1]An analog information source can be transformed into a discrete information source through the

process of sampling and quantization.

2]The input to the source encoder is a string of symbols occurring at a rate of rs symbols/sec. The
source encoder converts the symbol sequence into a binary sequence of 0’s and 1’s by assigning
code words to the symbols in the input sequence.

3] The channel encoder accomplishes the task of error control by systematically adding extra bit to
the output of the source coder. These extra bits themselves convey no information. They are used by
the receiver to detect and/or correct some of the errors in the information bearing bits..

Sampling Process : The process of transforming an analog waveform into a discrete waveform is
called as sampling process.

Sampling Theorem :

Statements
1. A band limited signal having no spectral component above ′__′Hz is completely described by
specifying values of signal at the instant of time separated by 1/2__ seconds. This statement is
called as the uniform sampling theorem.
i.e. fs = 1/2fm (sec)__ (1.1)
Where __ is called as the Nyquist interval.
2. A band limited signal of finite energy which has no frequency component higher than ′__′Hz,
may be completely recovered from the knowledge of samples taken at a rate of 2__ samples per
seconds. This is called as the Nyquist criterion.
i.e. fs ≥ 2fm__ (1.2)
The sampling rate fs = 2fm is also called as the Nyquist rate.

Sampling theorem in Time domain

Xdel(t)= ------ + ___ + del(t+Ts) + del(t) + del(t-Ts) + ____ + ___ − ___

Xdel (t)= sum{n=-inf} {inf} del(nTs)

The signal can be reconstructed properly only if it satisfies the following two criterions.
1. The message signal should be strictly bandlimited and 2. Sampling frequency Fs should be
equal to 2Fm.

ALIASING: The Nyquist rate, Fs= 2Fm, is the sampling rate, below
which aliasing occurs. To avoid aliasing, the Nyquist criterion, Fs ≥ 2Fm, must be satisfied
(Nyquist Criterian , Fs>=2Fm).
Antialiasing filters: 1. Pre-filtering
2. Post-filtering

1. Pre-filtering: In this method, the analog signal is pre-filtered so that the new maximum
frequency Fm’ is reduced to Fs/2.
2. Post-filtering: In this method, the filter cutoff frequency Fm” removes the
aliasing components, Fm” needs to be less than (Fs-Fm)

Sampling techniques:
[Link] Sampling : Also called instantaneous sampling
2. Practical Sampling:
(a) Natural Sampling
(b) Flattop Sampling

1. Ideal Sampling:

The working principle of this circuit is quite easy. It consists of a switch. When closing time t of
the switch approaches to zero, the output X∆(t) contains instantaneous value of the input signal
x(t). As the width of the pulse approaches to zero, the ideal sampling gives a train of impulses
of height which is equal to the instantaneous value of the input signal x(t) at the sampling
instant..

(a) Natural Sampling

Multiplying x(t) by x_p(t) [switching waveform].
Xs(t)= X(t)*X_p(t) (1)
in the sequence
Xs(t) has the shape of its corresponding analog segment during the pulse interval.

(b) Flattop sampling

In Flattop sampling, top of the each sample remains constant and is equal to the instantaneous
value of the input signal X(t) at the start of sampling. The duration of width of each sample is τ
and sampling rate is Fs. The simplest and most popular method to achieve flattop sampling is to
use a sample and hold circuit.

Aperture effect:

The amplitude of flat top signal should be

constant but practically it is not appearing so. This is because, the high frequency roll off of
Xs(f) acts as a low pass filter and attenuates the high frequency components. This effect is
called as an aperture effect. Aperture effect becomes more prominent with the increase in the
duration τ of the pulse.

To compensate this effect, an equalizer is used during reconstruction of the signal.

Reconstruction of the signal

The original message signal can be detected from the natural PAM by passing it through a low
pass filter. The low pass filter with cut off frequency equal to Fm removes high frequency ripple
and recovers the original modulating signal. In case of flat top PAM, an equalizer is used to reduce
the aperture effect.

Derivation for reconstruction of signal ( *IMP* )

Quantization:

Quantization is the process of mapping a large set of input values to a countable smaller set.

A device or algorithmic function that performs quantization is called a quantizer.

Step size (del) = [V_h-V_l] / [ M]

The quality of approximation may be improved by reducing the step size, thereby increasing the
number of allowable levels.

Quantization error:
The difference between an instantaneous
value of the input message signal x(t) and its quantized value Xq(t) is referred to as a
quantization error or quantization noise..

