SRI RAMAKRISHNA INSTITUTE OF TECHNOLOGY, COIMBATORE-10

An Autonomous Institution
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(Approved by AICTE, New Delhi – Affiliated to Anna University, Chennai)

DIGITAL COMMUNICATION
Low Pass Sampling Theorem

Dr.R.Lavanya
Assistant Professor(Sr.Gr.)/ECE
 Block Diagram of Typical Digital Communication
2
System

 If the message signal is analog in nature, eg. Speech or video. The first important step in any Digital
Communication System (DCS) is:
Transforming the information source to a form compatible with a digital system, i.e., analog to digital
conversion
 Formatting of Analog Data
3

 To transform an analog waveform into a form that is compatible with a digital

communication, the following steps are taken:
1. Sampling
2. Quantization

3. Encoding
SAMPLING
 Sampling Process
5

 Sampling process is a basic operation in digital communication.

 In this process the continuous-time analog signal is sampled by measuring its amplitude at
discrete instants of time.
 So, the continuous-time analog signal is converted into a corresponding sequence of samples that
are usually spaced uniformly in time.

Sampling

 It is necessary to choose the sampling rate properly, so the sequence of samples uniquely
defines the original analog signal
 Sampling Theorem
6

 Sampling theorem states that the sampling rate 𝑓𝑠 should be greater than or equal to twice the
message signal bandwidth 𝑊.

𝑓𝑠 ≥ 2𝑊

1 1
 In other words the sampling interval 𝑇𝑠 i.e., ( ) should be less than or equal to .
𝑓𝑠 2𝑊

1
𝑇𝑠 ≤
2𝑊
 Low Pass Sampling Theorem
7

 Deriving the sampling conditions for low pass signals.

 Let 𝑔 𝑡 be a message signal bandlimited to 𝑊 Hz.
1
 If the samples of this signals are taken at time interval 𝑇𝑠 ≤ , then these samples will define
2𝑊
the signal exactly, and the signal can be recovered properly at the receiver from the samples.
 Low Pass Sampling Theorem
8

Message signal 𝑔(𝑡)

Sampling signal s(𝑡)

Sampled signal 𝑔𝑠 (𝑡)

 Low Pass Sampling Theorem
9

 Let 𝑔 𝑡 be a message signal bandlimited to 𝑊 Hz.

 Let 𝑠 𝑡 be the sampling signal which is a periodic train of impulses having unit amplitude and
period 𝑇𝑠 .
 Let 𝑔 𝑡 and 𝑠 𝑡 be applied as inputs to a sampler which acts as a multiplier. Let the sampled
output be 𝑔𝑠 𝑡 .

Multiplier or impulse modulator

Message
signal 𝑔(𝑡)
⊗ Sampled
signal 𝑔𝑠 (𝑡)

Sampling
Fig : Sampler signal s(𝑡)
 Low Pass Sampling Theorem
10

 Thus the sampled output is given by

𝑔𝑠 𝑡 = 𝑔 𝑡 . 𝑠 𝑡 −−−− −(1)

 The sampling signal 𝑠 𝑡 is given by

𝑠 𝑡 = 𝛿(𝑡 − 𝑛 𝑇𝑠 ) −−−−−− −(2)

𝑛=−∞

 Substituting eqn. 2 in 1

𝑔𝑠 (𝑡) = 𝑔 𝑡 . 𝛿(𝑡 − 𝑛 𝑇𝑠 ) −−−−−− −(3)

𝑛=−∞
 Low Pass Sampling Theorem
11

 Eqn. 3 is the sampled output in time domain, in frequency domain

𝐺 𝑓 = 𝐹𝑇 𝑜𝑓 𝑔 𝑡
𝑆 𝑓 = 𝐹𝑇 𝑜𝑓 𝑠 𝑡
𝐺𝑠 𝑓 = 𝐹𝑇 𝑜𝑓 𝑔𝑠 𝑡

 Rewriting eqn. 1 in frequency domain

𝐺𝑠 𝑓 = 𝐺 𝑓 ∗ 𝑆 𝑓 −−− −(4)

 We know that Multiplication of two signals in time domain is equivalent to convolution of their
spectra in frequency domain.
 Low Pass Sampling Theorem
12

 The Fourier transform or the spectra of a periodic train of impulses is also a periodic train of
impulses with amplitude 𝑓𝑠 located at frequencies 0, ±𝑓𝑠 , ±2𝑓𝑠 , ±3𝑓𝑠 , ±4𝑓𝑠 .
 Therefore
∞

𝑆 𝑓 = 𝑓𝑠 . 𝛿(𝑓 − 𝑚 𝐹𝑠 ) −−−−−− −(5)

𝑚=−∞
where m is an integer
 Substituting eqn. 5 in 4
∞

𝐺𝑠 𝑓 = 𝐺 𝑓 ∗ 𝑓𝑠 𝛿(𝑓 − 𝑚 𝐹𝑠 ) −−−−−− −(6)

