5.

3 Comparison of Analog and Digital communications:

2. Distinguish between Analog and Digital communications.

Sl. No. Analog Communication Digital Communication

1. Transmitted signal is analog in nature Transmitted signal is analog or digital in
nature
2. Amplitude, frequency or phase variations in Amplitude, width or position of the
the transmitted signal represent the transmitted pulses is constant. The
information or message. information or message is transmitted I the
form of code words.
3. Noise immunity is poor. Noise immunity is excellent.
4. It is not possible to separate the noise and It is possible to separate the signal from the
signal. noise.
5. Coding is not possible. Coding techniques can be used to detect and
correct the errors.
6. Bandwidth required is low. Bandwidth required is higher.
7. FDM is used for multiplexing. TDM is used for multiplexing.
8. Not suitable for transmission of secret Due to coding techniques, it is suitable for
information in military applications. military applications.
9. Analog Modulation systems are AM, FM, DM systems are PCM, DM, ADM, DPCM,
PM, PAM, PWM, PPM, etc. etc.

5.4 LOW PASS SAMPLING:

 A message signal maybegin from a digital or analog source.
 If the message signal is analog in nature, then it has to be converted into digital form before it
can transmit by digital means.
 Sampling:The process by which the continuous-time signal is converted into a discrete–time
signal is called Sampling.
 Sampling operation is performed in accordance with the sampling theorem.

3. State and prove Nyquist sampling theorem. (Dec 2010) (or)

What is sampling? Explain and derive the expression of sampling. (Nov 2015) (or)
Describe the process of sampling and how the message signal is reconstructed from its
samples. Also illustrate the effect of aliasing with neat sketch. (Dec 2015) (or)
State the low pass sampling theorem and explain reconstruction of the signal from its
samples. [May 2016]

2
Sampling:The process by which the continuous-time signal is converted into a discrete–time signal is
called Sampling.

5.4.1 Sampling Theorem For Low-Pass Signals:-

Sampling theorem:
The bandpass signal g (t ) whose maximum bandwidth is 2W can be completely represented into and
recovered from its samples if it is sampled at the minimum rate of twice the bandwidth.
Sampling Theorem statements:
We may now state the sampling theorem* for band-limited signals of finite energy in two separate
parts
1. If a finite-energy signal g (t ) contains no frequencies higher than W hertz, it is completely
described by specifying its ordinates at a sequence of points spaced 1/2W seconds apart.
2. If a finite-energy signal g (t ) contains no frequencies higher than W hertz, it may be
completely recovered from its ordinates at a sequence of points spaced 1/2W seconds
apart.

Proof:-
 Consider an analog signal g (t ) that is Continuous in both time and Amplitude.
 Assume that g (t ) has infinite duration but finite energy.
 A segment of the signal g (t ) is depicted in Figure (1).
 Let the sample values of the signal g (t ) at times t  0,Ts ,2Ts ,...., be denoted by the series
g (nTs ), n  0,1,2,....
 We refer to Ts as the Sampling period and as the sampling rate.
 We define the discrete-time signal, g (t ) , that results from the sampling process as,

(1)
g (t )   g (nT ) (t nT )
n  
s s

where,  (t  nTs ) =Dirac delta function located at time t  n(Ts )

g (t ) = Sample values
g (t ) = Signal
 In equation (1) each delta function in the series is weighted by the corresponding sample value
of the input signal g (t ) .
 Figure. 1(b) illustrates the representation of g (t ) from sample values of g (t ) .

 From the definition of a delta function, we have

g (nTs ) (t  nTs )  g (t ) (t  nTs )

 Hence we may rewrite equation (1) in the equivalent form


g (t )  g (t )   (t nTs )
n  

 g (t )Ts (t ) (2)
where  T (t ) = Dirac comb (or) ideal sampling function
s

3
 Figure 1.Sampling process

 From equation (2), the discrete-time signal g (t ) is the output of an impulse modulator, which

operates on g (t ) as the modulating wave and  T (t ) as the carrier wave.

s

 This circuit-theoretic interpretation of g (t ) is depicted in Fig. (2)

Figure 2:Circuit-theoretic interpretation of the ideal sampling process as impulse modulation

 From the properties of the F.T., the multiplication of two time functions, as in equation (2), is
equivalent to the convolution of their respective Fourier transforms.

 Let G( f ) and G ( f ) denote the Fourier transforms of g (t ) and g (t ) , respectively.

