MODULE – 4

Digital Representation of Analog Signals

7.1 Introduction:
• The amplitude and angle forms of continuous wave (CW)
modulation were developed in early 1900s with analog source
usually voice.
• With the understanding of digital transmission, the transmission
of digital transmission over analog transmission came into
recognition.
• The 1st step in the evolution from analog to digital transmission is
the conversion of information sources like, voice and music which
are analog to digital.
• The 1st step here is the analog source is sampled at discrete times.
The resulting analog samples are then transmitted by means of
analog pulse modulation.
• The 2nd step is that the analog source is not only sampled at
discrete times but the samples are also quantized to discrete
levels.
• Two methods are used to digitally represent the analog source:
Pulse code Modulation and Delta Modulation.
7.2 Why Digitize Analog Sources?
The advantages of digital signal over analog:
• Digital systems are less sensitive to noise. For long transmission
lengths, the signal may be regenerated effectively error-free at
different points along the path and the original signal
transmitted over the remaining length.
• With digital systems it is easier to integrate different services,
for example, video and the accompanying soundtrack, into the
same transmission scheme.
• The transmission scheme can be relatively independent of
source . For example a digital transmission scheme that
transmits voice at 10 kbps could also be used to transmit
computer data at 10Kbps.
• Circuitry for handling digital signals is easier to repeat and
digital circuits are less sensitive to physical effects such as
vibration and temperature.
• Digital signals are simpler to characterize any typically do not
have the same amplitude range and variability as analog signals.
This makes the associated hardware easy to design.
While almost all media (eg: Cable, radio waves, optical
fiber) may be used for either analog or digital transmission,
digital techniques offer strategies for more efficient use of
those media:
• Various media sharing strategies, known as multiplexing
techniques, are more easily implemented with digital
transmission strategies.
• There are techniques for removing redundancy from a digital
transmission, so as to minimize the amount of information that
has to be transmitted. These technique come under the broad
classification of source coding.
• There are techniques for added controlled redundancy to a digital
transmission, such that the errors that occur during transmission
may be corrected at the receiver without any additional information.
These technique come under the general category of channel
coding.
• Digital techniques makes it easier to specify complex standards that
may be shared on a worldwide basis. This allows the development
of communication components with many different features (eg: a
cellular handset) and their interoperation with a different component
(eg: a base station) produced by a different manufacturer.
• Other channel compensation techniques, such as equalization,
especially adaptive versions are easier to implement with digital
transmission techniques.
• The majority of the advantages for digital transmission rely on the
availability of low cost microelectronics. This counterbalance the
advantage for analog transmission for transporting a large amount
of information in a very simple way.
7.3 Sampling Process:
• The Fourier transform of a periodic signal with the period T0,
consists of an infinite sequence of delta functions, occuring at
the integer multiples of the fundamental frequency f0 = 1/T0.
• From this we may state that making a signal periodic in the time
domain has the effect of sampling the spectrum of the signal in
the frequency domain.
• From the duality property of Fourier transform, sampling in the
time domain has the effect of making the spectrum of the signal
periodic in the frequency domain.
• The sampling process is usually described in the time domain.
It is the operation that is basic to digital processing and digital
communication.
• Through use sampling process, an analog signal is converted
into a corresponding sequence of samples that are usually
spaced uniformly in time.
• For such a procedure to have practical utility, these sampling rate
must be properly chosen, such that sequence of samples uniquely
defines the original analog signal.
• Consider an arbitrary signal g(t) of finite energy, which is
specified for all times.
• A segment of the signal signal g(t) is as shown in the figure 7.1a.
• Suppose we sample a signal g(t) instantaneously and at a uniform
rate, once every Ts seconds.
• We obtain a infinite sequence of samples spaced Ts seconds apart
and denoted by {g(nTs)}, where, n takes on all possible integer
values.
• Ts is referred as sampling period and its reciprocal fs = 1/Ts as the
sampling rate.
• This ideal form of sampling is called instantaneous sampling.
• Let g (t) denote the signal obtained by individually sampling
weighting the elements of a periodic sequence of direct delta
functions spaced Ts seconds apart by the sequence of numbers
{g(nTs)}, as shown in figure 7.1b.

(7.1)

• g (t) is referred as the ideal sampled signal.

• The term (t-nTs) represents the delta function positioned at
time t = nTs.
• The multiplying factor g(nTs) in equation 7.1 is referred as a
“mass” assigned to the delta function (t-nTs).
• A delta function weighted in this manner is a closely
approximated by a rectangular pulse of duration t and
amplitude (nTs)/t.
• Smaller the value of t, better will be the approximation.
• The ideal sampled signal g (t) has a mathematical form similar
to that of the Fourier transform of a periodic signal.
• This is established by comparing the equation 7.1 for g (t) with
the Fourier transform of a periodic signal given in equation 2.88.

