16 8 4 2 1 1/2 1/4

• The Y axis is evenly divided into 8 segments within the 0~1 (normalized) range,
any segments is 1/8.

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 slope
1-16
2-16
3-8
4-4
5-2
6-1
7-1/2
8-1/4

2020/2/27 Fig 2-9 A compression laws 13 polygonal lines 31

A compression law 13 polygonal lines

• As shown in Fig 2-9, first of the 2 sections have the same slopes (16), so they can be considered as
a straight line segment, so there are only 7 different polygonal lines with different slopes.

• Because the voice signal is bipolar signal, so there are a group of polygonal lines in the negative
direction, there are 7, which is close to the zero point of 1,2 segments slope are equal to 16, and
in the positive direction 1,2 segments have the same slope, so the four segments can be combined
into one, therefore, total 2 x (8-1) -1=13, so it was called 13 polygonal.

• In quantitative analysis the rule still is executed that there are 8 segments respectively in positive
and negative direction.

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Brief summary

• 1. Procedure of analog signal digitalize

• 2. concept of sampling theorem and sampling frequency

• 3. uniform quantization and non-uniform quantization

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 HOMEWORK

• 1、Write three steps of A/D conversion.

• 2、If sampling frequency fs =4000Hz，the band of voice

signal from 0 to 5000 Hz ， PAM communication can be
completed? Why? How to solve?

• 3、explain these terms: quantization , uniform quantization ,

non-uniform quantization

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