通信原理(英文版)4PPT课件
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通信原理(英文版)
Chapter 2 Signals
2.1 Classification of Signals
2.1.1 Deterministic signals and random signals
• What is deterministic signal? • What is random signal?
2.1.2 Energy signals and power signals
f (t) f (t T) t
Its frequency spectrum is
/2
C( jn0 )
1 T
/ 2 Ve j n0t dt
/ 2
1 T
V
jn 0
e
j n0 t
/ 2
V e j n0 / 2 e j n0 / 2
f (t) sin(t) Its frequency spefct(rtu)m: f (t 1)
0 t 1 t
C(
jn 0 )
1 T0
T0 / 2 s(t )e jn0t dt
T0 / 2
1 sin(t )e j 2nt dt
Solution: Let the expression of the rectangular pulse be
Then its frequency spectral density is
its
Fourier
tragns(fto)rm:
1
0
t /2 t /2
G() / 2 e jt dt 1 (e j / 2 e j / 2 ) sin( / 2)
2.1 Classification of Signals
2.1.1 Deterministic signals and random signals
• What is deterministic signal? • What is random signal?
2.1.2 Energy signals and power signals
f (t) f (t T) t
Its frequency spectrum is
/2
C( jn0 )
1 T
/ 2 Ve j n0t dt
/ 2
1 T
V
jn 0
e
j n0 t
/ 2
V e j n0 / 2 e j n0 / 2
f (t) sin(t) Its frequency spefct(rtu)m: f (t 1)
0 t 1 t
C(
jn 0 )
1 T0
T0 / 2 s(t )e jn0t dt
T0 / 2
1 sin(t )e j 2nt dt
Solution: Let the expression of the rectangular pulse be
Then its frequency spectral density is
its
Fourier
tragns(fto)rm:
1
0
t /2 t /2
G() / 2 e jt dt 1 (e j / 2 e j / 2 ) sin( / 2)
《通信原理》第04章模拟信号的数字化精品PPT课件
ห้องสมุดไป่ตู้
t
…
t
…
t
S(f)
( f ) Sk ( f ) Sˆ( f )
f
…
f
…
f
t
f
7
4.2.1 低通模拟信号的抽样
频谱混叠
S(f)
spectrum aliasing
f ( f )
f
Sk ( f )
…
…
f
8
4.2.1 低通模拟信号的抽样
ideal lowpass filter
抽样信号恢复低通滤波器
s(t)
s(t)
t
t
δT (t)
c (t)
t
t
sk(t)
sk(t)
t
t
3
4.2.1 低通模拟信号的抽样
band-limited signal
低通抽样定理 一个带宽有限信号 s (t) 的最高频率为 fH ，若
抽样频率 fs ≥ 2 fH ，则可以由抽样信号序列 sk (t) 无 失真地恢复原始信号 s (t) 。 说明
抽样频率与信号频率的关系曲线
fs 4B
3B
2B
B
O
B 2B 3B 4B 5B 6B
fL
15
4.2.2 带通模拟信号的抽样
带通抽样的频谱
fH = 4 kHz fL = 3 kHz B = 1 kHz
fs = 2 kHz
S(f)
−4B
0
4B
Sk( f )
bandpass sampling
f
−4fs −3fs −2fs −fs O fs 2fs 3fs 4fs
领域也有广泛应用
pulse amplitude modulation (PAM)
t
…
t
…
t
S(f)
( f ) Sk ( f ) Sˆ( f )
f
…
f
…
f
t
f
7
4.2.1 低通模拟信号的抽样
频谱混叠
S(f)
spectrum aliasing
f ( f )
f
Sk ( f )
…
…
f
8
4.2.1 低通模拟信号的抽样
ideal lowpass filter
抽样信号恢复低通滤波器
s(t)
s(t)
t
t
δT (t)
c (t)
t
t
sk(t)
sk(t)
t
t
3
4.2.1 低通模拟信号的抽样
band-limited signal
低通抽样定理 一个带宽有限信号 s (t) 的最高频率为 fH ，若
抽样频率 fs ≥ 2 fH ，则可以由抽样信号序列 sk (t) 无 失真地恢复原始信号 s (t) 。 说明
抽样频率与信号频率的关系曲线
fs 4B
3B
2B
B
O
B 2B 3B 4B 5B 6B
fL
15
4.2.2 带通模拟信号的抽样
带通抽样的频谱
fH = 4 kHz fL = 3 kHz B = 1 kHz
fs = 2 kHz
S(f)
−4B
0
4B
Sk( f )
bandpass sampling