• By Using the fact that a delta function is an even function of

time, we get,

• Where G(f) is the Fourier Transform of the original signal g(t)

and fs is the sampling rate.
• Equation 7.2 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.
• The Fourier transform of the ideal sampled signal g (t) can be
obtained by taking the Fourier transform on both sides of
equation 7.1 and noting that the Fourier transform of the delta
function is (t-nTs) equal to exp(-j2 fTs).
• Let G (f) denote the Fourier transform of g (t). Therefore,

• This relation is called discrete time Fourier transform.

• It may be viewed as a complex Fourier series representation of
the periodic frequency function G (f), with the sequence of
samples {g(nTs)} defining the coefficients of the expansion.
• Consider continuous time signal g(t) of finite energy and
infinite duration.
• Suppose, the signal g(t) is strictly band-limited, with no
frequency components higher than W hertz.
• That is the Fourier transform G(f) of the signal g(t) has the
property that G(f) is zero for | f |>= W, as in figure 7.2a.
• Suppose we choose sampling period Ts = 1/2 W. The
corresponding spectrum, G (f) of the sampled signal g(t) is as
shown in the figure 7.2b.
• Put Ts = 1/2 W in equation(7.3), yeilds,
• Substituting equation 7.4 in 7.6, we get,

• 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 by using the discrete time Fourier transform.
• Because g(t) is related to G(f) by the inverse Fourier transform,
it follows that the signal g(t) is itself uniquely determined by
the sample values g(n/2W) for - <n< .
• The sequence {g(n/2W)} has all the information contained in
g(t).
• Consider the reconstruction of the signal g(t) from the sequence of
sample values [g(n/2W)].
• Substituting equation 7.7 in the formula for the inverse Fourier
transform defining g(t) in terms of G(f), we get
• Eq. 7.9 provides interpolation formula for reconstructing the
original signal g(t) from the sequence of sample values
{g(n/2W)}, with sinc function, sinc(2Wt) playing the role of
interpolation function.
• Any impulse response that plays the same role as sinc(2Wt) is
referred to as a reconstruction filter.
• We may state the sampling Theorem for strictly band limited
signals of finite energy in two equal parts:
1) A band limited signal of finite energy, which only has
frequency components less than W Hertz, is completely
described by specifying the values of the signals at the instants
of time separated by 1/2W seconds.
2) A band-limited signal of finite energy, which only has
frequency components less than W Hertz, may be completely
recovered from a knowledge of its samples taken at the rate of
2W samples per second.
• The sampling rate of 2W samples per second, for a signal
bandwidth of W Hertz, is the Nyquist rate; its reciprocal 1/2W
(measured in seconds) is called Nyquist interval.
• The derivation of the sampling theorem is based on the
assumption that the signal g(t) is strictly band-limited.
• In practice, an information bearing signal is not strictly band
limited, with some degree of under sampling is encountered. As
a consequence some aliasing effect is produced.
• Aliasing refers to the phenomenon of a high frequency
component in the spectrum of the signal taking on the identity
of the lower frequency in the spectrum of its sampled version as
in figure 7.3b.
• Aliased spectrum shown by the solid curve in figure 7.3b
pertains to under sampled version of message signal represented
by the spectrum of figure 7.3a.
• To combat the effects of aliasing, we may use 2 corrective
measures.
Figure 7.3 a) Spectrum of a signal, b) spectrum of an under sampled version of the signal
1. Prior to sampling a low pass pre-alias filter is used to attenuate
those high- frequency components of the signal that are not
essential to the information being conveyed by the signal.
2. The filtered signal is sampled at a rate slightly higher than the
Nyquist rate.
• The use of sampling rate higher than the Nyquist rate also has
a beneficial effect of easing the design of the reconstruction
filter used to recover the original signal from its sampled
version.
• Consider the example of a message signal that has been pre-
alias(low-pass) filtered, resulting in the spectrum shown in
Figure 7.4a.
• The corresponding spectrum of instantaneously sampled
version of the signal is shown in figure 7.4b, assuming the
sample rate higher than the Nyquist rate.
• According to figure 7.4b, the design of the reconstruction filter
may be specified as follows(figure 7.4c):
❑ The reconstruction filter is low pass with a passband extending
from –W to W, which is itself determined by the pre-alias filter.
❑ The filter has a transition band extending (for positive
frequencies) from W to fs-W, where fs is the sampling rate.
❑ The reconstruction filter has a well defined transition band that
means it is physically recognizable.
Pulse Amplitude Modulation:
• In Pulse Amplitude Modulation (PAM), the amplitudes of
regularly spaced pulses are varied in proportion to the
corresponding sample values of a continuous message signal.
• The pulses can be rectangular form or some other appropriate
shape.
• Pulse amplitude Modulation here is similar to natural sampling
where message signal is multiplied by a train of rectangular
pulses.
• The waveform of the PAM signal is illustrated in the figure 7.5.
• The dashed curve in the figure 7.5 depicts the waveform of the
message signal m(t), the sequence of the amplitude modulated
rectangular pulses shown as solid lines represents the
corresponding PAM signal s(t).
• The two operations involved in the generation of the PAM
signal:
1. Instantaneous sampling of the message signal m(t) every Ts
seconds, where the sampling rate fs= 1/Ts is choosen in
accordance with the sampling theorem.
2. Lengthening the duration of each samples so obtained to some
constant value T.
• In digital circuit technology, these two operations are jointly
referred as “Sample and hold”.
• One important reason for intentionally lengthening the duration
of each sample is to avoid the use of an excessive channel
bandwidth. Since bandwidth is inversely proportional to pulse
duration.
• Let s(t) denote the sequence of flat-top pulses generated in the
manner described in figure 7.5.
• The PAM signal is expressed as,