𝑚=−∞
 Low Pass Sampling Theorem
13

 Interchanging the order of summation and convolution in eqn. 6, we get

∞

𝐺𝑠 𝑓 = 𝑓𝑠 𝐺 𝑓 ∗ 𝛿(𝑓 − 𝑚 𝐹𝑠 ) −−−−−− −(7)

𝑚=−∞
Note: Delta or impulse function is only defined at t=0, i.e, 𝛿 𝑡 = 1; 𝑡 = 0 𝑎𝑛𝑑 𝛿 𝑡 = 0; 𝑡 ≠ 0.
∞
The area of impulse function is unity: −∞
𝛿 𝑡 𝑑𝑡 = 1
∞
Property : −∞
𝛿 𝑡 . 𝑓(𝑡)𝑑𝑡 = 𝑓(0) ;
∞
−∞
𝛿 𝑡 − 1 . 𝑓(𝑡)𝑑𝑡 = 𝑓(1)
Thus the convolution of a function with delta function will result in the shift in the spectrum of that function. Thus eqn. 7
becomes
∞

𝐺𝑠 𝑓 = 𝑓𝑠 𝐺(𝑓 − 𝑚 𝐹𝑠 ) −−−−−− −(8)

𝑚=−∞
 Low Pass Sampling Theorem
14

 Hence eqn. 8 is the spectra of the sampled signal

∞

𝐺𝑠 𝑓 = 𝑓𝑠 𝐺(𝑓 − 𝑚 𝐹𝑠 ) −−−−−− −(8)

𝑚=−∞
 Hence the spectra of the sampled output consists of the spectra of 𝐺(𝑓) located at frequencies
0, ±𝑓𝑠 , ±2𝑓𝑠 , ±3𝑓𝑠 , ±4𝑓𝑠 , etc. ,
 Graphical Proof of Sampling Theorem
15

 Let the spectra of g(t) be G(f) as shown in

fig.

Case i: Fs=2W
Let the spectra of the sampled output be as
shown in fig. from the spectra it can be
observed that the message signal g(t) can
be recovered by using a Low pass filter with
flat passband and sharp attenuation
characteristics. Thus the LPF has to be ideal
which is not practical.
 Graphical Proof of Sampling Theorem
16

Case ii: Fs > 2W ; eg. Fs = 3W

The spectra of the sampled output for the case when Fs>2W, is shown in fig. below, it can be
observed that the message signal g(t) can be recovered by using a practical Low pass filter due to
the presence of guard bands.
 Graphical Proof of Sampling Theorem
17

Case iii: Fs < 2W ; eg. Fs=1W

If Fs is chosen less than 2W, then the spectral components of the sampled output overlap and the
message signal cannot be recoverd by using the low pass filter.

 Thus Fs should be greater than or equal to 2W as stated by the sampling theorem to recover
the message signal.
 Analytical Proof of Sampling Theorem
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 Assuming the sampling rate, Fs=2W

 The sampled signal can be written as in eqn. 3

𝑔𝑠 𝑡 = 𝑔 𝑡 . 𝑠 𝑡 −−−− − 1

𝑠 𝑡 = 𝛿(𝑡 − 𝑛 𝑇𝑠 ) −−−−−− −(2)

𝑛=−∞

𝑔𝑠 (𝑡) = 𝑔 𝑡 . 𝛿(𝑡 − 𝑛 𝑇𝑠 ) −−−−−− −(3)

𝑛=−∞

The above eqn can be rewritten as

𝑔𝑠 𝑡 = 𝑔 𝑛𝑡𝑠 . 𝛿(𝑡 − 𝑛 𝑇𝑠 )
𝑛=−∞
 ∞

𝑔𝑠 𝑡 = 𝑔 𝑛𝑡𝑠 . 𝛿(𝑡 − 𝑛 𝑇𝑠 )
𝑛=−∞
 Taking Fourier Transform on both sides,

∞ ∞

𝐺𝑠 𝑓 = 𝐹𝑇 𝑔 𝑛𝑡𝑠 . 𝛿(𝑡 − 𝑛 𝑇𝑠 ) = 𝑔 𝑛𝑡𝑠 . 𝐹𝑇 𝛿 𝑡 − 𝑛 𝑇𝑠

𝑛=−∞ 𝑛=−∞

∞
𝐹𝑇 𝛿 𝑡 − 𝑛 𝑇𝑠 = −∞
𝛿 𝑡 − 𝑛𝑇𝑠 . 𝑒 −𝑗2𝜋𝑓𝑡 𝑑𝑡
 By the property of delta function
∞
𝛿 𝑡 − 𝑛𝑇𝑠 . 𝑒 −𝑗2𝜋𝑓𝑡 𝑑𝑡 = 𝑒 −𝑗2𝜋𝑓𝑛𝑇𝑠
−∞
∞
 Therefore, 𝐺𝑠 𝑓 = 𝑛=−∞ 𝑔 𝑛𝑡𝑠 . 𝑒 −𝑗2𝜋𝑓𝑛𝑇𝑠
1 𝜋𝑓𝑛
1 𝑛 −𝑗2𝜋𝑓𝑛(2𝑊) 𝑛
 Substituting 𝑇𝑠 = in 𝐺𝑠 𝑓 it becomes: 𝐺𝑠 𝑓 = ∞
𝑛=−∞ 𝑔 .𝑒 = ∞
𝑛=−∞ 𝑔 . 𝑒 −𝑗( 𝑊 )
2𝑊 2𝑊 2𝑊