f
−4fs −3fs −2fs −fs O fs 2fs 3fs 4fs
领域也有广泛应用
pulse amplitude modulation (PAM)
通信原理——Chapter4
n 1
}
jc t
Re{2 cn e
n 1Байду номын сангаас
j ( n0 c ) t
e
jc t
} Re{g (t )e
}
where g (t ) 2 cn e
n 1 j ( n0 c ) t
(4 8)
Because v(t ) is a bandpass waveform with nonzero
The complex envelope is a function of the modulating
4-2
g (t ) g m(t ) Thus, g performs a mapping operation on m(t ). This was shown in Fig. 4 1. Table 4 1 gives the "big picture" of the modulation problem. In the receiver, the inverse function m[ g ] is required.
Re{c1}Re{c2 } a1a2 1 [Re{ c c } Re{ c 1 2 1 c2 }] 2 1 2
4-3
jc t jc ( t ) Rv ( ) 1 [Re{ g ( t ) e g ( t ) e } 2 jc t
Re{g (t )e
4-1
Definition. Modulation is the process of imparting the source information onto a bandpass signal with a carrier frequency f c by the introduction of amplitude or phase perturbations or both. This bandpass signal is called the modulated signal s (t ), and the baseband signal is called the modulating signal m(t ).
}
jc t
Re{2 cn e
n 1Байду номын сангаас
j ( n0 c ) t
e
jc t
} Re{g (t )e
}
where g (t ) 2 cn e
n 1 j ( n0 c ) t
(4 8)
Because v(t ) is a bandpass waveform with nonzero
The complex envelope is a function of the modulating
4-2
g (t ) g m(t ) Thus, g performs a mapping operation on m(t ). This was shown in Fig. 4 1. Table 4 1 gives the "big picture" of the modulation problem. In the receiver, the inverse function m[ g ] is required.
Re{c1}Re{c2 } a1a2 1 [Re{ c c } Re{ c 1 2 1 c2 }] 2 1 2
4-3
jc t jc ( t ) Rv ( ) 1 [Re{ g ( t ) e g ( t ) e } 2 jc t
Re{g (t )e
4-1
Definition. Modulation is the process of imparting the source information onto a bandpass signal with a carrier frequency f c by the introduction of amplitude or phase perturbations or both. This bandpass signal is called the modulated signal s (t ), and the baseband signal is called the modulating signal m(t ).
通信原理(英文版)课件
36
l 4-ary coding channel model
0
0
1
Transmitting end
2
1
Receiving end
2
3
3
Figure 1.4.12 4-ary coding channel model
37
1.4.4 Influence of channel characteristics on signal transmission
2
1.2 Message, information & signal
lMessage：speech, letters, figures, images…
lInformation：effective content of message. Different types of messages may contain the same information
# Information content I = I [ P(x) ]，P(x) – Occurrence probability
# Definition：I = loga [1/P(x)] = -logaP(x) # Usually, set a = 2, the unit of the information content will be called a bit.