• Where, Ts is the sampling period and m(nTs) is the sample value

of m(t) obtained at time t=nTs.
• The h(t) is a standard rectangular pulse of unit amplitude and
duration T, defined as, (figure 7.6a)
• The instantaneous sampled version of m(t) and pulse h(t) as
shown by,
s(t) = m(t) * h(t) (7.15)
• Taking Fourier transform of both sides eq. (7.15) and
recognizing that convolution of two time functions is
transformed into multiplication of their respective Fourier
transforms, we get
S(f) = M(f) H(f) (7.16)
• Where, S(f) = F[s(t)], M(f) = F[m(t)], H(f) = F[h(t)] is as
shown in Figure 7.7a, b,c.
• From eq. 7.2 the Fourier Transform, M(f) is related to the
Fourier Transform, M(f) of the original message signal m(t) as
follows:
• Where fs =1/Ts is the sampling rate.
• Substituting equation 7.17 in equation 7.16, we get

• Given PAM signal s(t) whose Fourier transform is S(f) is

defined as eq.7.18.
• In order to recover the original message signal m(t): The first
step in the reconstruction is passing s(t) through low pass filter
whose frequency response.
• The message signal is limited to bandwidth W and the sampling
rate fs is larger than the Nyquist rate 2W.
• From equation 7.18, we find that the spectrum of the resulting
filter output is equal to M(f) H(f).
• This output is equivalent to passing the original message signal
m(t) through another low pass filter of transfer function H(f) as
in Figure 7.7d.
• From eq (7.11) the Fourier transform of the rectangular pulse
h(t) is given by

• By using Flat top samples to generate a PAM signal, amplitude

distortion as well as delay of T/2 is introduced.
• This effect is rather similar to the variation in transmission with
frequency that is caused by the finite size of the scanning
aperture in television.
• The distortion caused by the use of pulse-amplitude modulation
to transmit an analog information bearing signal is referred as
aperture effect.
• This distortion may be corrected an equalizer in cascade with
the low pass reconstruction filter, as shown in the figure 7.8.
• The equalizer has the effect of decreasing the in-band loss of the
reconstruction filter as the frequency increases in such a manner
as to compensate for the aperture effect. The amplitude response
of the equalizer is given by,

• as illustrated in figure 7.7e.

