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Lect 7N

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17 views

Lect 7N

Uploaded by

omaralsharqawi4016
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Communication System

Third Year

First Term (2021-2022)

Reference : Communication Systems Engineering, John G. Proakis and Masoud Salehi, 2nd Ed.

Lecturer: Dr. Rna Ghallab


Instructions
Chapter 4

Analog-to-digital conversion
Analog-to-digital conversion

Analog-to-digital conversion is (basically) a 2 step process:


1. Sampling
• Convert from continuous-time analog signal 𝑥𝑎 (𝑡), into a discrete-time signal by taking
the “samples” at discrete-time intervals (Ts).

2. Quantization
• Convert from discrete-time continuous valued signal to discrete time discrete valued
signal.
3. Encoding
1. Sampling

• Sampling is the processes of converting continuous 𝑥𝑎 (𝑡), into a discrete-time signal by taking the
“samples” at discrete-time intervals.
– Sampling analog signals makes them discrete in time but still continuous valued
– If done properly (Nyquist theorem is satisfied), sampling does not introduce distortion
• Sampled values:
– The value of the function at the sampling points
• Sampling interval:
– The time that separates sampling points (interval b/w samples),
– If the signal is slowly varying, then fewer samples per second will be required than if the waveform
is rapidly varying
– So, the optimum sampling rate depends on maximum frequency component present in the signal.
1. Sampling
• Sampling Rate (or sampling frequency fs):
The rate at which the signal is sampled, expressed as the number of
samples per second reciprocal of the sampling interval), 1/Ts = fs
• Nyquist Sampling Theorem (or Nyquist Criterion):
✓ The sampling is performed at a proper rate, no information is lost about the original signal and it can
be properly reconstructed without any distortion under the condition;
𝑓𝑠 ≥ 2𝑓𝑚𝑎𝑥
✓ Nyquist rate is the minimum sampling rate of (2W) samples per second, for an analog signal
bandwidth of W Hz. 𝑓𝑁𝑞 = 2𝑓𝑚𝑎𝑥

• aliasing or simply Overlap:


✓ The phenomenon of the presence of high spectrum of the original analog signal.
✓ 𝑓𝑠 < 2𝑓𝑚𝑎𝑥
1. Sampling
Aliasing or simply Overlap when 𝑓𝑠 < 2𝑓𝑚𝑎𝑥
1. Sampling
• Methods of Sampling:
1. Ideal Sampling (Impulse sampling)
❑ An impulse at each sampling instant
❑ It accomplished by the multiplication of the signal
𝑥𝑎 (𝑡), by a train of impulse functions select sample
values at regular intervals.

𝒙𝒔 (𝒕) = 𝒙(𝒕) ෍ 𝜹(𝒕 − 𝒏𝒕𝒔 )


𝒏=−∞ Represent the Fourier Series
1.Ideal Sampling (Impulse sampling)
Fourier Transform of the sampled signal is the Fourier Transform of the original signal at rate of 1/Ts
1. Sampling
• Methods of Sampling:
2. Natural Sampling (Pulse sampling)

❑ A pulse of short width with varying amplitude with


natural tops.
❑ If we multiply x(t) by a train of rectangular pulses
𝑥𝑃 (𝑡) , we obtain a gated waveform that
approximates the ideal sampled waveform, known as
natural sampling gating.

𝒙𝒔 𝒕 = 𝒙 𝒕 ∗ 𝒙𝒑 𝒕
2. Natural Sampling (Pulse sampling)
• Each pulse in xp(t) has width Ts and amplitude 1/T, the top of each pulse follows the variation of the
signal being sampled Xs (f) is the replication of X(f) periodically every fs Xs (f) is weighted by Cn (
Fourier Series Coefficient).
• The problem with a natural sampled waveform is that the tops of the sample pulses are not flat and it is
not compatible with a digital system since the amplitude of each sample has infinite number of possible
values.
• Another technique known as flat top sampling is used to alleviate this problem.
1. Sampling
• Methods of Sampling:
3. Flat Top Sampling
❑ A pulse of short width with varying amplitude with flat tops.
❑ Here, the pulse is held to a constant height for the whole sample period
❑ Flat top sampling is obtained by the convolution of the signal obtained after ideal sampling with a unity
amplitude rectangular pulse, p(t).
❑ This technique is used to realize Sample-and -Hold (S/H) operation
1. Sampling
• Methods of Sampling:
3. Flat Top Sampling
Recovery of original signal
• For no aliasing; 𝑓𝑠 ≥ 2𝑓𝑚𝑎𝑥 , 𝑓𝑚𝑎𝑥 = B
• One way of recovering the original signal from sampled signal 𝒙𝒔 𝒕 is to
pass it through the low pass filter (LPF) as shown below:
Recovery of original signal
• In case of 𝑓𝑠 < 2𝑓𝑚𝑎𝑥 , then the aliasing is occurred at the receiver.
• The original transmitted signal is not recovery correctly as an irreversible overlap of
the spectral replicas is produced
Recovery of original signal

• If the waveform is undersampled (i.e. fs < 2B) then the


signal at the output of the filter will be different from
the original signal spectrum due to the spectral
overlap in the samples.

• This is the outcome of aliasing!


• This implies that whenever the sampling condition is
not met.
Antialiasing Filter
Antialiasing Filter
• Aliasing is prevented by forcing the bandwidth of the sampled signal to satisfy the requirement of the
Sampling Theorem
Pulse Amplitude Modulation
Application of Sampling Theorem
PAM/TDM

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