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EC T62 - DIGITAL COMMUNICATION

BASE BAND TRANSMISSION


1. Block diagram of a Digital Communication System:

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EC T62 - DIGITAL COMMUNICATION

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EC T62 - DIGITAL COMMUNICATION

2. Baseband System:

The Sampling Theoram:

If the sampling rate in any pulse modulation system exceeds twice the maximum
signal frequency, the original signal can be reconstructed in the receiver with
minimal distortion.

Where,

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Nyquist rate is the rate at which sampling of a signal is done so that


overlapping of frequency (aliasing) does not take place. When the sampling rate
become exactly equal to 2fm samples per second, then the specific rate is known
as Nyquist rate. It is also known as the minimum sampling rate and given by:
fs =2fm
Nyquist Interval is given by-

Types of sampling:

a) Impulse sampling:

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EC T62 - DIGITAL COMMUNICATION

to the frequency-domain
convolution-

fs > 2fm

fs < 2fm

Figure (1): Impulse Sampling

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b) Natural sampling:

(1)

Figure (2): Natural Sampling

Now, by applying the Fourier series, the periodic pulse train can expressed as-
(2)

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Now, consider-

Where T is the pulse width and 1/T is the pulse amplitude and

Combining Eq. (1) and (2) it yields-

Using fourier translation property-

c) Flat-Top Sampling (Sample and Hold):

1
3

(1)

1
3

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Figure (3): Flat-Top Sampling

Effect of under sampling: ALIASING:

It is the effect in which overlapping of a frequency components takes place at


the frequency higher than Nyquist rate. Signal loss may occur due to aliasing effect.
We can say that aliasing is the phenomena in which a high frequency component in
the frequency spectrum of a signal takes identity of a lower frequency component in
the same spectrum of the sampled signal.

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Because of overlapping due to process of aliasing, sometimes it is not possible to


overcome the sampled signal x(t) from the sampled signal g(t) by applying the process
of low pass filtering since the spectral components in the overlap regions . Hence this
causes the signal to destroy.
The Effect of Aliasing can be reduced:

1) Pre alias filter (the waveform is shown below) must be used to limit band of
frequency of the required signal fm Hz.
2) Sampling frequency fs must be selected such that fs > 2fm.

Need of Oversampling:

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3. Wave form representation of binary digits:

Figure (1):
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PCM Waveform Types:

These waveforms are also called Line Codes because of its application in
digital telephony.
When pulse modulation is applied to a non-binary symbol, the resulting
waveform is called M-ary PCM waveform.

The PCM waveform falls into following groups:

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Parameters to be considered while selecting a PCM waveform:

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4. Pulse Code Modulation (PCM):

Pulse-code modulation (PCM) refers to a system in which the standard values of a


quantized wave (explained in the following paragraphs) are indicated by a series of
coded pulses. When these pulses are decoded, they indicate the standard values of the
original quantized wave. these codes may be binary, in which the symbol for each
quantized element will consist of pulses and spaces: ternary, where the code for each
element consists of any one of three distinct kinds of values (such as positive pulses,
negative pulses, and spaces); or n-ary, in which the code for each element consists of
nay number (n) of distinct values.
Figure 1 shows the relationship between decimal numbers, binary numbers, and
a pulse-code waveform that represents the numbers. The table is for a 16-level code;
that is, 16 standard values of a quantized wave could be represented by these pulse
groups. Only the presence or absence of the pulses is important. The next step up
would be a 32-level code, with each decimal number represented by a series of five
binary digits, rather than the four digits of figure 1. Six-digit groups would provide a
64-level code, seven digits a 128-level code, and so forth. Figure 2 shows the
application of pulse-coded groups to the standard values of a quantized wave.

Figure-1.

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Figure-2. Pulse-code modulation of a quantized wave

PCM Transmitter and Receiver:

 The band pass filter limits the frequency of the analog input signal to the standard
voice-band frequency range of 300 Hz to 3000 Hz.
 The sample- and- hold circuit periodically samples the analog input signal and
converts those samples to a multilevel PAM signal.
 The analog-to-digital converter (ADC) converts the PAM samples to parallel PCM
codes, which are converted to serial binary data in the parallel-to-serial converter
and then outputted onto the transmission linear serial digital pulses.
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 The transmission line repeaters are placed at prescribed distances to regenerate the
digital pulses. --In the receiver, the serial-to-parallel converter converts serial
pulses received from the transmission line to parallel PCM codes.
 The digital-to-analog converter (DAC) converts the parallel PCM codes to
multilevel PAM signals. The hold circuit is basically a low pass filter that converts
the PAM signals back to its original analog form.

Quantization Process:
Quantization is the process of converting an infinite number of possibilities to a
finite number of conditions. Analog signals contain an infinite number of amplitude
possibilities. Converting an analog signal to a PCM code with a limited number of
combinations requires quantization. It happens after the sampling process.
It is of two types-
1) Uniform Quantization
2) Non-Uniform Quantization.