• The noise performance of a PAM system can never be better
than baseband signal transmission.
• For transmission for long distances, PAM would be used only
as a means of message processing for time- division
multiplexing, from which conversion to come other form of
pulse modulation is made.
Time-Division Multiplexing:
• The sampling theorem forms the basis for transmitting the
information contained in a band-limited message signal m(t) as
a sequence of samples of m(t) taken uniformly at a rate that is
usually higher than the Nyquist rate.
• Important feature of sampling process is conservation of time.
• The transmission of message samples only engages the
communication channel for only a fraction of the sampling
interval on a periodic basis, and in this way some of the time
interval between adjacent samples is cleared for use by other
independent message sources on a time-shared basis.
• Hence we obtain a TDM (Time Division Multiplex) system,
which enables the joint utilization of common communication
channel by a independent message sources without mutual
interference among them.
• TDM is as shown in figure 7.9.
• Each input message signal is first restricted in bandwidth by a
low-pass pre-alias filter to move the frequencies that are
nonessential to an adequate signal representation.
• The low-pass filter outputs are applied to commutator, which is
implemented using a electronic switching circuitary.
• The functions of the commutator is two folds:
1) To take narrow sample of each of the N input messages at a rate
of fs that is slightly higher than 2W, where W is the cutoff
frequency of the pre-alias filter.
2) To sequentially interleave these N samples inside the sampling
interval Ts
• The multiplexed signal is applied to the pulse modulator, the
purpose of which is to transform the multiplexed signal into a
form suitable for transmission over the common channel.
• The use of TDM introduces a bandwidth expansion factor N,
because the scheme must squeeze the N samples derived from
N independent message sources into a time slot equal to one
sampling interval.
• At the receiving end of the system, the received signal is applied
to pulse demodulator, which performs the reverse operation of
pulse modulator.
• The narrow samples produced at the pulse demodulator output
are distributed to the appropriate low-pass reconstruction filters
by means of a decommutator, which operates in synchronism
with the commutator in the transmitter.
• Thus synchronization is essential for a satisfactory operation of
the system.
• The way this synchronization is implemented depends naturally
on the method of pulse modulation used to transmit the
multiplexed sequence samples.
Pulse Position modulation:
• In Pulse Position modulation, we may use the increased
bandwidth consumed by pulses to obtain an improvement in
noise performance by representing the sample values of the
message signal by some property of the pulse other than
amplitude.
• In Pulse Duration Modulation (PDM), the samples of the
message signals are used to vary the duration of individual
pulses. This modulation is also called as Pulse width
Modulation and Pulse Length Modulation.
• The modulating signal may vary the time of occurrence of the
leading edge, trailing edge, or both edges of the pulse.
• In Figure 7.10c the trailing edge of each pulse is varied in
accordance with the message signal assumed to be sinusoidal as
in figure 7.10a.
• The periodic pulse carrier is shown in figure 7.10b.
• In PDM, long pulse expend (spend)considerable power during
the pulse while bearing no additional information.
• If this unused power is subtracted from PDM, so that only time
transitions are preserved, we obtain a more efficient type of
pulse modulation known as pulse position modulation (PPM).
• In PPM, the position of a pulse relative to its unmodulated time
of occurrence is varied in accordance with the message signal,
as shown in figure 7.10d for the case of sinusoidal modulation.
• Let Ts denote the sample duration.
• Using the sample m(nTs) of a message signal m(t) to modulate
the position of the nth pulse, we obtain the PPM signal
• Which kp is the sensitivity of the pulse position modulator and
g(t) denotes standard pulse of interest.
• The different pulse constituting the PPM signal s(t) must be
strictly non-overlapping.
• A sufficient condition for this requirement to be satisfied is to
have,

