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DIGI COM 1 - Introduction and Sampling

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31 views29 pages

DIGI COM 1 - Introduction and Sampling

Uploaded by

Sachin Sharma
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Digital Communication

Communication
• Main purpose of communication is to transfer information from a
source to a recipient via a channel or medium.

• Basic block diagram of a communicationsystem:

Source Transmitter Channel Receiver Destination


Why Digital Communications?
➢ Digital communication is rugged in the sense that it is more
immune to channel noise, distortion and interference.
(Two-state signal representation)
➢ Regeneration versus Amplification

– Easy to regenerate the distorted signal

– Regenerative repeaters along the transmission path


can detect a digital signal and retransmit a new,
clean (noise free) signal
– These repeaters prevent accumulation of noise
along the path
– This is not possible with analog communication
systems
➢ Hardware is more flexible
– Digital hardware implementation is flexible and low
cost
➢ Easier and more efficient to multiplex several digital
signals
➢ Can combine different signal types – data, voice, text, etc.
– Data communication in computers is digital in nature whereas
voice communication between people is analog in nature
– The two types of communication are difficult to combine over
the same medium in the analog domain.
– Using digital techniques, it is possible to combine both
format for transmission through a common medium

➢ Encryption and privacy techniques are easier to implement

Hence

❖ Better overall performance

❖ Digital communication is inherently more efficient than analog


in realizing the exchange of SNR for bandwidth
Disadvantages

• Requires reliable “synchronization”

• Requires A/D conversions at high rate

• Requires larger bandwidth

Performance metric

• for analog systems it is fidelity

• for digital it is the bit rate and error probability.


What are the Features of a Good
Communication System?
• Small signal power (measured in Watts or dBm)
• Large data rate (measured in bits/sec)
• Small bandwidth (measured in Hertz)
• Low distortion (measured in SNR or bit error rate)
• Low cost - with digital communications, large complexity does
not always result in large cost

In practice, there must be tradeoffs made in achieving


these goals

41
Source
ANALOG-TO-DIGITAL CONVERSION
(Pulse Code Modulation)
PCM
• PCM consists of three steps to digitize an analog
signal:
1. Sampling
2. Quantization
3. Binary encoding
• Before we sample, we have to filter the signal
to limit the maximum frequency of the signal as
it affects the sampling rate.
• Filtering should ensure that we do not distort
the signal
Sampling
• The process of generating pulses of zero width and of amplitude
equal to the instantaneous amplitude of the analog signal.
• The no. of pulses per second is called “sampling rate”
• Analog signal is sampled every TS secs.
• Ts is referred to as the sampling interval.
• fs = 1/Ts is called the sampling rate or sampling frequency
• The process is referred to as Pulse Amplitude Modulation (PAM)
• The outcome is a signal with analog (non integer) values
According to the Nyquist theorem, the sampling rate must be
at least 2 times the highest frequency contained in the signal.
There are 3 sampling methods:
• Ideal - an impulse at each sampling instant
• Natural - a pulse of short width with varying amplitude
• Flat top - sample and hold, like natural but with single amplitude
value
Ideal Sampling ( or Impulse Sampling)

Is accomplished by the multiplication of the signal x(t) by the uniform train of


impulses (comb function)
Consider the instantaneous sampling of the analog signal x(t)

Train of impulse functions select sample values at regular intervals



x s (t ) = x (t )  (t − n Ts )
n = −

Fourier Series representation:


 
2

j n s t

n=−
(t − n T s ) =
Ts
1
n=−
 e ,  s =
Ts
Ideal Sampling ( or Impulse Sampling)

It shows that the Fourier Transform of the sampled signal is the


Fourier Transform of the original signal at rate of 1/Ts


Ts is called the Nyquist interval: It is the longest time interval that can be used for
sampling a bandlimited signal and still allow reconstruction of the signal at the
receiver without distortion
Natural Sampling

a pulse of short width with varying amplitude


Natural Sampling

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

x s (t ) = x ( t ) x p( t )

= x ( t)  c ne j 2 n f s t

n=−

X s ( f ) =  [ x (t ) x p(t )]

j 2 n f s t
=  n=−
c n [ x ( t ) e

=  cn X [ f − nf s]
n = −

Fourier Series Coefficient


c n = ( / Ts) sinc ( n / Ts )
•Each pulse in xp(t) has width  and amplitude  /Ts
•The top of each pulse follows the variation of the signal being
sampled
•Xs (f) is the replication of X(f) periodically every fs Hz
•Xs (f) is weighted by Cn
•The problem with a natural sampled waveform is that the tops of the
sample pulses are not flat.
➢ 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
• sampled pulse of short width with flat tops
• Sample-and-Hold (S/H) operation
Flat top sampling (Frequency domain)

◼Flat top sampling becomes identical to ideal sampling as the width of the pulses
become shorter
Aperture Effect in Natural and Flat top sampling

➢Aperture Effect Distorted output


• Soln.
✓ Flatter sinc envelope
• position of 1st minima (Ts /  )
• Hence 
✓ Using equalizer
• Problem of reducing 
• Output amplitude  /Ts
• If  = Ts sampled output independent of 
• But Aperture Effect
Recovering the Analog Signal

Pass sampled signal Xs(f) through a Low Pass filter (LPF)

If fs > 2B then we recover x(t) exactly


Else we run into some problems and signal is not
fully recovered
Aliasing (Spectral Folding) Effect
Spectral components of adjacent samples overlap
• When fs < 2fm (under sampling)
Solution : Anti-Aliasing Analog Filter

• All physically realizable signals are not completely band limited


• If there is a significant amount of energy in frequencies above half the
sampling frequency (fs/ 2), aliasing will occur
• Aliasing can be prevented by first passing the analog signal through
an anti-aliasing filter (also called a pre filter) before sampling is
performed
• The anti-aliasing filter is simply a LPF with cutoff frequency equal to
half the sample rate

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