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 ∞
 Using the spectra of the sampled signal, 𝐺𝑠 𝑓 = 𝑓𝑠 𝑚=−∞ 𝐺(𝑓 − 𝑚 𝐹𝑠 )
 Taking m=0 in the frequency range –W to W, 𝐺𝑠 𝑓 becomes, 𝐺𝑠 𝑓 = 𝑓𝑠 𝐺(𝑓)
 Rewriting, ∞
1 𝑛 −𝑗
𝜋𝑓𝑛
1
𝐺 𝑓 = 𝑓 𝐺𝑠 (𝑓)  𝐺(𝑓) = 𝑔 .𝑒 𝑊
𝑠
𝑓𝑠 2𝑊
𝑛=−∞
 Taking Inverse Fourier transform both sides of the above equation
∞
1 𝑛 −𝑗
𝜋𝑓𝑛
𝑔(𝑡) = 𝐼𝐹𝑇 𝑔 .𝑒 𝑊
𝑓𝑠 2𝑊
𝑛=−∞

∞ ∞
1 𝑛 −𝑗
𝜋𝑓𝑛
= 𝑔 .𝑒 𝑊 . 𝑒 𝑗2𝜋𝑓𝑡 𝑑𝑓
𝑓𝑠 2𝑊
−∞ 𝑛=−∞

 In the range -W to W 𝑊 ∞
1 𝑛 −𝑗
𝜋𝑓𝑛
= 𝑔 .𝑒 𝑊 . 𝑒 𝑗2𝜋𝑓𝑡 𝑑𝑓
𝑓𝑠 2𝑊
−𝑊 𝑛=−∞
∞ 𝑊
∞ 𝑊
1 𝑛 𝜋𝑓𝑛 1 𝑛 𝑛
−𝑗𝜋𝑓 𝑊−2𝑡
= 𝑔 . 𝑒
−𝑗 𝑊
. 𝑒 𝑗2𝜋𝑓𝑡 𝑑𝑓 = 𝑔 . 𝑒 𝑑𝑓
2𝑊 2𝑊 2𝑊 2𝑊
𝑛=−∞ −𝑊
20 𝑛=−∞ −𝑊
 ∞ 𝑊
1 𝑛 𝑛
−𝑗𝜋𝑓 𝑊−2𝑡
= 𝑔 . 𝑒 𝑑𝑓
2𝑊 2𝑊
𝑛=−∞ −𝑊

𝑛 𝑊
∞ −𝑗𝜋𝑓 𝑊−2𝑡
1 𝑛 𝑒
= 𝑔 . 𝑛
2𝑊 2𝑊 −𝑗𝜋
𝑛=−∞ 𝑊 − 2𝑡 −𝑊

∞ 𝑛 𝑛
−𝑗𝜋𝑊 𝑊−2𝑡 𝑗𝜋𝑊 𝑊−2𝑡
1 𝑛 𝑒 𝑒
= 𝑔 . 𝑛 − 𝑛
2𝑊 2𝑊 −𝑗𝜋 − 2𝑡 −𝑗𝜋
𝑛=−∞ 𝑊 𝑊 − 2𝑡

∞ 𝑛 𝑛 ∞ 𝑛
1 𝑛 𝑒
𝑗𝜋𝑊 𝑊−2𝑡
− 𝑒
−𝑗𝜋𝑊 𝑊−2𝑡 𝑛 𝑠𝑖𝑛 𝜋𝑊 − 2𝑡
= 𝑔 . 𝑊
= 𝑔 . 𝑛 2𝑊 𝑛
2𝑊 2𝑊 𝜋𝑊
𝑛=−∞ 𝑗𝜋
𝑊
− 2𝑡 𝑛=−∞ 𝑊 − 2𝑡
∞
𝑛 𝑛
= 𝑔 . 𝑠𝑖𝑛𝑐 𝑊 − 2𝑡 ------------------(A)
2𝑊 𝑊
𝑛=−∞

 This is the response of an ideal low pass filter having passband of –W, W. This shows that the original message signal g(t)
21 can be obtained perfectly at the output of LPF when Fs =2W.
 Sampling Theorem
22

 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. (using a low
pass filter of cutoff freq. W)

 Nyquist rate (fs)

 The sampling rate of 2W samples per second for a signal bandwidth of W hertz
 Nyquist interval (Ts)
 1/2W (measured in seconds)
 References
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 1. S. Haykin, “Digital Communications”, John Wiley, 2014 reprint.

 2. B. Sklar, “Digital Communication Fundamentals and Applications”,
2nd Edition, Pearson Education, 2013.
THANK YOU