 For the Fourier transform of  T (t ) , we have

s


(3)
F [Ts (t )]  f s   ( f  mf )
m  
s

where F [] signifies the Fourier transform operation, and f s is the sampling rate.

4
 Thus, transforming equation (2) into the frequency domain, we obtain

(4)
G ( f )  G( f )  [ f s   ( f  mf )]
m  
s

where  denotes convolution.

 Interchanging the orders of summation and convolution yields.

(5)
G ( f )  f s  G( f )   ( f
m  
 mf s )

 From the properties of a delta function, we find that convolution of G( f ) and  ( f  mfs )

equals G( f  mfs ) .

 Hence, we may simplify equation (5) as follows:


(6)
G ( f )  f s  G( f
m  
 mf s )

 From equation (6) G ( f ) represents a spectrum that is periodic in the frequency f with period

f s , but not necessarily continuous.

 In other words, the process of uniformly sampling a signal in the time domain resultsin a
periodic spectrum in the frequency domain with a period equal to the sampling rate.
 Thus, G ( f ) represents a periodic extension of the original spectrum G( f ) .

 Another useful expression for the Fourier Transform G ( f ) may be obtained by taking the
Fourier Transform of both sides of Eq. (1) and noting that the F.T. of the Delta function
 (t  nts ) is equal to exp(2nfTs ) .
 We may thus write

(7)
G ( f )   g (nT )
m  
s exp(  j 2nfTs )

 This relation may be viewed as a complex F.S. representation of the periodic frequency
function G ( f ) , with the sequence of samples g (nTs ) defining the coefficients of the
expansion.

 Note that in the F.S. defined by Eq. (7) the usual roles of time and frequency have been
interchanged.

 These relations are applied to any continuous-time signal g (t ) of finite energy and infinite
duration.
 Suppose, however that the signal is strictly band limited, with no frequency components
higher than W hertz.

5
 That is the F.T. G( f ) of the signal g (t ) has the property that G( f ) is zero for f  W , as

illustrated in Fig. 3(a)002E

 The shape of the spectrum shown in this figure is intended for the purpose of illustration only.
 The fact that the signal g (t ) has finite energy means that the area under the curve of the energy
2
spectral density G( f ) is likewise finite.

 Suppose also that we choose the samplingperiod Ts  1/ 2W .

 Then the corresponding spectrum G ( f ) of the sampled signal g  (t ) is as shown in Fig. 3 (b).

Figure 3: (a) Spectrum of signal g (t ) . (b) Spectrum of sampled signal g (t ) for a sampling

rate f s  2W . (c) Ideal amplitude response of reconstruction filter.

 substituting Ts  1/ 2W in Eq. (7) yields


 n   jnf 
G ( f )   g  2W  exp  
m W 
 (08)

6
 Putting fs=2W in Eq. (6), we have
1
G( f )  G ( f ) , -W< f < W (09)
2W

 It follows from Eq. (8) that we may also write

1 
 n   jnf 
G( f ) 
2W
 g  2W  exp  
m W 
 , -W< f < W (10)

 Therefore, if the sample values g (n / 2W ) of the signal g (t ) are specified for all time, then the
F.T. G( f ) of the signal is uniquely determined by using the F.S. of Eq. (10)
 Because g (t ) is related to G( f ) by the inverse F.T., it follows that the signal g (t ) is itself
uniquely determined by the sample values g (n / 2W ) for    n   .
 In other words, the sequence g (n / 2W ) contains all the information of g (t ) .

 Consider next the problem in reconstructing the signal g (t ) from the sequence of sample
values g (n / 2W ) .
 Substituting Eq. (10) in the formula for the inverse F.T. defining g (t ) in terms of G( f ) , we
get

g (t )   G( f ) exp( j 2ft )df


1   n   jnf 
W
 W 2W n g  exp  
   2W 
 exp( j 2ft )df
 W 
 Interchanging the order of summation and integration:

 n  1   n  (11)
W
g (t )   g     2W df
n  2W  2W W
exp  j 2f  t 

 The integral term in Eq. (11) may be readily evaluated, yielding*


 n  sin(2Wt  n ) (12)
g (t )   g  2W 
n   (2Wt  n )

 We may simplify the notation in Eq. (12) by using the sinc function, defined as
sin(x) (13)
sincx =
x
wherexis an independent variable.

