Communication Engineering [AGM]
Communication Engineering
UNIT V - Notes
Pulse Modulation Techniques:
Sampling Theorem & Types of Sampling
Sampling of Band-pass signal
Linear and Non-linear Quantization
Aliasing Effect
Aperture Effect
Reconstruction of Filter
Time Division Multiplexing
Pulse Amplitude Modulation
Pulse Time Modulation
PCM, DM, ADM
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Communication Engineering [AGM]
The signals we use in the real world, such as our voices, are called "analog" signals. To process these
signals in computers, we need to convert the signals to "digital" form. While an analog signal is
continuous in both time and amplitude, a digital signal is discrete in both time and amplitude. To
convert a signal from continuous time to discrete time, a process called sampling is used. The value
of the signal is measured at certain intervals in time. Each measurement is referred to as a sample.
1] Sampling Theorem:
Statement: A continuous time signal can be represented in its samples and can be recovered back
when sampling frequency 𝑓𝑠 is greater than or equal to the twice the highest frequency component
of message signal. i. e.
𝑓𝑠 ≥ 2 𝑓𝑚
Nyquist Criterion:
It gives minimum sampling rate at which signal can be converted into samples and can be recovered
back without distortion. It is also known as Nyquist Rate. It is given by the formula:
fN = 2 fm
2] Types of Sampling:
1. Ideal sampling [Impulse Sampling]
Impulse sampling can be performed by multiplying input signal x(t) with impulse train of period 'T'.
Here, the amplitude of impulse changes with respect to amplitude of input signal x(t).
The output of sampler is given by
𝑦(𝑡) = 𝑥(𝑡) ∗ 𝑖𝑚𝑝𝑢𝑙𝑠𝑒 𝑡𝑟𝑎𝑖𝑛
𝑦(𝑡) = 𝑥(𝑡) ∗ ∑ 𝛿(𝑡 − 𝑛𝑇)
𝑛=−∞
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This is called ideal sampling or impulse sampling. This cannot be used practically because pulse width
cannot be zero and the generation of impulse train is not possible practically.
2. Natural Sampling:
Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse train of
period T. i.e. input signal x(t) is multiplied with pulse train as shown below:
𝑦(𝑡) = 𝑥(𝑡) ∗ 𝑝𝑢𝑙𝑠𝑒 𝑡𝑟𝑎𝑖𝑛
3. Flat Top Sampling:
During transmission, noise is introduced at top of the transmission pulse which can be easily
removed if the pulse is in the form of flat top. Here, the top of the samples are flat i.e. they have
constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat top sampling
makes use of sample and hold circuit.
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3] Sampling of Band Pass Signals:
In case of band pass signals, the spectrum of band pass signal is X [ω] = 0 for the frequencies outside
the range f1 ≤ f ≤ f2 . The frequency f1 is always greater than zero. Plus, there is no aliasing effect
when fs > 2f2 .
But it has two disadvantages:
The sampling rate is large in proportion with f2 . This has practical limitations.
The sampled signal spectrum has spectral gaps.
To overcome this, the band pass theorem states that the input signal x(t) can be converted into its
samples and can be recovered back without distortion when sampling frequency fs < 2f2 .
For band pass signals of bandwidth 2fm and the minimum sampling rate f = 2 ∗ Bandwidth = 4fm
The spectrum of sampled signal is given by
∞
1
Y[ω] = ∑ X[ω − 2n ∗ Bandwidth]
T
n=−∞
4] Aliasing Effect:
Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a
signal, taking on the identity of a low-frequency component in the spectrum of its sampled version.”
The corrective measures taken to reduce the effect of Aliasing are −
In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the
sampler, to eliminate the high frequency components, which are unwanted.
The signal which is sampled after filtering, is sampled at a rate slightly higher than the
Nyquist rate.
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5] Aperture Effect:
It takes place in Flat Top Sampling. Aperture error is the difference between the actual value of the
input signal, and the flat-topped sample value. The magnitude of this difference is related to the
input frequency and sampling width. It would appear that reducing the sampling aperture to near
zero would eliminate this form of distortion.
6] Quantization:
The digitization of analog signals involves the rounding off of the values which are approximately
equal to the analog values. The method of sampling chooses a few points on the analog signal and
then these points are joined to round off the value to a near stabilized value. Such a process is called
as Quantization.
An analog signal to get converted into digital, has to undergo sampling and quantizing. The
quantizing of an analog signal is done by discretizing the signal with a number of quantization levels.
Quantization is representing the sampled values of the amplitude by a finite set of levels, which
means converting a continuous-amplitude sample into a discrete-time signal. The following figure
shows how an analog signal gets quantized. The blue line represents analog signal while the brown
one represents the quantized signal.
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IMPORTANT QUESTIONS ON UNIT-V
[Following questions were asked in previous university exams: W2017, S2018, W2018, S2019 & W2019]
Q1. What is Pulse Modulation? Explain Pulse Amplitude Modulation with suitable waveforms.
Q2. Explain the working of Delta Modulation (DM).
Q3. Draw and explain block diagram of Time Division Multiplexing system.
Q4. Explain the principle of operation of PCM.
Q5. Explain Sampling theorem and its Nyquist Criteria.
Q6. Explain Pulse Position Modulation (PPM) with its advantages.
Q7. Explain Aliasing Effect? How it can be eliminated?
Q8. What is Sampling Theorem? Explain different types of sampling.
Q9. What is Quantization? Explain linear and non-linear quantization.
Q10. Explain PAM generation.
Reference Book:
Communication Engineering, U. A. Bakshi & A. P. Godse, Technical Publications
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