0
ω
0
Ideal characteristic
Ideal characteristic: phase --- () = k ;
group delay --- () = d()/d = k
Influence of distortion: waveform distortion, inter-symbol interference
现代通信原理课件(英文版)(ppt 35页)
are defined on continuum. 4. Digital communication system transfers information
from a digital source to the intended receiver(sink) 5. Analog communication system transfers
2) Note: The general principles of digital and analog modulation apply to all types of channels, although channel characteristics may impose constraints that favor a particular type of signaling
15
1.2 Digital and Analog source and system
2 the advantage of digital system
1）Relatively inexpensive digital circuits may be used
2） Privacy is preserved by using data encryption
8
1.2 Digital and Analog source and system
• The generation of communication system
Information input m(t)
Signal processing
Carrier circuits
Transmitter
channel noise
1. Selection of the information-bearing
from a digital source to the intended receiver(sink) 5. Analog communication system transfers
2) Note: The general principles of digital and analog modulation apply to all types of channels, although channel characteristics may impose constraints that favor a particular type of signaling
15
1.2 Digital and Analog source and system
2 the advantage of digital system
1）Relatively inexpensive digital circuits may be used
2） Privacy is preserved by using data encryption
8
1.2 Digital and Analog source and system
• The generation of communication system
Information input m(t)
Signal processing
Carrier circuits
Transmitter
channel noise
1. Selection of the information-bearing
无线电通信原理 第四章(英文)
2020/7/18
10
the time varying discrete-time impulse response model for a multipath radio channel
2020/7/18
11
N 1
hb (t,t ) ai (t,t ) exp[ ji (t,t )] (t ti (t)) i0
Local area: no greater than 6m outdoor Local area: no greater than 2m indoor
2020/7/18
14
Parameters of Mobile Multipath Channels
• Time Dispersion Parameters • Coherent bandwidth • Doppler Spread • Coherence Time
•We need a few major parameters for easy –Compare different channels (delay, bandwidth, spectrum, etc) –Develop design guide lines for wireless signals
-If the vehicle is moving directly towards the transmitter
f 26.82 165Hz 0.162
-If the vehicle is moving perpendicular to the angle of arrival of
the transmitted signal
t : time variation due to motion/Doppler shift
通信原理(英文版)
6
【Example 2.4】Find the waveform and the frequency spectral density of a sample function. Solution: The definition of the sample function is
sin t Sa ( t ) t
d(t)
1
(f)
0
t
0
f
meaning of d function: It is a pulse with infinite height, infinitesimal width, and unit area. Sa(t) has the following property:
Physical
F ( ) lim
/2 / 2
cos 0 te
jt
sin[( 0 ) / 2] sin[( 0 ) / 2] dt lim 2 ( ) / 2 ( ) / 2 0 0
The frequency spectral density of d(t):
( f ) d (t )e
jt
d (t ) 0
t 0
dt 1 d (t )dt 1
7
d(t)
and its frequency spectral density:
f (t ) f (t 1) t
1
Its frequency spectrum:
1 C ( jn 0 ) T0
T0 / 2
T0 / 2
s(t )e
【Example 2.4】Find the waveform and the frequency spectral density of a sample function. Solution: The definition of the sample function is
sin t Sa ( t ) t
d(t)
1
(f)
0
t
0
f
meaning of d function: It is a pulse with infinite height, infinitesimal width, and unit area. Sa(t) has the following property:
Physical
F ( ) lim
/2 / 2