7
  The sinc function exhibits an important property known as the interpolatory property, which is
describes as follows:
1, x  0 (14)
sincx = 
0, x  1,2,....
 Using the definition of the sinc function, we may rewrite Eq. (12) as follows:

 n  (15)
g (t )   g  2W  sinc (2Wt  n)
n

 Eq. (15) provides an interpolation formula for reconstructing the original signal g (t ) from the
sequence of sample values g (n / 2W ) ,
 The sinc function sinc(2Wt)playing the role of an interpolation function.

 Each sample is multiplied by a delayed version of the interpolation function, and all the
resulting waveforms are added to obtain.
 It is important that Eq. (15) represents the response of an ideal low-pass filter of
bandwidthWand zero transmission delay, which is produced by an input signal consisting of
the sequence of samples g (n / 2W ) for    n   .

 From the spectrum in Fig. 3 (b), the original signal g (t ) may be recovered exactly from the
sequence of samples g (n / 2W ) by passing it through an ideal low-pass filter of bandwidth W.
 This is illustrated in block diagrammatic form in Fig. (4).

Figure 4: Reconstruction filter

 The ideal amplitude response of the reconstruction filter is shown in Fig. 3(c).

(1) Signal Space Interpolation

4. Explain in detail about signal space interpolation.

 We may develop another interpretation of Eq. (15) b using the property of the function
sinc(2Wt– n), where n is an integer, is one of a family of shifted sinc functions that are
mutually orthogonal.
 To prove this property, we use the formula

8
   (16)
 g1 (t ) g 2 * (t )dt   G1 (t )G2 * (t )df
 

Where g1 (t ) and G1 ( f ) form a F.T. pair; likewise for g 2 (t ) and G2 ( f ) .

 This relation may be viewed as a generalization of Rayleigh’s energy theorem.

 Put
  n 
g1 (t )  sinc (2Wt  n) =sinc 2W  t  
  2W 
and
  m 
g 2 (t )  sinc (2Wt  m) =sinc 2W  t  
  2W 
 For a sinc pulse, sinc(2Wt), we have the F.T pair:

sinc (2Wt )  1 rect  f 

(17)
2W  2W 

where, on the right side, the definition of a rectangular function is used, namely
 1 1 (18)
1,   x 
 2 2
rect(x) = 
0, x  1
 2

 The functions of g1 (t ) and g 2 (t ) are time-shifted versions of the sinc pulse sinc(2Wt).
 Using the time shifting property of the F.T., we may express the F.T.s‟ of g1 (t ) and g 2 (t ) , as
follows, respectively.
jnf
rect  f  exp 
1
G1 ( f ) 
2W  2W  W
and
jmf
rect  f  exp 
1
G2 ( f ) 
2W  2W  W
 Hence, the use of these two F.T.s‟ inEq. (16) yields

 jf 
2 W
 1 


sinc (2Wt  n) sinc (2Wt  m) dt=  
 2W 
W  W (n  m)df
exp

sin (n  m)
=
2W (n  m)

1
= sinc (n  m)
2W
 This result equals 1/2Wwhen n  m , and zero when n  m (see Eq. (14)).
 We therefore have

9
   1 (19)
 ,n  m
 sinc (2Wt  n) sinc (2Wt  m) dt=  2W
 0, n  m

 Eq.(15) represents the expansion of the signal g (t ) as an infinite sum of orthogonal functions
with the coefficients of the expansion, defined by
 (20)
 n 
g   2w  g (t ) sinc (2Wt  n) dt
 2W  

 The coefficients g (n / 2W ) of this expansion are coordinates in an infinite-dimensional signal

space.
 In this space each signal corresponds to precisely one point and each point to one signal.

(2) Statement of the Sampling Theorem:

5. Give the statement of sampling theorem. (Nov 2013, Dec 2010, May 2012)

 The sampling theorem* for band-limited signals of finite energy in two separate parts
1. If a finite-energy signal g (t ) contains no frequencies higher than W hertz, it is completely
determined by specifying its ordinates at a sequence of points spaced 1/2W seconds apart.
2. If a finite-energy signal g (t ) contains no frequencies higher than W hertz, it may be
completely recovered from its ordinates at a sequence of points spaced 1/2W seconds
apart.
 Part 1 is a restatement of Eq. (10), and part 2 is restatement of Eq. (15).
 Nyquist rate:The minimum sampling rate of 2W samples per second, for a signal bandwidth of
W hertz, iscalled the Nyquist rate.
 Nyquist interval :The reciprocal, 1/2W, is called the Nyquist interval.
 The sampling theorem is the beginning for the interchangeability of analog signals and
digital sequences, which is so valuable in digital communication systems.
5.4.2 Types of sampling (Practical Sampling):
6. What are the types of sampling? (or)What is natural sampling and flat top sampling? (May 2010)
1. Ideal Sampling (or)Instantaneous sampling (or) Impulse sampling:

Fig 5(a) Functional diagram of a Fig 5(b) Message x(t ) and sampled x (t )

10
 switching sampler signals

 Ideal sampling is same as instantaneous sampling.

 Fig.5(a)shows the switching sampler.
 If closing time't' of the switch approaches zero the output x (t ) gives only

instantaneous value. The waveforms are shown in Fig. 5(b).

 Since the width of the pulse approaches zero, the instantaneous sampling gives train of
impulsesin x (t ) . The area of each impulse in the sampled version is equal to

instantaneous value of input signal x(t ) .

2. Natural Sampling (or) Chopper Sampling:

Figure6. Natural sampling

 Although instantaneous sampling is a convenient model, a more practical way of
sampling a band-limited analog signal m(t) is performed by high-speed switching
circuits.
 An equivalent circuit employing a mechanical switch and the resulting sampled signal
are shown in Fig. 6(a) and (b),respectively.
 The sampled signal xns (t ) can be written as

xns (t )  m(t ) x p (t )  (1)

Where x p (t ) is the periodic train of rectangular pulses with period Ts, and each

rectangular pulse in x p (t ) has width d and unit amplitude.

 The sampling here is termed natural sampling, since the top of each pulse in xns (t )
retains the shape of its corresponding analog segment during the pulse interval.
3. Flat-Top Sampling (or) Rectangular Pulse Shaping:

11
 Figure 7.Flat-top Sampling
 The simplest and thus most popular practical sampling method is actually performed
by a functional block termed the sample-and-hold (S/H) circuit [Fig. 7(a)].
 This circuit produces a flat-top sampled signal xs (t ) [Fig. 7(b)].

5.4.3 Comparison of Various Sampling Techniques

7. Compare the types of sampling.

12
5.5 Aliasing:

8. What is meant by aliasing effect, how it could be rectified?(Dec 2015)(or)

Write a detailed note on Aliasing and Signal Restoration. (06m) (April/May 2018) (or)
Explain in detail about aliasing.(or)
Explain in detail about the effects of undersampling.(Nov 2015)(or)
Write short notes on signal reconstruction (3m) (Nov 2017)
[Apr - 2019]
Aliasing Phenomenon

 Derivation of the sampling theorem, is based on the assumption that the signal g (t ) is strictly
band-limited.
 In practice, the information-bearing signalfrom the source is not a strictly band-limited signal.
 So, it resultsin some degree of undersampling.
 As a result, aliasing is produced by the sampling process.

Figure 8. (a) Spectrum of a signal. (b) Spectrum of an undersampled version of the signal,
exhibiting the aliasing phenomenon.

 Aliasing effect:
 Aliasing refers to the phenomenon of a high-frequency component in the spectrum of
the signal interferes and appears as lower frequency in the spectrum of its sampled
version, (as illustrated in Fig.)
 The aliased spectrum shown by the solid curve in Fig. 8(b) is related to an “undersampled”
version of the message signal represented by the spectrumof Fig. (a).
 To reduce the effects of aliasing in practice, thereare two corrective measures:
1. Before sampling, a low-pass anti-alias filter is used to attenuate those high-
frequencycomponents of the message signal that are not essential to the information
being conveyedby the signal.
2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

13
  The use of a sampling rate higher than the Nyquist rate eases the design of the synthesis filter
which is used to recover the original signal from its sampledversion.
 Consider the example of a message signal that has been anti-alias (low-pass) filtered,
resulting in the spectrum shown in Fig. 9(a).
 The spectrum of theinstantaneously sampled version of the signal is shown in Fig. 9(b),
assuming a samplingrate higher than the Nyquist rate.
 Fromfig. 9(b), the design of a physically realizable reconstruction filter to recoverthe original
signal from its uniformly sampled version may be achieved as follows (seeFig. 9(c)):
 The reconstruction filter is of a low-pass kind with a passband extending from  W to
W , which is itself determined by the anti-alias filter.
 The filter has a non-zero transition band extending (for positive frequencies) from W
to f s  W , where f s is the sampling rate.