When the steps of a sampled signal are uniform in size the quantization is called
uniform quantization. With uniform quantization the signal-to-noise ratio (SNR) is
worse for low level signals than for high level signals.
Similarly, when the steps of the sampled signal are non-uniform in size the
quantization is called non-uniform quantization. The non-uniform quantization can
provide fine quantization of the weak signals and coarse quantization of the strong
signals. In this case the quantization noise is proportional to signal size.

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There are two ways of achieving Non-Uniform Quantization:


1) Using a non-uniform quantizer with non-uniform characteristics.
2) Making use of Companding (means to compress the signal at the transmitter
side and expanding the compressed signal at the receiver side).

Companding Characteristics:
1) µ-Law characteristics:

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EC T62 - DIGITAL COMMUNICATION

2) A-Law characteristics:
Used mainly in europe. It is defined as-

Where A is positive constant, x and y are same as above. An A-Law characteristics is


shown above in figure b. Standard value of A is 87.6

Signal-to-Noise ratio of quantized pulses (S/N)q:

Let the quantizer error variance is found to be-

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error

Now, the peak power of the analog signal can be expressed as-

Therefore,

PCM word size:


Let,

5. Differential Pulse Code Modulation (DPCM):

A schematic diagram for the basic DPCM modulator is shown in Figure below.
In this an error sample ep(kTs) is fed.

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We will now develop a simple analytical structure for a DPCM encoding


scheme to bring out the role that nay be played by the prediction unit.

A block schematic diagram of a DPCM demodulator is shown in Figure below.

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Single-Tap Prediction in DPCM:

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Calculation of SQNR for DPCM:

6. Delta Modulation:
If the sampling interval ‘Ts’ in DPCM is reduced considerably, i.e. if we sample a
band limited signal at a rate much faster than the Nyquist sampling rate, the adjacent
samples should have higher correlation as shown below.

The sample-to-sample amplitude difference will usually be very small. So, one
may even think of only 1-bit quantization of the difference signal. The principle of
Delta Modulation (DM) is based on this premise.
Delta modulation is also viewed as a 1-bit DPCM scheme. The 1-bit quantizer
is equivalent to a two-level comparator (also called as a hard limiter). Figure shows
the schematic arrangement for generating a delta-modulated signal. Note that,

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Features of Delta Modulation:

Now,

Here, ‘s’ is half of the step-size δ.

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Figure below shows a scheme for DM demodulation. The input to the


demodulator is a binary sequence and the demodulator normally starts with no prior
information about the incoming sequence.

Now,

From the close loop including the delay-element in the accumulation unit in the Delta
modulator structure, we can write-

Hence, we may express the error signal as,

Advantages of a Delta Modulator over DPCM:

As one sample of x (kTs) is represented by only one bit after delta modulation, no
elaborate word-level synchronization is necessary at the input of the demodulator.
This reduces hardware complexity compared to a PCM or DPCM demodulator. Bit-
timing synchronization is, however, necessary if the demodulator in implemented
digitally.
Overall complexity of a delta modulator-demodulator is less compared to
DPCM as the predictor unit is absent in DM.

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Limitations of Delta Modulation:

a) Slope over load distortion: If the input signal amplitude changes fast, the step-by-
step accumulation process may not catch up with the rate of change. This happens
initially when the demodulator starts operation from cold-start but is usually of
negligible effect for speech. However, if this phenomenon occurs frequently
(which indirectly implies smaller value of auto-correlation co-efficient Rxx (τ) over
a short time interval) the quality of the received signal suffers. The received signal
is said to suffer from slope-overload distortion. An intuitive remedy for this
problem is to increase the step-size δ

b) Granular noise: If the step-size is made arbitrarily large to avoid slope-overload


distortion, it may lead to ‘granular noise’. Imagine that the input speech signal is
fluctuating but very close to zero over limited time duration. This may happen due
to pauses between sentences or else. During such moments, our delta modulator is
likely to produce a fairly long sequence of 101010…., reflecting that the
accumulator output is close but alternating around the input signal. This
phenomenon is manifested at the output of the delta demodulator as a small but
perceptible noisy background. This is known as ‘granular noise’.
An expert listener can recognize the crackling sound. This noise should be kept
well within a tolerable limit while deciding the step-size. Larger step-size increases
the granular noise while smaller step size increases the degree of slope-overload
distortion. In the first level of design, more care is given to avoid the slope-
overload distortion. We will briefly discuss about this approach while keeping the
step-size fixed. A more efficient approach of adapting the step-size, leading to
Adaptive Delta Modulation (ADM), is excluded.

Condition for avoiding slope overload: If an input signal changes more than half of
the step size (i.e. by ‘s’) within a sampling interval, there will be slope-overload
distortion. So, the desired limiting condition on the input signal x(t) for avoiding
slope-overloading is,

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7. Adaptive Delta Modulation (ADM):

The principle underlying all ADM algorithms is two-fold:

1) If the successive errors are of opposite polarity, then the DM is operating in its
granular mode; in this case, it may be advantageous to reduce the step-size.
2) If, however successive errors are of the same polarity, then the DM is operating
in its slope overload mode; in this case, the step-size should be increased.