cos 0 te
jt
sin[( 0 ) / 2] sin[( 0 ) / 2] dt lim 2 ( ) / 2 ( ) / 2 0 0
The frequency spectral density of d(t):
( f ) d (t )e
jt
d (t ) 0
t 0
dt 1 d (t )dt 1
7
d(t)
and its frequency spectral density:
f (t ) f (t 1) t
1
Its frequency spectrum:
1 C ( jn 0 ) T0
T0 / 2
T0 / 2
s(t )e
通信原理 张水英版课件
4
通信发展概况
2. 近代：
1837年：莫尔斯发明电报系统。 1876年：贝尔发明电话。
5
通信发展概况
3.现代 20世纪60年代以后：
数字通信技术进入高速发展阶段。
近20多年：
数字通信迅猛发展； 光纤通信也携手同行。 两者都成为现代通行网的主要支柱。
6
通信发展概况
塞缪尔·莫尔斯 （Samuel
Finley Breese Morse
44
【例如】
有一个4PSK数字通信系统，现传输 20个码元，其中错了2个码元，那 么误码率为10%；如果每个错误码 元有一个比特错误，那么误比特率 为5%；如果每个错误码元有两个比 特错误，那么误比特率为10%；
45
【例如】
已知某四进制数字传输系统的信息传 输速率为2400（bit/s），接收端在 半小时内共收到216个错误码元，试 计算该系统的误码率。
缺点：
占用频带宽 对同步要求高、系统和设备比较复杂
17
1.3 通信系统的分类及通信方式
一、通信系统的分类
按消息的物理特征 按调制方式 按信号特征 按传输媒介 按复用方式
18
1、按消息的物理特征
电报通信系统 电话通信系统 数据通信系统 图像通信系统
。 。 。 。
19
2、按调制方式分类
基带传输：将未经调制的信号直接传送，
第1章
绪论
内容
1.1 通信的基本概念 1.2 通信系统的模型 1.3 通信系统分类及通信方式 1.4 信息的度量 1.5 通信系统的主要性能指标
2
1.1 通信的基本概念
消息、信息和信号 通信 通信发展概况
3
通信发展概况
1. 古代：
通信发展概况
2. 近代：
1837年：莫尔斯发明电报系统。 1876年：贝尔发明电话。
5
通信发展概况
3.现代 20世纪60年代以后：
数字通信技术进入高速发展阶段。
近20多年：
数字通信迅猛发展； 光纤通信也携手同行。 两者都成为现代通行网的主要支柱。
6
通信发展概况
塞缪尔·莫尔斯 （Samuel
Finley Breese Morse
44
【例如】
有一个4PSK数字通信系统，现传输 20个码元，其中错了2个码元，那 么误码率为10%；如果每个错误码 元有一个比特错误，那么误比特率 为5%；如果每个错误码元有两个比 特错误，那么误比特率为10%；
45
【例如】
已知某四进制数字传输系统的信息传 输速率为2400（bit/s），接收端在 半小时内共收到216个错误码元，试 计算该系统的误码率。
缺点：
占用频带宽 对同步要求高、系统和设备比较复杂
17
1.3 通信系统的分类及通信方式
一、通信系统的分类
按消息的物理特征 按调制方式 按信号特征 按传输媒介 按复用方式
18
1、按消息的物理特征
电报通信系统 电话通信系统 数据通信系统 图像通信系统
。 。 。 。
19
2、按调制方式分类
基带传输：将未经调制的信号直接传送，
第1章
绪论
内容
1.1 通信的基本概念 1.2 通信系统的模型 1.3 通信系统分类及通信方式 1.4 信息的度量 1.5 通信系统的主要性能指标
2
1.1 通信的基本概念
消息、信息和信号 通信 通信发展概况
3
通信发展概况
1. 古代：
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with cut-off frequency fH .
➢ Time domain: When the ideal low-pass filter is excited by the sampled pulse sequence, the output of the filter is the sum of the impulse responses, as shown in the figure. The sum of these impulse responses composes the original signal.
original signal can be restored from the sampled signal.
4
Here, the condition of restoration of the origial signal is
fs 2 fH
2fH is called Nyquist sampling rate. The corresponding smallest sampling time interval is called Nyquist samplhe highest frequency of signal s(t) has been assumed less than fH, therefore if frequency interval fs 2fH, then each displaced spectrum S(f) of the original signal contained in Sk(f)
Chapter 4 Digitiation of analog signal
4.1 Introduction
Two categories of information sources: analog signal, digital signal
Three steps of A/D conversion: sampling, quantization , coding
The above equation shows that spectrum Sk(f) of the sampled signal is the superposition of infinite spectra S(f) with frequency interval fs, since S(f - nfs) is the result of displacement nfs of the signal frequency spectrum S(f) on the
sampling process ＝ periodical unit impulse analog signal Practically,
sampling process ＝ periodical narrow pulse analog signal Sampling theorem: If the highest frequency of a continuous
Analog signal
s(t)
2
Proof of sampling theorem Let: s(t) － signal with highest frequency less than fH
T(t) － periodical unit impulse with repetition period T and repetition frequency fs = 1/T, then the sampled signal is:
analog signal s(t) is less than fH , and if it is sampled by
periodic impulses with interval time T 1/2fH , then s(t) can be completely decided by these samples.