• which in turn requires that,

• Assuming equation 7.21 is satisfied and there is no interference
between adjacent pulses of the PPM signal s(t), then the signal
samples m(nTs) can be recovered perfectly.
• If the message signal m(t) is strictly band-limited, it follows
from the sampling theorem that the original message signal m(t)
can be recovered from the PPM signal s(t) without distortion.
Generation of PPM Wave:
• The PPM signal described by equation 7.20 may be generated
using the system described in figure 7.11.
• The message signal m(t) is first converted into PAM signal by
means of a sample and hold circuit, generating a staircase
waveform u(t).
• The pulse duration T of the sample and hold circuit is the same
as the sampling duration Ts.
• The operation is illustrated in figure 7.12b for the message
signal m(t) shown in figure 7.12a.
• The signal u(t) is added to a sawtooth wave fig 7.12c, yielding
the combined signal v(t) shown in figure 7.12d.
• The combined signal v(t) is applied to the threshold detector that
produces a very narrow pulse each time v(t) crosses zero in
negative going direction.
• The resulting sequence of impulses i(t) is shown in figure 7.12e.
• The PPM signal s(t) is generated by using this sequence of
impulses to excite a filter whose impulse response is defined by
the standard pulse g(t).
Detection of PPM Waves:
• Consider a PPM wave s(t) with uniform sampling as defined in
equation 7.20 and 7.21.
• Assume that the message m(t) is strictly band-limited.
• The operation of one type of PPM receiver may proceed as
follows:
➢ Convert the received PPM wave into a PDM wave with the
same modulation.
➢ Integrate this PDM wave using a device with finite integration
time, thereby computing the area under each pulse of the PDM
wave.
➢ Sample the output of the integrator at a uniform rate to produce
PAM wave, whose pulse amplitudes are proportional to the
signal samples m(nTs) of the original PPM wave s(t).
➢ Finally demodulate the PAM wave to recover the message
signal m(t).
• All the operations here are linear.
• A practical PPM receiver includes a non linear device called a
slicer at its input end.
• The input output characteristics of an ideal slicer is as shown in
the figure 7.13, where slicing level is normally set at appropriate
half the peak pulse amplitude of the received PPM wave.
• The function of the slicer is to preserve the positions of the
edges of the received signal and remove everything else.
• It does so by producing almost rectangular pulse with fairly
sharp leading and trailing edges at the same instants as the
corresponding edges of the received pulses.
• Thus in a loose sense, the slicer acts as a noise cleaning device
in that final noise level at the output of the receiver is greatly
reduced by eliminating all the noise in the received PPM wave
except in the neighborhood of the leading and trailing edges.
• The output of the slicer is differentiated and then half wave
rectified yielding a very short pulse each time the amplitude of
a pulse in the received PPM wave passes through the slicing
level.
• Figure 7.14a shows the nth pulse of a PPM wave and Figure
7.14b shows the short pulse produced by the slicing device.
• In figure 7.14c an appropriate delay is applied to the short pulse,
and the corresponding PDM pulse is shown in figure 7.14d.
• Having converted the received (noisy) PPM wave into a PDM
wave with the same modulation, the receiver then proceeds to
reconstruct the original baseband signal m(t) in the manner
described above.
Noise in Pulse-Position Modulation: (Self study)
• In a PPM system, the transmitted information is contained in the
relative positions of the modulated pulses.
• The presence of additive noise affects the performance of such
a system by falsifying the time at which the modulated pulses
are judges to occur.
• Immunity to the noise can be established by making the pulse
build up so rapidly that the time interval during which noise can
exert any perturbation is very short.
• Additive noise will have no effect on the pulse positions if the
received pulses were perfectly rectangular, because the
presence of noise introduces only vertical perturbations.
• The reception of perfectly rectangular pulses implies an
infinite channel bandwidth, which is of course impractical.
• With the finite channel bandwidth in practice, the received
pulses have a finite rise time and so the performance of the PPM
receiver is affected by noise.
• As with a CW modulation system, the noise performance of a
PPM system may be described interms of the output signal to
noise ratio.
• To find the noise improvement produced by PPM over baseband
transmission of a message signal, we may use the figure of merit
defined as the output signal to noise ratio of the PPM system
divide by the channel signal to noise ratio.
• The average power of the additive noise at the front end of the
receiver is small compared with the peak pulse power.
• It is assumed that there are two crossings of the slicing level for
each pulse, one for the leading edge and one for the trailing
edge.
• A Gaussian noise will have occasional peaks that produce
additional crossings of the slicing level, however the occasional
noise peaks are mistaken for message pulses.
• The analysis neglects the false pulses produced by the high
noise peaks.
• It is apparent that these high pulses have high a finite though
small probability of occurrence when compared with the peak
amplitude of the pulses.
• As transmission bandwidth increased indefinitely, the
accompanying increase in average noise power eventually
causes the false pulses to occur often enough, thereby causing
loss of the wanted message signal at the receiver output.
• The PPM system suffers from the threshold effect.
Bandwidth – Noise Trade-off (Self Study)
• In context of noise performance, a PPM system represents the
optimum form of analog pulse modulation.
• The noise analysis of a PPM system presented in the preceding
The Quantization Process:
• A continuous signal such as voice has a continuous range of
amplitudes and therefore its samples have a continuous amplitude
range.
• Within the finite amplitude range of signal, we find an finite number
of amplitude levels.
• It is not necessary to transmit the exact amplitudes of the samples.
• Any human sense (eyes, ears) as ultimate receiver can detect only
finite intensity differences.
• This means that the original continuous signal may be approximated
by a signal constructed of discrete amplitudes selected on the basis of
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 the original continuous signal.
• Amplitude quantization is defined as the process of transforming the
sample amplitude m(nTs) of a message signal m(t) at time t=nTs into
v
a discrete amplitude (nTs) taken from a finite set of possible
amplitudes.
• Here the quantization process is memoryless and instanteneous which
means that the transformation at time t=nTs is not affected by the
earlier or latter samples of the message signal. This form of
quantization is not optimum but it is commonly used one.
• When dealing with a memoryless quantizer, we may simplify the
notation by dropping the time index.
• We may use symbol m in place of m(nTs) as shown in figure 7.20a.
• As shown in figure 7.20b, the signal amplitude m is specified by the
index k if it lies inside the interval.
• (7.32)
• Where L is the total number of amplitude levels used in the
quantizer.
• The amplitude mk, k=1,2,3, . . . . . , L are called decision levels or
decision thresholds.
• At the quantizer output the index k is transformed into an amplitude
vk that represents all amplitudes of the interval Ik; the amplitudes vk,
k= 1,2,3,……..L are called representation levels or reconstruction
levels.
• The spacing between two adjacent representation levels is called a
quantum or step-size.
• The quantizer output v equals vk if the input signal sample m
belongs to the interval Ik. The mapping (figure 7.20a)

