communication theory
communication theory
COEC 324
Chapter 1
Introduction to digital communication
1 Analog-to-Digital Conversion
a. sampling theorem
b. Ideal sampling
c. Practical Sampling
d. Quantization
e. Pulse Code Modulation
f. Differential Pulse code Modulation (DPCM)
g. Adaptive Differential PCM (ADPCM)
h. Delta Modulation
i. PCM Waveform Types
3 Digital Modulation
a. Amplitude Shift Keying (ASK)
b. Phase Shift Keying (PSK)
c. Quadrature Amplitude Modulation (QAM)
d. Binary Frequency-Shift Keying (BFSK)
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Introduction to Digital Communication
• Communications is the Process of Transmitting Information from a Source to a Destination
Source Destination
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Introduction to Digital Communication
➢ Why Digital Communication?
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Introduction to Digital Communication
Input signal output signal Message
Message
Transducer Transducer
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Baseband vs. Passband Communication Systems
▪ Communication systems can be classified into two groups depending on the range of frequencies they use to
transmit information. These communication systems are classified into BASEBAND or PASSBAND system.
Baseband transmission sends the information signal as it is without modulation (without frequency shifting) while
passband transmission shifts the signal to be transmitted in frequency to a higher frequency and then transmits
it, where at the receiver the signal is shifted back to its original frequency.
▪ Almost all sources of information generate baseband signals. Baseband signals are those that have frequencies
relatively close to zero such as the human voice (20 Hz – 5 kHz) and the video signal from a TV camera (0 Hz –
5.5 MHz). A plot of an audio signal and its frequency spectrum are shown below, where it is seen that the most of
the power of the audio signal is concentrated in the frequency range from (0 – 4 kHz)
Digital Modulation
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Sampling Theory
Analog-to-Digital Conversion
➢ The digital signal is discrete in time (Sampling) and limited number of values (Quantization)
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Analog-to-Digital Conversion
Sampling Theory
