Digital communication
Lecture two
Objective of Lecture:
• Sampling theorem
1.8 Sampling theorem:
Sampling of the signals is the fundamental operation in digital communication. A
continuous time signal is first converted to discrete time signal by sampling process.
Also it should be possible to recover or reconstruct the signal completely from its
samples.
The sampling theorem state that:
1- A band limited signal of finite energy, which has no frequency components
higher than W Hz, is completely described by specifying the values of the
signal at instant of time separated by 1/2W second and
2- A band limited signal of finite energy, which has no frequency components
higher than W Hz, may be completely recovered from the knowledge of its
samples taken at the rate of 2W samples per second.
Proof of sampling theorem:
Let x(t) the continuous time signal shown in figure below, its band width does not
contain any frequency components higher than W Hz. A sampling function
samples this signal regularly at the rate of fS sample per second.
Assume an analog waveform, 𝑥(𝑡) with a Fourier transform, 𝑋(𝑓), which is zero
outside the interval (−𝑓𝑚 < 𝑓 < 𝑓𝑚). The sampling of 𝑥(𝑡) can viewed as the
product of 𝑥(𝑡) with periodic train of unit impulse function 𝑥𝛿(𝑡) defined as
Digital communication
𝑥𝛿(𝑡) = ∑ 𝛿(𝑡 − 𝑛𝑇𝑠)
𝑛=−∞
The sifting property of unit impulse state that
𝑥(𝑡)𝛿(𝑡 − 𝑡0) = 𝑥(𝑡0)𝛿(𝑡 − 𝑡0)
Using this property so that:
∞
𝑥𝑠(𝑡) = 𝑥(𝑡)𝑥𝛿(𝑡) = ∑ 𝑥(𝑡)𝛿(𝑡 − 𝑛𝑇𝑠)
𝑛=−∞
∞
= ∑ 𝑥(𝑛𝑇𝑠)𝛿(𝑡 − 𝑛𝑇𝑠)
𝑛=−∞
Notice that the Fourier transform of an impulse train is another impulse train.
∞
1
𝑋𝛿(𝑓) = ∑ 𝛿(𝑓 − 𝑛𝑓𝑠)
𝑇
𝑠
𝑛=−∞
2
Digital communication
Convolution with an impulse function simply shifts the original function:
𝑋(𝑓) ∗ 𝛿(𝑓 − 𝑛𝑓𝑠)
We can now solve for the transform 𝑋𝑠(𝑓) of the sampled waveform:
𝑋(𝑓) ∗ 𝛿(𝑓 − 𝑛𝑓𝑠) = 𝑋(𝑓 − 𝑛𝑓𝑠)
So that
1 1
𝑋 (𝑓) = 𝑋(𝑓) ∗ 𝑋 (𝑓) = 𝑋(𝑓) ∗ [ ∑∞ 𝛿(𝑓 − 𝑛𝑓 )] = ∑∞ 𝑋(𝑓 −
𝑠 𝛿 𝑇𝑠 𝑛=−∞ 𝑠 𝑇𝑠 𝑛=−∞
𝑛𝑓𝑠)
When the sampling rate is chosen 𝑓𝑠 = 2𝑓𝑚 each spectral replicate is separated from
each of its neighbors by a frequency band exactly equal to 𝑓𝑠 hertz, and the analog
waveform ca theoretically be completely recovered from the samples, by the use of
filtering. It should be clear that if 𝑓𝑠 > 2𝑓𝑚, the replications will be move farther
apart in frequency making it easier to perform the filtering operation.
When the sampling rate is reduced, such that 𝑓𝑠 < 2𝑓𝑚, the replications will overlap,
as shown in figure below, and some information will be lost. This phenomenon is
called aliasing.
3
Digital communication
Sampled spectrum 𝑓𝑠 > 2𝑓𝑚
Sampled spectrum 𝑓𝑠 < 2𝑓𝑚
A bandlimited signal having no spectral components above 𝑓𝑚 hertz can be
1
determined uniquely by values sampled at uniform intervals of 𝑇𝑠 ≤ 2𝑓 𝑠𝑒𝑐.
𝑚
1
The sampling rate is 𝑓𝑠 =
𝑇𝑠
So that 𝑓𝑠 ≥ 2𝑓𝑚. The sampling rate 𝑓𝑠 = 2𝑓𝑚 is called Nyquist rate.
4
Digital communication
Example: Find the Nyquist rate and Nyquist interval for the following signals.
sin(500𝜋𝑡)
i- 𝑚(𝑡) =
𝜋𝑡
1
ii- 𝑚(𝑡) = cos(4000𝜋𝑡) cos(1000𝜋𝑡)
2𝜋
Solution:
i- 𝑤𝑡 = 500𝜋𝑡 ∴ 2𝜋𝑓 = 500𝜋 → 𝑓 = 250𝐻𝑧
1 1
Nyquist interval = = = 2 𝑚𝑠𝑒𝑐.
2𝑓𝑚𝑎𝑥 2×250
Nyquist rate =2𝑓𝑚𝑎𝑥 = 2 × 250 = 500𝐻𝑧
1 1
ii- 𝑚(𝑡) = { ( ) ( )}
[ cos 4000𝜋𝑡 − 1000𝜋𝑡 + cos 4000𝜋𝑡 + 1000𝜋𝑡 ]
2𝜋 2
1
= {cos(3000𝜋𝑡) + cos(5000𝜋𝑡)}
4𝜋
Then the highest frequency is 2500Hz
1 1
Nyquist interval = = = 0.2 𝑚𝑠𝑒𝑐.
2𝑓𝑚𝑎𝑥 2×2500
Nyquist rate =2𝑓𝑚𝑎𝑥 = 2 × 2500 = 5000𝐻𝑧
H. W:
Find the Nyquist interval and Nyquist rate for the following:
i- 1
cos(400𝜋𝑡) . cos(200𝜋𝑡)
2𝜋
ii- 1 𝑠𝑖𝑛𝜋𝑡
𝜋
5
Digital communication
Example:
A waveform [20+20sin(500t+30o] is to be sampled periodically and
reproduced from these sample values. Find maximum allowable time interval
between sample values, how many sample values are needed to be stored in
order to reproduce 1 sec of this waveform?.
Solution:
𝑥(𝑡) = 20 + 20 sin(500𝑡 + 300)
𝑤 = 500 → 2𝜋𝑓 = 500 → 𝑓 = 79.58 𝐻𝑧
Minimum sampling rate will be twice of the signal frequency:
𝑓𝑠(min) = 2 × 79.58 = 159.15 𝐻𝑧
1 1
𝑇𝑠(𝑚𝑎𝑥) = = 159.15 = 6.283 𝑚𝑠𝑒𝑐.
𝑓
𝑠(min)
1
Number of sample in 1𝑠𝑒𝑐 = = 159.16 ≈ 160 𝑠𝑎𝑚𝑝𝑙𝑒
6.283𝑚𝑠𝑒𝑐