• is the quantizer characteristic, which is a staircase function by

definition.
• Quantizer can be of a uniform or non uniform type.
• In uniform quantizer, the representation levels are uniformly spaced,
otherwise quantizer non-uniform.
• The quantizer characteristics can be of midtread or midrise type.
• Figure 7.21b shows the input-output characteristic of a uniform
quantizer of the midrise type in which the origin lies in the middle
of a rising part of the staircase like graph.
• The midtread and midrise types of uniform quantizers illustrated in
Figure 7.21 are symmetric about the origin.
Quantization noise:
• The use of quantization introduces an error defined as difference
between the input signal m and the output signal v. this error is
called quantization noise.
• Figure 7.22 illustrates a typical variation of the quantization
noise as a function of time, assuming the use of a uniform
quantizer of the midtread type.
• Let the quantizer input m be the sample value of a zero-mean
random variable M.
• A quantizer g(.) maps the input random variable M of
continuous amplitude into discrete random variable V.
• The sample value m and v are related by equation (7.33).
• Let the quantization error be denoted by the random variable Q
of sample value q. thus
q=m–v (7.34)
Q=M–V (7.35)
• With the input M having zero mean, and the quantizer assumed
to be symmetric as in figure 7.21, it follows that the quantizer
output V and therefore the quantization error Q, will also have
zero mean.
• A partial statistical characterization of the quantizer in terms of
output signal to noise ratio, the mean-square value of the
quantization error Q, need to be found.
• Consider then an input m of continuous amplitude in the range
(-mmax, mmax).
• Assuming a uniform quantizer of the midrise type illustrated in
figure 7.21b, the step size of the quantizer is given by,
 = 2 mmax/L (7.36)
• Where, L is the total number of representation levels.
• For a uniform quantizer, the quantization error Q will have its
sample values bounded by -/2 <= q <= /2.
• If the step size  is sufficiently small (i.e representation levels
L is sufficiently large), it is reasonable to assume that
quantization error Q is uniformly distributed random variable,
and the interfering effect of the quantization noise on the
quantizer input is similar to that of thermal noise.
• The Probability density function of the quantization error Q is
as follows,

• For this to be true, we must ensure that the incoming signal does
not overload the quantizer.
• The mean of the quantization error being zero, its variance 2Q
error is the same as the mean square value
• Typically, The L-ary number k denoting the kth representation
level of the quantizer, is transmitted to the receiver in binary
form.
• Let R denote the number of bits per sample used in the
construction of the binary code. Then,
• Equation (7.44) shows that the output signal to noise ratio of the
quantizer increases exponentially with increasing number of
bits per sample, R.
• An increase in R requires a proportionate increase in the channel
bandwidth BT. Thus we see that the use of binary code for the
representation of a message signal provides more efficient
method than either frequency modulation (FM) or pulse
position modulation (PPM) for the trade-off of increased
channel bandwidth for the improved noise performance.
• We presume that FM and PPM systems are limited by receiver
noise.
7.9 Pulse Code Modulation (PCM):
• In Pulse Code Modulation (PCM) a message signal is
represented by a sequence of coded pulses which is
accomplished by representing the signal in discrete form in both
time and amplitude.
• The basic operations performed at the transmitter of the PCM
system are sampling, quantizing and encoding as shown in
figure 7.23a.
• The low pass filter prior to sampling is included to prevent the
aliasing of the message signal.
• The quantizing and encoding operations are usually performed
in same circuit, which is called an analog to digital converter.
• The basic operation in the receiver are regeneration of impaired
signals, decoding and reconstruction of the train of quantized
samples shown in figure 7.23c.
• Regeneration occurs at the intermediate points along the
transmission path as necessary, as indicated in figure 7.23b.
• When the time division multiplexing is used, it becomes
necessary to synchronize the receiver to the transmitter for the
overall system to operate satisfactorily.
Sampling:
• The incoming message signal is sampled with the train of
narrow rectangular pulses so as to closely approximate the
instantaneous sampling process.
• To ensure perfect reconstruction of the message signal at the
receiver, the sampling rate must be greater than twice the
highest frequency component W of the message signal in
accordance with the sampling theorem.
• In practice, a pre-alias(low-pass) filter is used at the front end
of the sampler in order to exclude frequencies greater than W
before sampling.
• The application of sampling permits the reduction of the
continuously varying message signal to a limited number of
discrete values per second.
Quantization:
• The sampled version of the message signal is then quantized,
thereby providing a new representation of the signal that is
discrete in both time and amplitude.
• It is preferable to use a variable separation between the
representation levels.
• For eg, the range of voltages covered by voice signals, from the
peaks of loud talk to the weak passages of weak talk, is on the
order of 1000 to 1.
• By using nonuniform quantizer with the feature that the step
size increases as the separation from the origin of the input-
output amplitude characteristic is increased, the large end step
of the quantizer can take care of the possible excursions of the
voice signal into the large amplitude ranges that occur relatively
infrequently.
• The weak passage need more protection and favored at the
expense of the loud passages.
• In his way, a nearly uniform percentage precision is achieved
throughout the greater part of the amplitude range of the input
signal, with the result that fewer steps are needed than would be
the case if a uniform quantizer were used.
• The use of nonuniform quantizer is equivalent to passing the
baseband signal through a 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 defined by
• Where, m and v are the normalized input and output voltages
and µ is a positive constant.
• In figure 7.24a, we have plotted the µ-law for varying µ.
• The case of uniform quantization corresponds to µ=0. For a
given value of µ, the reciprocal slope of the compression curve,
which defines the quantum steps, is given by the derivative of
|m| with respect to |v|, that is