Fig 9 (a) Anti-alias filtered spectrum of an information-bearing signal. (b) Spectrumof

instantaneously sampled version of the signal, assuming the use of a sampling rate greaterthan the
Nyquist rate. (c)Idealized amplitude response of the reconstruction filter.

 The non-zero transition band of the filter assures physical realizability, it is shown as dashed
linesto emphasize the arbitrary way of actually realizing it.
****

14
5.6 Signal Reconstruction

Reconstruction of a message process from its samples

9. Explain in detail about the reconstruction message process from its samples. (or)
Derive the mean square value of error in reconstruction process. (Dec 2015)
 This process completes the sampling process.
 Consider a wide-sense stationary message process X (t ) with autocorrelation function RX ( )
and power spectral density S X ( f ) .
 We assume that

S x ( f )  0 for f  W (01)

 Consider an infinite sequence of samples taken at a uniform rate equal to 2W , that is, twice
the highest frequency component of the process.
 Using X ' (t ) to denote the reconstructed process, based on this infinite sequence of samples,
we may write

 n  (02)
X ' (t )   X  2W  sinc (2Wt  n)
n  

where X (n / 2W ) is the random variable obtained by sampling or observing the message process

X (t ) at time t  n / 2W .
 The mean-square value of the error between the original message process X (t ) and the
reconstructed message process X ' (t ) equals

ξ= E[( X (t )  X ' (t )) ] 2

= E[( X (t )]  2E[ X (t ) X ' (t )]  E[( X ' (t )) ] (03)

2 2

 The first expectation term on the right side of Eq. (03) as the mean-square value of X (t ) ,
which equals RX (0) ; thus

E[( X 2 (t )] = RX (0) (04)

 For second expectation term, use Eq. (02) and so write

 
 n  
E[ X (t ) X ' (t )] = E  X (t )  X  sinc (2Wt  n)
 n  2W  

 Interchanging the order of summation and expectation:

15
 
  n 
E[ X (t ) X ' (t )] =  E  X (t ) X  2W sinc (2Wt  n)
n


 n 
= R
n
X t 
 2W 
 sinc (2Wt  n) (05)

 For a stationary process, the expectation E[ X (t ) X ' (t )] is independent of time t.

 Hence, putting t=0in the right side of Eq. (05) and recognizing that
 n   n 
RX    = RX  
 2w   2W 
 It may be written as

 n 
E[ X (t ) X ' (t )] = R
n  
X   sinc (n)
 2W 
(06)

 n 
 The term RX   represents sample of the autocorrelation function RX ( ) taken at   n / 2W .
 2W 
 Now, since the power spectral density S X ( f ) or equivalently the F.T. of RX ( ) is zero for

f  W , we may represent RX ( ) in terms of its samples taken at   n / 2W as follows


 n 
RX ( ) = R
n
X  sinc (2W  n)
 2W 
(07)


 n 
If   0  RX (0) = R
n
X  sinc (n)
 2W 

Accordingly, we deduce from Eqs. (06) and (07) that

(06)  E[ X (t ) X ' (t )] = RX (0) (08)

 For third and final expectation term on the right side of Eq. (03), we again use Eq. (02) and so
write
   n  
 k  
2
E[( X ' (t )) ] = E  X   sinc ( 2Wt  n )  X  sinc (2Wt  k )
n  2W  k   2W  

  
 n   k  
= E   sinc (2Wt  n)  X  X   sinc (2Wt  k )
n k   2W   2W  
 Interchanging the order of expectation and inner summation:

 
  n   k 
E[( X ' (t ))2 ] = 
n
sinc ( 2Wt  n )  EX  X   sinc (2Wt  k )
k    2W   2W 

16
  
n  k 
=  sinc (2Wt  n)  R
n k 
X  2W  sinc (2Wt  k ) (09)

 However, in view of Eq. (07), the inner summation on the right side ofEq. (09)equals
 n 
RX  t  .
 2W 
 Hence, we may simplify Eq. (09) as follows

E[( X ' (t ))2 ]  n  (10)
= R
n
X  2W  sinc (2Wt  n)
t 

= RX (0)
 Finally, substituting Eqs. (04) , (08), into (10), we get the result
ξ =0
as should be expected.