Thus by varying the step-size in accordance with this principle, the Delta
Modulator is enabled to cope with changes in the input signal.
Figure below shows the block diagram of ADM based on increasing or
decreasing the step-size by 50 % at each iteration of the adaptive process. The
algorithm for adaptation of step-size is defined by-

Where, is the step-size at iteration (time step) of n algorithm, and mq[n] is the 1-
bit quantizer output that equals ±1.

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Let,

The result is plotted in figure below: Part (a) is for LDM and Part (b) is for ADM.

Observations made from the waveforms:


1) ADM tracks the changes in input sinusoidal signal much better than LDM. This
improvement in the performance of ADM is due to adaptation of the step-size
in successive iteration of the algorithm. The reduced step-size in ADM results
in small quantization error than LDM.
2) The improved tracking performance of the ADM results in and output signal
with much lower bit rate as compared to LDM on an average.

8. Detection of signals in Gaussian noise:

Noise in Communication Systems is often assumed to be Additive White Gaussian


Noise (AWGN).

Additive: Noise is usually additive in that it adds to the information bearing signal. A
model of the received signal with additive noise is shown below.

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The signal (information bearing) is at its weakest (most vulnerable) at the receiver
input. Noise at the other points (e.g. Receiver) can also be referred to the input.
The noise is uncorrelated with the signal, i.e. independent of the signal and we
may state, for average powers
Output Power = Signal Power + Noise Power
= (S+N)
White: As we have stated noise is assumed to have a uniform noise power spectral
density, given that the noise is not band limited by some filter bandwidth.
We have denoted noise power spectral density by po  f  .
White noise = po  f  = Constant
Also Noise power = po Bn

Gaussian: We generally assume that noise voltage amplitudes have a Gaussian or


Normal distribution.

Derivation:

For binary channel, the transmitted signal over symbol interval (0, T) is
represented by-

(2)

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Here,
a) Demodulation means recovery of waveforms to an undistorted baseband pulse.
b) Detection indicates the decision making process in selecting the digital meaning of
the waveform.
c) If error correcting coding is not present, the detector output consists of estimates of
message symbols (or bits), (also called hard-decisions).

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Let,

So that,

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Maximum likelihood detection of binary signals in Gaussian noise:


Let,
-

For Since the conditional density function p (z/si) is called likelihood of si, so
the formulation shown below is called Maximum likelihood ratio.
(A)

Where, γ = threshold signal. Also if the likelihood p (z/si) are symmetrical, then the
substitution of p (z/s1) and p (z/s2) in (A) yields-

Error Probability PB:

(B)

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Similarly when s2(t) is sent then,

(C)

(D)

Combining (B) and (D), we get-

And Q(x) can be expressed as-

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Vector representation of signals and noise:

(3 (a))

(3 (b))

When Kj=1, then the signal space is called orthonormal space. From geometrical point
of view each ѱj (t) is mutually perpendicular to each of the other ѱk (t) for j≠0.

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Where, Sk + n and Sj + n are the resultant vector around cloud points Sk and Sj.

Let, the Waveform Energy Ei can be derived as-

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For, Kj=1

9. Error probability performance of binary signaling:

1) Unipolar Signaling:

As per the definition of orthogonal signaling, s1(t) and s2(t) have zero
correlation over each symbol time duration.

Because s2(t) is zero in this case, so the pulses are clearly unipolar.

A correlator can be used as receiving filter for such signaling as shown in figure b
below-

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Thus the optimum decision threshold used in filter is given by-

Then, the detector detects that s1(t) was sent otherwise s2(t) was sent.
Let, the energy difference signal is-

Since s2(t)=0,

Then bit error probability is-

Since,
Eb (Average energy per bit) = A2T/2
2) Bipolar Signaling:
Bipolar signaling is an example of baseband antipodal signaling (Antipodal signaling
means binary signaling that are mirror image of each other i.e. s1(t) = -s2(t)), where-

Correlator detector for bipolar signaling is shown below-

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Figure b, uses two correlator as a receiving filter.


 The first correlator multiplies and integrates the noisy I/P r(t) with the
prototype signal s1(t).
 The second correlator multiplies and integrates the noisy I/P r(t) with the
prototype signal s2(t)
The receiver then seeks the largest O/P voltage (i.e. the best match to make detection).
So, the test statics used for detection as shown in figure is-

For antipodal signals, a1 = -a2, therefore,

Thus,
 If z(T) is positive, then signal is detected as s1(t).
 If z(T) is negative then signal is detected as s2(t).

Bit error probability is-

Where,

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10. Inter symbol interference (ISI):

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Methods to reduce ISI:


 Pulse shaping.
 Equalization.
 Eye Pattern.

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11. Eye Pattern:


The amount of ISI and noise that is present in the received signal can be viewed on
an oscilloscope. Specifically, we may display the received signal on the vertical input
with the horizontal sweep rate set at 1/T. The resulting oscilloscope display is called
an eye pattern because of its resemblance to the human eye. Examples of two eye
patterns, one for binary PAM and the other for quaternary (M = 4) PAM, are
illustrated in figure (a) below.

The effect of ISI is to cause the eye to close, thereby reducing the margin for additive
noise to cause errors. Figure (b) illustrates the effect of ISI in reducing the opening of
the eye.

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