sk(t)s(t)T(t) s(k)T
Let the Fourier transform of sk(t) is Sk(f), then Sk(f)S(f) (f)
where,
Sk(f) － spectrum of sk(t) S(f) － spectrum of s(t) ( f ) － spectrum of T(t)
(f ) is the frequency
spectrum of periodical unit impulse, it can be found as
(f)T1n (f nsf)
3
Substituting (f)T1n(f nfs) into Sk(f)S(f) (f)
obtain S k(f)T 1 S (f)n (f n fs) T 1 S (f n fs)
is not superposed with each other, as shown in the figure.
Thus frequency spectrum S(f) of signal s(t) can be separated from Sk(f), and s(t) can be easily obtained from S(f), i.e. the
Most popular A/D conversion method: Pulse Code Modulation (PCM)
1
4.2 Sampling of analog signal
4.2.1 Sampling of low-pass analog signal
Usually sampling at equal time interval T Theoretically,
5
Method of restoration of original signal from sampled signal:
➢ Frequency domain: When fs 2fH , the original signal can be
separated from the sampled signal by an ideal low-pass filer
➢ Time domain: When the ideal low-pass filter is excited by the sampled pulse sequence, the output of the filter is the sum of the impulse responses, as shown in the figure. The sum of these impulse responses composes the original signal.
original signal can be restored from the sampled signal.
4
Here, the condition of restoration of the origial signal is
fs 2 fH
2fH is called Nyquist sampling rate. The corresponding smallest sampling time interval is called Nyquist samplhe highest frequency of signal s(t) has been assumed less than fH, therefore if frequency interval fs 2fH, then each displaced spectrum S(f) of the original signal contained in Sk(f)
Chapter 4 Digitiation of analog signal
4.1 Introduction
Two categories of information sources: analog signal, digital signal
Three steps of A/D conversion: sampling, quantization , coding
The above equation shows that spectrum Sk(f) of the sampled signal is the superposition of infinite spectra S(f) with frequency interval fs, since S(f - nfs) is the result of displacement nfs of the signal frequency spectrum S(f) on the
sampling process ＝ periodical unit impulse analog signal Practically,
sampling process ＝ periodical narrow pulse analog signal Sampling theorem: If the highest frequency of a continuous
Analog signal
s(t)
2
Proof of sampling theorem Let: s(t) － signal with highest frequency less than fH
T(t) － periodical unit impulse with repetition period T and repetition frequency fs = 1/T, then the sampled signal is:
analog signal s(t) is less than fH , and if it is sampled by
periodic impulses with interval time T 1/2fH , then s(t) can be completely decided by these samples.
sk(t)s(t)T(t) s(k)T
Let the Fourier transform of sk(t) is Sk(f), then Sk(f)S(f) (f)
where,
Sk(f) － spectrum of sk(t) S(f) － spectrum of s(t) ( f ) － spectrum of T(t)
(f ) is the frequency
spectrum of periodical unit impulse, it can be found as
(f)T1n (f nsf)
3
Substituting (f)T1n(f nfs) into Sk(f)S(f) (f)
obtain S k(f)T 1 S (f)n (f n fs) T 1 S (f n fs)
is not superposed with each other, as shown in the figure.
Thus frequency spectrum S(f) of signal s(t) can be separated from Sk(f), and s(t) can be easily obtained from S(f), i.e. the
Most popular A/D conversion method: Pulse Code Modulation (PCM)
1
4.2 Sampling of analog signal
4.2.1 Sampling of low-pass analog signal
Usually sampling at equal time interval T Theoretically,
5
Method of restoration of original signal from sampled signal:
➢ Frequency domain: When fs 2fH , the original signal can be
separated from the sampled signal by an ideal low-pass filer