(7.48)
• Therefore the µ-law is neither strictly linear nor strictly
logarithmic, but it is approximately linear at low input levels
corresponding to µ|m| << 1, and approximately logarithmis at
high input levels corresponding to µ|m| >> 1.
• Another compression law that is used in practice is the so-
called A-law defined by,

(7.49)
•
• which is as shown in Figure 7.24b.
• Practical values of A tend to be in the vicinity of 100.
• The case of uniform quantization corresponds to A=1.
• The reciprocal slope of this compression curve is given by the
derivative of |m| with respect to |v|, as shown by,

(7.50)
• Thus the quantum steps over the central linear segment, which
have the dominant effect on small signals are diminished by the
factor A/ (1+ log A).
• This is about 25dB in practice, as compared with uniform
quantization.
• In order to restore the signal samples to their correct relative
level, we must use a device in the receiver with the
characteristic complementary to the compressor. Such a device
is called an expander.
• The compression and expansion laws are exactly inverse, so that
except for the effect of quantization, the expander output is
equal to the compressor input. The combination of expander and
compressor is called compander.
• In actual PCM systems, the companding circuitary does not
produce an exact replica of the nonlinear compression curves
shown in figure 7.24.
• It provides a piecewise linear approximation to the desired
curve.
• By using a large enough numbers of linear segments, the
approximation can approach the true compression curve very
closely.
Encoding:
• In combining the process of sampling and quantizing, the
specification of a continuous message signal becomes limited to
a discrete set of values, but not in the form suited for
transmission over a line or radio path.
• To exploit the advantages of sampling and quantizing for the
purpose of making the transmitted signal more robust to noise,
interference and other channel degradation.
• An encoding process is required to translate the discrete set of
sample values to a more appropriate form of signal.
• The plan of representing each of this discrete set of values as a
particular arrangement of discrete events is called a code.
• One of the discrete events in a code is called a code element or
symbol.
• For example, the presence or absence of a pulse is a symbol.
• A particular arrangement of symbols used in a code to represent
a single value of the discrete set is called a code word or
character.
• In a binary code, each symbol may be either of two distinct
values or kinds, such as presence or absence of a pulse.
• The two symbols of a binary code are denoted as 0 and 1.
• In ternary code, each symbol may be one of the three distinct
values or kinds.
• The maximum advantage over the effects of noise in a
transmission medium is obtained by using a binary code,
because a binary symbol withstands a relatively high level of
noise and is easy to regenerate.
• Suppose that in a binary code, each code word consists of R
bits.
• R represents the number of bits per sample.
• Using such code, we may represent a total 2R distinct numbers.
• For eg: A sample quantized into one of 256 levels may be
represented by an 8 bit code word.
• There are several ways of establishing a one-to-one
correspondence between representation levels and codewords.
• A convenient method is to express the ordinal number of the
representation level as binary number.
• In binary number system each digit has a place-value that is a
power of 2 as shown in table 7.2, for the case of four bits per
sample.(i.e, R=4).
Line Codes:
• It is in a line code that a binary stream of data takes on an
electrical representation.
• Any one of the several line codes can be used for the electrical
representation of a binary data stream.
• Figure 7.25 displays the waveforms of the five important line
codes for example data stream 01101001. The line codes often
used the terminology Nonreturn- to-Zero (NRZ) or Return-to-
Zero (RZ).
• Return to zero implies that the pulse shape used to represent the
bit always returns to the 0 volts or the neutral level before the
end of the bit .
• Nonreturn-to-zero indicates that the pulse does not necessarily
return to the neutral level before the end of the bit.
1. Unipolar NonReturn-to-Zero (NRZ) signaling:
• Here symbol 1 is represented by transmitting a pulse of
amplitude A for the duration of the symbol, and symbol 0 is
represented by switching off the pulse as in figure 7.25a.
• This line code is referred as on-off signaling.
• A disadvantage of on-off signaling is the waste of power due
to the transmitted DC level.
2. Polar NonReturn-to-Zero (NRZ) signaling:
• Here symbol 1 and 0 is represented by transmitting a pulses of
amplitude +A and –A as in figure 7.25b.
• This line is relatively easy to generate and is more power-
efficient than its unipolar counterpart.
3. Unipolar Return-to-Zero (RZ) signaling:
• Here symbol 1 is represented by rectangular pulse of amplitude
A and half-symbol width, and symbol 0 is represented by
transmitting no pulse as in figure 7.25c.
• An attractive feature of this line code is the presence of delta
functions at f= 0, + or – 1/Tb in the power spectrum of the
transmitted signal, which can be used for bit-timing recovery at
the receiver.
• Its disadvantage is that it requires 3 dB more power than polar
return-to-zero signaling for the same probability of the symbol
error.
4. Bipolar Return-to-Zero (BRZ) signaling:
• Here line codes uses three amplitude levels as shown in figure
7.25d.
• Positive and negative pulses of equal amplitude (i.e, +A and –
A) are used alternately for symbol 1, with each pulse having a
half-symbol width. No pulse is always used for symbol 0.
• A useful property of the BRZ signaling is that the power