 We may therefore state the sampling theorem for message processes as follows.
 If a stationary message process contains no frequencies higher than W hertz, it may be
reconstructed from its samples at a sequence of points spaced 1/2Wseconds apart with
zero mean squared error (i.e., Zero error power).

5.7 Quantization
10. Explain in detail about the quantization process. [Apr 2010, Apr 2011]
(or)
Illustrate and describe the types of quantizer? Describe the midtread and midrise type
characteristics of uniform quantizer with a suitable diagram. [Dec 2016]

 A continuous signal (i.e., voice) has a continuous range of amplitudes and therefore its
samples also have a continuous amplitude range.
 In other words, within the finite amplituderange of the signal, there are infinite number of
amplitude levels.
 It is not necessary in fact to transmit the exact amplitudes of the samples.

 Any human sense (the ear or the eye), can detect only finite intensity differences.
 So, the original continuous signal will be approximated by a signal constructed of discrete
amplitudes.
 The existence of a finite number of discrete amplitude levels is a basic condition of pulse-code
modulation.

 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 a discrete amplitude v(nTs) taken from a
finite set of possible amplitudes.

17
 Fig: 10. Description of a memoryless quantizer

 Assume that the quantization process ismemoryless and instantaneous.

 It means, the transformation at time t = nTs is not affected by earlier or later samples of the
message signal.
 This simple form of scalar quantization is commonly used in practice.

 When dealing with a memoryless quantizer, we may simplify the notation by dropping the
time index.
 The symbolm in place of m(nTs)as indicated in the block diagram of a quantizer shown in
Figure 10a.
 Then, as shown in Figure. 10b, the signal amplitude m is specified by the index k if it lies
inside the partition cell

where L is the total number of amplitude levels used in the quantizer.

 The discrete amplitudesmk,k = 1, 2, ... , L, at the quantizer input 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 cell .
 The discrete amplitudes vk ,k = 1, 2, ... , L, are called representation levels or reconstruction
levels,
 The spacing between two adjacent representation levels is called a quantum size or step-size.
 Thus, the quantizer output v equals vk if the input signal sample m belongs to the interval .
 The mapping,

is the quantizer characteristic, which is a staircase function by definition.

 Types of quantizers:
 Quantizers can be of a uniform or nonuniform type.
 In a uniform quantizer, therepresentation levels are uniformly spaced; otherwise, the
quantizer is nonuniform.

5.7.1 Uniform & non-uniform quantization:

11. Illustrate and describe the types of quantizer? Describe the midtread and midrise type
characteristics of uniform quantizer with a suitable diagram. [Dec 2016]
 In a uniform quantizer, the representation levels are uniformly spaced; otherwise, the
quantizer is nonuniform.

18
 5.7.1.1 Uniform Quantization
 The quantizer characteristic can also be of a midtread or midrise type.
 Midtread:
 Figure 11(a) shows the input–output characteristic of a uniform quantizer of the
midtread type
 It is so called because the origin lies in the middle of a tread of the staircaselike
graph.

Figure 11. Two types of quantization: (a) midtread and (b) midrise.
 Midrise:
 Figure 11(b) shows the corresponding input–output characteristic of a uniform
quantizer of themidrise type.
 It is so called becausethe origin lies in the middle of a rising part of the staircaselike
graph.
 Note that both the midtread and midrise types of uniformquantizers aresymmetric about the
origin.

5.7.1.2 Nonuniform Quantization

12. Explain non-uniform quantization. (Apr 2010, Apr 2011, May 2014)

 The sampled version of the message signal will be quantized.

 Quantization provides a newrepresentation of the signal that is discrete in both time and
amplitude.
 In some applications,it is preferred to use a variable separation between therepresentation
levels.
 For example, 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 of1000 to 1.

19
  By using a nonuniformquantizer with the feature that the step size increasesas the separation
from the origin of the input–output amplitude characteristic isincreased
 The large end-step of the quantizer can take care of possible excursions ofthe voice signal
into the large amplitude ranges that occur in rare.

 The use of a nonuniformquantizer is equivalent to passing the message signalthrough a

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

  law

 A particular form of compression law that is used in practice is the so called   law defined
by

(01)
where the logarithm is the natural logarithm; m andv are respectively the normalized input and
output voltages, and  is a positive constant.

Figure 12. Compression laws. (a) m-law. (b) A-law.

