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2. Baseband System:
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,
Types of sampling:
a) Impulse sampling:
to the frequency-domain
convolution-
fs > 2fm
fs < 2fm
b) Natural sampling:
(1)
Now, by applying the Fourier series, the periodic pulse train can expressed as-
(2)
Now, consider-
Where T is the pulse width and 1/T is the pulse amplitude and
1
3
(1)
1
3
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:
Figure (1):
Prepared by- Sohail Akbar 10
EC T62 - DIGITAL COMMUNICATION
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.
Figure-1.
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.
Prepared by- Sohail Akbar 15
EC T62 - DIGITAL COMMUNICATION
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.
Companding Characteristics:
1) µ-Law characteristics:
2) A-Law characteristics:
Used mainly in europe. It is defined as-
error
Now, the peak power of the analog signal can be expressed as-
Therefore,
A schematic diagram for the basic DPCM modulator is shown in Figure below.
In this an error sample ep(kTs) is fed.
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,
Now,
Now,
From the close loop including the delay-element in the accumulation unit in the Delta
modulator structure, we can write-
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.
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
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,
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.
Let,
The result is plotted in figure below: Part (a) is for LDM and Part (b) is for ADM.
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.
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
Derivation:
For binary channel, the transmitted signal over symbol interval (0, T) is
represented by-
(2)
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).
Let,
So that,
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-
(B)
(C)
(D)
(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.
Where, Sk + n and Sj + n are the resultant vector around cloud points Sk and Sj.
For, Kj=1
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-
Then, the detector detects that s1(t) was sent otherwise s2(t) was sent.
Let, the energy difference signal is-
Since s2(t)=0,
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-
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).
Where,
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.