Module 5

DIGITAL REPRESENTATION
OF ANALOG SIGNALS
 Introduction
• Era of CW Modulation and its drawbacks
• Improvement of Digital Transmission from 1930 to
1960
• Process of Analog to digital conversion–
SAMPLING
QUANTIZATION
 Why Digitize Analog Sources
 Digital Signals are less sensitive to NOISE than analog.
For long transmission lengths, the signal may be
regenerated effectively error-free at different points
along the path and the original signal is transmitted
over the remaining length.
 With digital systems it is easier to integrate different
services ( Ex-Video and Audio)
 The transmission scheme is relatively independent of
source.
 Why Digitize Analog Sources
 Circuitry for handling digital signals is easier to repeat
and digital circuits are less sensitive to physical effects
such as vibrations & temperature.
 Simpler to characterize and typically do not have the
same amplitude range and variability as analog signals.
Thus design of associated hardware is easy to design.
 Various media sharing schemes are easy in digital
transmission strategies.
 Why Digitize Analog Sources
 There are techniques for removing redundancy from
digital transmission, so as to minimize the amount to
be transmitted.
 There are techniques for adding controlled redundancy
in the signal in digital transmission.
 Digital techniques make it easier to specify complex
standards that may be shared on worldwide basis.
 The sampling Process
 It is usually described in time domain
 It is an operation that is basic to digital signal
processing and digital communications.
 Through the use of Sampling process the analog signal
is converted into corresponding sequence of samples
that are usually spaced uniformly in time.
 For the same it is necessary that we choose the
sampling rate properly, such that samples uniquely
defines original signal.
The sampling Process
 The sampling Process
 Consider an Arbitrary signal of finite energy, and considering
that we sample it once after Every Ts seconds apart and
denoted by {g(nTs)}, where n takes on all possible integer values.
• Ts is refered as Sampling period and to its reciprocal fs=1/Ts as
the sampling rate.
This ideal form of sampling is called Instantaneous Sampling.
 The sampling Process
 Let gδ(t) denote the signal obtained by individually weighting the
elements of periodic sequence of dirac delta functions spaced as
Ts seconds apart by sequence of numbers {g(nTs)} as shown by,
 gδ(t)= σ∞
𝑛=−∞ 𝑔 𝑛𝑇𝑠 δ(𝑡 − 𝑛𝑇𝑠) ----1
gδ(t) is as the ideal sampled signal
δ(𝑡 − 𝑛𝑇𝑠)represents a delta function positioned at time
t=nTs
 The sampling Process
 The idea sampled signal gδ(t) has mathematical form similar to
that of Fourier transform of periodic signal. This is readily
established by comparing equation 1 with Fourier transform of
periodic signal as mentioned in chapter 2. By applying duality
property,

 gδ(t)= fs σ∞
𝑚=−∞ 𝐺 𝑓 − 𝑚𝑓𝑠 -----------------------2
where G(f) is the fourier transform of ideal sampled signal
f𝑠 represents the sampling rate.
 The sampling Process
 Thus equation 2 states that the process of uniformly sampling a
continuous time signal of finite energy results in a periodic
spectrum with a period equal to the sampling rate.
 Fourier transform of signal gδ(t) can be written in alias form
where δ(𝑡 − 𝑛𝑇𝑠) can be replaced by exp(-j2ΠnfTs).
 Then let Gδ(f) denote the fourier transform of gδ(t), thus we may
write equation 2 as,
Gδ(f) = σ∞
𝑛=−∞ 𝑔 𝑛𝑇𝑠 exp(-j2ΠnfTs) -----3
This is called discrete time fourier transform.
 The sampling Process
 Suppose the signal g(t) is strictly band limited with NO frequency
components higher than W hertz. That is G(f) of it is Zero for
|f| ≥W.
Suppose we choose sampling period Ts=1/2W then equation 3
yields,
Gδ(f) = σ∞
𝑛=−∞ 𝑔 𝑛/2𝑊 exp(-j2Πnf/W) ------4
Also equation 2 can be rewritten as ,
Gδ(f) = fs G(f)+ σ∞
𝑚=−∞ 𝐺 𝑓 − 𝑚𝑓𝑠 ----5
m≠0
The sampling Process
 The sampling Process
 Thus under 2 following conditions,
1. G(f)=0 for |f| ≥W
2. Fs=2W
Then we find from equation 5 that,
G(f)= 1/2W Gδ(f), for -W <f< W -------------6
On substituting equation 4 in 6 we get,
G(f) = 1/2W σ∞
𝑛=−∞ 𝑔 𝑛/2𝑊 exp(-j2Πnf/W) -------7
 The sampling Process
Therefore if the sample values g(n/2W) of a signal g(t) are
specified for all time, then the fourier transform G(f) of the
signal is uniquely determined using the discret time fourier
transform equation 7.
HW- Have an overview of 4 step equations used for
reconstructing the original signal g(t) from sequence of samples
[g(n/2W)].
 The sampling Process
We may now state the sampling theorem for strictly band
limited signals of finite energy into 2 equivalent parts,
1. A band limited signal of finite energy which only has
frequency component less than W hertz, is completely
described by specifying the values of signal at instants of
time separated by 1/2W seconds.
2. A band limited signal of finite energy which only has
frequency component less than W hertz, may be completely
recovered from a knowledge of its samples taken at rate of
2W samples per second.
 The sampling Process
The sampling rate of 2W samples per second for a signal
bandwidth of W Hertz is called NYQUIST RATE.