spectrum of the transmitted signal has no DC component and
relatively insignificant low-frequency components for the case
when symbol 1 and 0 occur with equal probability. This line
code is also called alternate mark inversion (AMI) signalling.
5. Split-Phase (Manchester Code):
• Here symbol 1 is represented by a positive pulse of amplitude
A followed by a negative pulse of amplitude- A, with both
pulses being a half- symbol wide as shown in figure 7.25e.
• For symbol 0, the polarities of these two pulses are reversed.
• The Manchester code suppresses the DC component and has
relatively insignificant low-frequency components, regardless
of signal statistics. This property is essential in some
applications.
Differential Encoding:
• This method is used to encode information interms of signal
transitions.
• A transition is used to designate symbol 0 in the incoming
binary data while no transmission is used to designate symbol 1
as illustrated in figure 7.26.
• In figure 7.26b, we show differentially encoded data stream for
the example data specified in figure 7.26a.
• The original stream data stream used here is the same as that
used in figure 7.25.
• The waveform of differentially encoded data is shown in figure
7.26c, assuming the use of unipolar return to zero signaling.
• From figure 7.26, it is apparent that a differentially encoded
signal may be inverted without affecting its interpretation.
• The original binary information is recovered simply by
comparing the polarity of adjacent binary symbols to establish
whether or not a transition has occurred.
• The differential encoding requires the use of a reference bit
before initiating the encoding process.
• In Figure 7.26, symbol 1 is used as the reference bit.
Regenerative Repeater:
• The most important feature of any digital system lies in the ability
to control the effects of distortion and noise produced by
transmitting a digital signal through channel.
• This capability is accomplished by reconstructing the signal by
means of a chain of regenerative repeater located at a sufficiently
close spacing along the transmission route.
• A shown in figure 7.27, three basic functions are performed by
the regenerative repeater: Equalization, timing and decision
making.
• The equalizer shapes the received pulses so as to compensate for
the effects of amplitude and phase distortions produced by the
transmission characteristics of the channel.
• The timing circuitry provides a periodic pulse train, derived from
the received pulses, for sampling the equalized pulses at the
instants of time where the signal-to-noise ratio is maximum.
• The sample so extracted is compared to a predetermined
threshold in the decision making device. In each bit interval a
decision is then made whether the received symbol is a 1 or 0
on the basis of whether the threshold is exceeded or not.
• If the threshold is exceeded, a clean new pulse representing
symbol 1 is transmitted to the next repeater.
• Otherwise, a clean new pulse representing symbol 0 is
transmitted.
• In this way, the accumulation of distortion and noise in a
repeater span is completely removed, provided that the
disturbance is not too large to cause an error in the decision-
making process.
• Ideally except for delay, the regenerated signal is exactly the
same as the signal originally transmitted.
• In practice, the regenerated signal departs from the original
signal for two main reasons:
• 1. The unavoidable presence of channel noise and interference
causes the repeater to make wrong decisions occasionally,
thereby introducing bit errors into the regenerated signal.
• 2. If the spacing between received pulses deviates from its
assigned value, a jitter is introduced into the regenerated pulse
position, thereby causing distortion.
Decoding:
• The first operation in the receiver is to regenerate (i.e., reshape
and clean up) the received pulses one last time.
• These clean pulses are then regrouped into code words and
decoded (i.e, mapped back) into a quantized PAM signal.
• The decoding process involves generating a pulse the amplitude
of which is the linear sum of all the pulses in the code word,
with each pulse being weighted by its place value (20, 21, 22, 23,
. . . . . . . . . . , 2R-1) in the code, where R is the number of bits
per sample.
Filtering:
• The final operation in the receiver is to recover the message
signal wave by passing the decoder output through a low-pass
reconstruction filter whose cut-off frequency is equal to the
message bandwidth W.
• Assuming that the transmission path is error free, the recovered
signal includes no noise with the exception of the initial
distortion introduced by the quantization process.
Multiplexing:
• In application using PCM, it is natural to multiplex different
message sources by time division, whereby each source keeps
it individuality throughout the journey from the transmitter to
the receiver .
• This individuality accounts for the comparative ease with which
message sources may be dropped or reinserted in a time-
division multiplex system.
• As the number of independent message sources is increased, the
time interval that may be allotted to each source has to be
reduced, since all of them must be accommodated into a time
interval equal to the reciprocal of the sampling rate.
• This inturn means that the allowable duration of a code word
representing a single sample is reduced.
• However, pulses tend to become more difficult to generate and
to transmit as their duration is reduced.
• If the pulses become too short, impairments in the transmission
medium begin to interfere with the proper operation of the
system.
• It is necessary to restrict the number of independent message
sources that can be included within a time-division group.
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