the band-limited 𝑓0 Hz lowpass analog signal
𝐺 𝑓 = 0, 𝑖𝑛 𝑐𝑎𝑠𝑒 𝑜𝑓 𝑓 > 𝑓0
that has been sampling can be reconstructed exactly (without any error) from infinite
sequence of samples. If the sampling rate (at least) 2𝑓0 samples per seconds
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Sampling Theory
𝑥(𝑡)
𝑥(𝑡) 𝑥𝑠 (𝑡)
1
𝑓𝑠 =
𝑠(𝑡) 𝑇𝑠
Sampling frequency 𝑓𝑠
𝑇𝑠
𝑥𝑠 𝑡
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Sampling Theory
𝑥(𝑡) 𝑥𝑠 𝑡 = 𝑥 𝑡 𝑠(𝑡)
∞
𝑠(𝑡)
𝑇𝑠
𝑠(𝑡) = 𝛿(𝑡 − 𝑛𝑇𝑠 )
𝑛=−∞
𝑥𝑠 𝑡
We want to get the Fourier transform of 𝑥𝑠 𝑡
𝑋𝑠 𝑓 = 𝑋 𝑓 ∗ 𝑆(𝑓)
this method more complex
18
Sampling Theory
𝑥𝑠 𝑡 = 𝑥 𝑡 𝑠(𝑡)
𝑥(𝑡)
∞
𝑠(𝑡) 𝑠(𝑡) = 𝛿(𝑡 − 𝑛𝑇𝑠 )
𝑇𝑠 𝑛=−∞
1
𝐷𝑛 = න 𝑠(𝑡)𝑒 −𝑗𝑛𝜔𝑠 𝑡 𝑑𝑡
𝑇𝑠
𝑇𝑠
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Sampling Theory
𝑥𝑠 𝑡 = 𝑥 𝑡 𝑠(𝑡)
𝑥(𝑡) ∞
𝑥𝑠 𝑡 𝑇𝑠 𝑇𝑠
2 2
1 −𝑗𝑛𝜔 𝑡
1
𝐷𝑛 = න𝑠 𝑡 𝑒 𝑠 𝑑𝑡 = න 𝛿(𝑡)𝑒 −𝑗𝑛𝜔𝑠 𝑡 𝑑𝑡
𝑇𝑠 𝑇𝑠
𝑇 𝑇
− 2𝑠 − 2𝑠
𝛿 𝑡 𝑥 𝑡 =𝛿 0 𝑥 𝑡
1
න𝛿 𝑡 𝑥 𝑡 = 𝑥 0 𝐷𝑛 = = 𝑓𝑠
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𝑇𝑠
Sampling Theory
𝑥𝑠 𝑡 = 𝑥 𝑡 𝑠(𝑡)
𝑥(𝑡) ∞
𝑥𝑠 𝑡
∞ ∞ ∞
∞ ∞
𝑠 𝑡 = 𝑓𝑠 𝑒 𝑗𝑛𝜔𝑠𝑡
𝑛=−∞
∞ ∞ ∞
𝑥𝑠 𝑡 ∞
𝑋𝑠 𝑓 = 𝑓𝑠 𝑋(𝑓 − 𝑛𝑓𝑠 )
𝑛=−∞
1
𝑇𝑠
23 −𝐵 𝐵 𝑓𝑠 − 𝐵 𝑓𝑠 𝑓𝑠 + 𝐵
Sampling Theory
𝑥(𝑡)
1
𝑠(𝑡) 𝑇𝑠
𝑇𝑠
𝑥𝑠 𝑡 −𝐵 𝐵 𝑓𝑠 − 𝐵 𝑓𝑠 𝑓𝑠 + 𝐵
𝐵 ≤ 𝑓𝑠 − B
2𝐵 ≤ 𝑓𝑠
Sampling Condition 𝑓𝑠 ≥ 2𝐵
24
Sampling Theory
Sampling Condition
If 𝑓𝑠 < 2𝐵
If 𝑓𝑠 = 2𝐵
Nyquist rate aliasing
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Signal Reconstruction
𝑥(𝑡) 𝑓𝑠 ≥ 2𝐵
𝑠(𝑡)
𝑇𝑠
−𝐵 𝐵
𝑥𝑠 𝑡 𝑓𝑠 = 2𝐵
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Signal Reconstruction
𝑋𝑠 𝑓 𝐻 𝑓 𝑋𝑟 𝑓
𝑥𝑠 𝑡 ℎ(𝑡) 𝑥𝑟 𝑡
𝑇𝑠 𝑓 <𝐵
𝐻 𝑓 =ቐ
0 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
𝑓𝑖𝑛𝑑 𝑋𝑟 𝑓 ?
𝐻 𝑓 ∶ 𝑡ℎ𝑒 𝑡𝑟𝑎𝑛𝑠𝑓𝑒𝑟 𝑓𝑢𝑛𝑐𝑡𝑖𝑜𝑛 𝑓𝑜𝑟 𝑙𝑜𝑤 𝑝𝑎𝑠𝑠 𝑓𝑖𝑙𝑡𝑒𝑟
ℎ 𝑡 : 𝑡ℎ𝑒 𝑖𝑚𝑝𝑢𝑙𝑠𝑒 𝑟𝑒𝑠𝑝𝑜𝑛𝑐𝑒 𝑓𝑖𝑛𝑑 𝑥𝑟 𝑡 ?
27 𝑓𝑖𝑛𝑑 ℎ 𝑡 ?
Signal Reconstruction
𝑋𝑠 𝑓 𝐻 𝑓 𝑋𝑟 𝑓
𝑥𝑠 𝑡 ℎ(𝑡) 𝑥𝑟 𝑡 𝑋𝑟 𝑓 = 𝑋𝑠 𝑓 . 𝐻 𝑓
∞
𝑗2𝜋𝑓𝑡
𝑥𝑟 𝑡 = 𝑥𝑠 𝑡 ∗ ℎ(𝑡)
ℎ 𝑡 = න𝐻 𝑓 𝑒 𝑑𝑓
−∞
𝑓𝑔
ℎ 𝑡 = න 𝑇𝑠 𝑒 𝑗2𝜋𝑓𝑡 𝑑𝑓
−𝑓𝑔
28
Signal Reconstruction
ℎ 𝑡 =0
ℎ 𝑡 = 2𝑇𝑠 𝐵 𝑠𝑖𝑛𝑐 ( 2𝜋𝐵𝑡)
𝑠𝑖𝑛𝑐 (2𝜋𝐵𝑡) = 0
𝐴𝑠𝑠𝑢𝑚𝑖𝑛𝑔 𝑛𝑦𝑞𝑢𝑖𝑠𝑡 𝑠𝑚𝑎𝑝𝑙𝑖𝑛𝑔 𝑟𝑎𝑡𝑒, 2𝑇𝑠 𝐵 = 1 2𝜋𝐵𝑡 = ∓ 𝑛𝜋
2𝐵𝑡 = ∓ 𝑛
𝑛
𝑡= ∓
2𝐵
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ℎ 𝑡 = 𝑠𝑖𝑛𝑐 ( 2𝜋𝐵𝑡)
Signal Reconstruction
∞
𝑥𝑟 𝑡 = 𝑥𝑠 𝑡 ∗ ℎ(𝑡)
∞
𝑥𝑟 𝑡 = 𝑥 𝑛𝑇𝑠 ℎ 𝑡 − 𝑛𝑇𝑠
𝑛=−∞
∞
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Signal Reconstruction
∞
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Practical Sampling
Practical Signal Sampling
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