Pulse code modulation:

The pulse code modulation technique samples the input signal x(t) at frequency Fs ≥ 2W.
The input signal x(t) is passed through the low pass filter of cut off frequency W Hz, which will
block all the frequencies above W Hz. Thus, x(t) is bandlimited. The sample and hold circuit
then samples this signal at a rate of fS. Sampling frequency fS is selected sufficiently above the
Nyquist rate to avoid aliasing. i.e. (Fs>=2W).
The output of sample and hold circuit is called as x(nTs) which is discrete in time and
continuous in amplitude. A ‘q’ level quantizer compares input x(nTs) with the fixed digital
levels. It assigns only one of the digital levels to x(nTs) which results in minimum distortion or
error. This error is called as the quantization error. Thus, the output of a quantizer is a digital
level called Xq(nTs). The quantized signal level Xq(nTs) is given to a binary encoder. This
encoder converts the input signal level to an ‘n’ digit binary word. The encoder is also called as a
digitizer.
It is not possible to transmit each bit of a binary word separately over the transmission line.
Therefore, ‘n’ binary digits are converted to a serial bit stream by using a parallel to serial
convertor to generate a baseband signal.
Transmission Bandwidth and signaling rate of PCM

PCM receiver:
The regenerator at the receiver reshapes the pulses and removes the noise.
The signal is then applied to the serial to parallel convertor which converts the serial data into the
parallel data to form a digital word. The word is then converted to its equivalent analog value
Xq(t) along with the sample and hold circuit. The output of the sample and hold circuit is applied
to the low pass filter to get the signal Yd(t).

Quantization noise power:

Noise Power = V^2 noise/ R
Signal to noise power ratio of a quantizer increases exponentially with the increase in number of bits
per sample.

Signal to quantization noise ratio for sinusoidal signal:

P= 1.8 +6n (in dB)

Signal to quantization noise ratio for audio signal:

P= 6n (in dB)

Non-uniform quantization:

The quantization error depends on step size. When the steps are uniform in size, the quantization is
called as uniform quantization. Such a system would
be wasteful for speech signal as many of the quantizing steps would rarely be used. In a system
that uses uniform quantization, the quantization noise is same for all signal levels. Therefore,
with uniform quantization, the signal to noise ratio (SNR) is worse for low level signal than for
high level signals.
Non-uniform quantization can provide fine quantization for the weak signals and course
quantization for the strong signals. Thus, in case of non-uniform quantization, quantization noise
can be made proportional to the signal size. The effect is to improve the overall SNR by reducing
noise for the predominant weak signals.

Companding characteristics: (Do it from PDF):

Differential Pulse Code Modulation:

In the use of PCM for digitization of voice or video signal, the signal is sampled at a rate slightly
higher than the Nyquist rate. The resulting sampled signal then exhibits a high correlation
between adjacent samples. The meaning of this high correlation is that, the signal does not change
rapidly from one sample to the next. When these highly correlated samples are encoded
as in the standard PCM system, the resulting encoded signal contains redundant information. By
removing this redundancy before encoding, the overall bit rate will be decreased and the number
of bits required to transmit one sample will also be reduced. This type of digital pulse
modulation scheme is called as Differential Pulse Code Modulation (DPCM).

DPCM Transmitter: The differential pulse code modulation works on the principle of prediction.
The value of the present sample can be predicted from the past samples. (Do it from PDF):

DPCM receiver:
The decoder first reconstructs
the quantized error signal from the incoming binary signal. The prediction filter output and the
quantized error signal are added to give the quantized version of the output signal. Thus, the
signal at the receiver differs from the actual signal by quantization error q(nTs) which is
introduced permanently in the reconstructed signal.

Delta Modulation:
In Pulse Code Modulation technique, the signaling rate increases if the number of bits per sample
is increased. This also increases the transmission bandwidth. To overcome this drawback, a Delta
Modulation technique is introduced. Delta modulation technique transmits only one bit per
sample.
In Delta modulation technique the input signal x(t) is approximated to a step signal with fixed
step size. This is done by comparing the present sample value with the previous one. The difference
between the input signal and the staircase approximated signal is confined to two
levels i.e. +δ or –δ. If the difference is positive, the approximated signal is increased by one step
i.e. δ. If the difference is negative, the approximated signal is decreased by one step i.e. −δ. If the
step is increased, bit ‘1’ is transmitted and if the step is decreased, bit ‘0’ is transmitted. Thus, for
each sample only one bit is transmitted.

Delta Modulation transmitter

Delta Modulation receiver

Advantages of Delta Modulation technique: 1. Delta Modulator transmits only one bit per sample.
Thus, the signaling rate and transmission bandwidth is quite small as compared to the PCM
technique.
2. The implementation of transmitter and receiver is very simple. There is no need to use the A/D
Converter
Disadvantages of Delta Modulation technique:
1. Slope overload distortion
2. Granular Noise

Adaptive Delta Modulation: To overcome the drawback of slope overload distortion and granular
noise in Delta Modulation,
the step size δ is to be made adaptive to the variations in input signal x(t). Particularly, when the
slope of the input signal x(t) is large, the step size is to be increased and when the input signal is
varying slowly, the step size is to be decreased. The modulation technique developed to achieve
the adaptive step size is called as Adaptive Delta Modulation.
Adaptive Delta Modulation transmitter
Adaptive Delta Modulation receiver