Its reciprocal 1/2W is called NYQUIST INTERVAL

 PULSE AMPLITUDE MODULATION
 It is most simplest and basic form of Analog Pulse Modulation
 Here the amplitudes of regularly spaced pulses are varied in
proportion to the corresponding sample values of continuous
message signal; the pulses can be of rectangular form or
some other appropriate shape.
PULSE AMPLITUDE MODULATION
 PULSE AMPLITUDE MODULATION
 The dashed line shows the Message signal m(t) and the
sequence of amplitude modulated rectangular pulses shown
as solid lines represent the corresponding PAM signal S(t).
 There are 2 operations involved in generation of PAM signal
 Instantaneous sampling of the message signal m(t) every
Ts seconds, where the sampling rate fs=1/Ts is chosen in
accordance with the sampling theorem
 Lengthening the duration of each sample so obtained to
some instant value T.
 PULSE AMPLITUDE MODULATION
 In digital ckt technology these 02 operations are jointly
referred as “ Sample & Hold “.
 Reason for intentionally lengthening duration of each sample
is to avoid excess channel bandwidth, since Bandwidth is
inversely proportional to pulse duration.
 Further care to be taken on how long the T ( sample duration)
to be taken can be done by analysis as shown,
PULSE AMPLITUDE MODULATION
PULSE AMPLITUDE MODULATION
PULSE AMPLITUDE MODULATION
PULSE AMPLITUDE MODULATION
 PULSE AMPLITUDE MODULATION
There fore substitutions yield the equation for S(f) as,

S(f) = fs σ∞
𝑘=−∞ 𝑀 𝑓 − 𝑘𝑓𝑠 . 𝐻(𝑓)

Recovering m(t) from PAM signal S(t)

PAM Signal S(t) Message

Signal m(t)
RECONSTRUCTION EQUILIZER
FILTER
Some final remarks on PAM-
 The transmission of PAM imposes stringent requirements on
amplitude and phase responses of the channel, because of
relatively short duration of pulses transmitted.
 Noise performance of PAM system can never be better than
baseband transmission.
 For long distance transmission PAM is used only as a means of
message processing for TDM, from which conversion to some
other form of pulse modulation is subsequently made.
TIME DIVISION MULTIPLEXING
 PULSE POSITION MODULATION
In pulse modulation system, we may use the increased bandwidth
consumed by the pulses to obtain an improvement in noise
performance by representing sample values of the message signal
by some property of the pulse other than amplitude.
2 forms of such modulation schemes are,
1. Pulse Duration Modulation [PDM] or Pulse width
Modulation [PWM] or Pulse Length Modulation [PDM]
2. Pulse Position Modulation [PPM]
 Pulse Duration Modulation [PDM]
 In Pulse Duration Modulation [PDM] the samples of the
message signal are used to vary the duration of the individual
pulses.
 Its also referred as PWM or PLM.
 The modulating signal may vary the time of occurrence of
leading edge, the trailing edge or both edges of the pulse.
 In PDM long pulses expend considerable power during the pulse
while bearing NO ADDITIONAL information.
 PPM
If the unused power is subtracted from PDM, so that only time
transitions are preserved We obtain more efficient type of pulse
modulation known as PPM.

In PPM position of the pulse is relative to its unmodulated

time of occurrence is varied in accordance with the message signal.

Let Ts denote sample duration then the PPM Signal can be

expressed as, S(t) = σ∞
𝑛=−∞ 𝑔 𝑡 − 𝑛𝑇𝑠 − 𝐾𝑝𝑚(𝑛𝑇𝑠)

Where Kp is sensitivity of the Pulse Position Modulator and g(t)

denotes standard pulse of Interest.

Strictly the pulses should not overlap.

GENERATION OF PPM WAVE
 DETECTION OF PPM WAVE
The PPM receiver may proceed as follows,
• Convert the received PPM wave into a PDM wave with same
modulation

• Integrate this PDM wave using a device with a finite integration

time, thereby computing the area under each pulse of PDM wave

• Sample the output of Integrator at a uniform rate to produce

PAM wave whose amplitudes are proportional to signal samples
m(nTs) of original PPM wave s(t)

• Finally, demodulate the PAM wave to recover the message signal

m(t)
 THE QUANTIZATION PROCESS
• The original continuous signal may be approximated by a signal
constructed of discrete amplitudes selected on a minimum error
basis from an available set.
• If we assign the discrete amplitude levels with sufficiently close
spacing we may make the approximated signal practically
indistinguishable from original CW signal.
• Amplitude quantization is defined as process of transforming the
sample amplitude m(nTs) of a message signal m(t) at time t=nTs
into discrete amplitudes v(nTs) taken from a finite set of possible
amplitudes.
 THE QUANTIZATION PROCESS
• Assuming the quantization process to be Memory less and
Instantaneous which means the samples are not affected by
earlier or later samples of message signals.
 THE QUANTIZATION PROCESS
• We use symbol ‘m’ in place of m(nTs). Then from figure the signal
amplitude m is specified by the index k if it lies inside the interval,
• Τk: { mk <m< mk+1}, k= 1,2 ,…..L
• Where L is total number of amplitude levels used in quantizer
• The amplitudes mk, k=1,2,3…..L are called decision levels or
decision thresholds
• At quantizer output V equals Vk, k= spacing between 2 adjacent
representation levels is called QUANTUM or STEP SIZE.
 THE QUANTIZATION PROCESS
• The Mapping we see, V= g(m) is the quantizer characteristic
which is a STAIRCASE WAVE by definition.
• Considering only UNIFORM quantizer only, its characteristics can
also be of Midtrade type or Midrise type.
 THE QUANTIZATION PROCESS
• The Mapping we see,

MIDRISE MIDTREAD
THE QUANTIZATION PROCESS
 PULSE CODE MODULATION
 It is most basic form of Digital Pulse Modulation
 Here a message signal is represented by sequence of
coded pulses , which is accomplished by representing
signal in discrete form in both time and amplitude.
 Lets study the Generation and Detection steps of PCM
Signal
 PULSE CODE MODULATION
 Basic Elements of PCM System are demonstrated 3
sections namely, Transmitter, Transmission Path and
Receiver.
PULSE CODE MODULATION
Quantization
• Non uniform quantizer with variable step size used.

• Non uniform quantizer is equivalent to passing a baseband

signal through compressor and then applying the compressed
signal to a uniform quantizer.

• A particular form of compression law that is used in practice

is the so called,

• μ Law

• A Law
 μ Law
• m and v are normalized input and output voltages.
• |v|= Log(1+μ|m|)
Log( 1+μ)
A Law
Encoding
Line Codes

Unipolar NRZ

Polar NRZ

Unipolar RZ

Bipolar RZ

Split Phase (Manchestor Code)

Unipolar NRZ- 1 represented by Amplitude A & 0 by No Pulse

Polar NRZ - 1 represented by Amplitude +A & 0 by Amplitude –A

Unipolar RZ - 1 represented by Amplitude +A of half width &

0 by no pulse

Bipolar RZ- 1 represented by Amplitude +A & -A alternatively of

half width and No pulse for 0.

Split Phase (Manchestor Code)- 1 represented by + ve pulse of

Amplitude A and -ve pulse of amplitude –A with both pulses being
half wide . For 0 the polarities of these 2 pulses are reversed.
 Differential Encoding
In this method , a transition is used to designate symbol 0 in the
incoming binary Data stream while no transition is used to
designate symbol 1
Regeneration
Regeneration
• Achieved through chain of Regenerative repeaters located along
transmission route.
• 3 basic functions performed are Equalization, Timing and
decision making
• Equalizer shapes received pulses so as to shape so as to
compensate effects of amplitude and phase distortions in
transmission lines
• Timing ckt produces provides periodic pulse train, derived from
received pulse for sampling equalized pulses
• The extracted sample is compared with predetermined
THRESHOLD to decide 0 or 1.
In practice if regenerative signal departs from original signal is may
be due to 2 main reasons,
 Unavoidable presence of channel noise and interference causes
repeater to make wrong decision occasionally, there by
introducing bit erros
 If spacing between received pulses deviates from its assigned
value , a jitter is introduced into regenerated pulse position.
DECODING
 In the receiver first operation is to regenerate received pulses
 These clean pulses are regrouped into code words and decoded
into a PAM signal
 It is exactly opposite to that of Encoding
FILTERING
 The final operation in receiver is to recover the message signal by
passing through low pass reconstruction filter whose cut off
frequency is W hertz.
MULTIPLEXING
 In applications of PCM, many message signals are multiplexed
using TIME DIVISION where by each source maintains
individuality through out the journey from transmitter to
receiver.

 VOCODER
• Vocoding is a common term used for process of converting audio to a
digital format Suitable for cellular transmission.
COMPLETE