 For convenience of presentation, the input to the quantizer and its output are both normalized
to occupy a dimensionless range of values from zero to one, as shown in Figure12(a); here
  law is plotted for varying  .
 Practical values of  tend to be approximately 255. The case of uniform quantization
corresponds to   0 .

20
 For a given value of  ,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,

(02)

 The   law is neither strictly linear nor strictly logarithmic

 But it is approximately linear at low input levels corresponding to  m  1 and approximately

logarithmic at high input levels corresponding to  m  1.

A-Law:

 Another compression law that is used in practice is the so-called A-law, defined by

(03)
which is shown plotted in Figure 12(b). Typical values of A used in practice tend to be in the
vicinity of 100. The case of uniform quantization corresponds to A  1 .

 The reciprocal slope of this second compression curve is given by the derivative of m with

respect to v as shown by

(04)
 From the first line of Eq. (04), 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 typically about 25 dB in practice, as comparedwith uniform quantization.

***

21
5.8 Quantization noise:

13. With proper diagram explain the noise due to quantization in digitalization process.
[May 2006, 2013], [Dec 2005, 2008, 2012, 2013, 2014]
Derive the expression for signal to noise ratio of uniform quantizer.
[April 2018, Nov 2017]

5.8.1 Illustration of Quantization noise:

Quantization introduces an error, defined as, the difference between the input signal m and the output
signal v. The error is called quantization noise.

 Figure 13 shows 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 variableM.(If the input
has a nonzero mean, it can be always removed by subtracting the mean from the input and
then adding it back after quantization.)

Figure13. Illustration of Quantization process

 A quantizer g () maps the inputrandom variableMof continuous amplitude into a discrete
random variable V; their respective sample values m and v are related by Equation
v=g(m) (01)
 Let the quantization errorbe denoted by the random variable Q of sample value q.

22
 We may thus write
q=m-v (02)
or, correspondingly,
Q=M-V (03)
 With the input M having zero mean, and the quantizer assumed to be symmetric as in Figure
5.10, it follows that the quantizer output Vand therefore the quantization error Q, will also
have zero mean.
 So, for the characterization of the quantizer in terms of output signal-to- quantization noise
ratio, find the mean-square value of the quantization error Q.
 Consider then an input m of continuous amplitude in the range (-mmax, mmax).

5.8.2 Signal to noise ratio of uniform quantizer:

 Assuming a uniform quantizer of the midrise type illustrated in Figure 3.10b, we find that the
step-size of the quantizer is given by

2mmax
 (04)
L

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 thestep-size  is sufficiently small (i.e., the number of representation levels L is sufficiently

large)
 It isreasonable to assume that the quantization error Q is a uniformly distributed random
variable, and the effect of the quantization noise on the quantizer input is similar to that of
thermal noise.
 Thus the probability density function of the quantization error Q is expressed as follows:

(05)
 For this to be true, the incoming signal does not overload the quantizer.
 Then, with the mean of the quantization error being zero, its variance  Q2 is the same as the

mean-square value:

(06)

23
 Substituting Equation (5) into (6), we get

(07)
 Typically, the L-arynumber k, denoting the Kth representation level of the quantizer,
istransmitted to the receiver in binary form.
 Let R denote the number of bits per sampleused in the construction of the binary code.
 We may then write
L  2R (08)
or, equivalently,
R  log 2L (09)
 Hence, substituting Equation (8) into (4), we get the step size
2mmax
 (10)
2R
 Thus the use of Equation (10) in (7) yields

1
 Q2  mmax
2
2 2 R (11)
3
 Let P denote the average power of the message signal m(t). We may then express the output
signal-to-noise ratio of a uniform quantizer as
P
( SNR) O =
 Q2
3P 2 R
= 2
2 (12)
mmax
 Equation (12) 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 (transmission) bandwidth BT.

 The use of a binary code for the representation of a message signal (as in pulse- code
modulation) provides a more efficient method than either frequency modulation (FM) or
pulse-position modulation (PPM) for the trade-off of increased channel bandwidth for
improved noise performance.

 In making this statement, we assume that the FM and PPM systems are limited by receiver
noise, whereas the binary-coded modulation system is limited by quantization noise.

24
5.9 Logarithmic Companding of speech signal

14. Explain in detail about Logarithmic Companding of speech signal. (or)

In detail explain lograthmic companding of speech signals (4m) [Nov 2017]
 Introduction Pulse code modulation (PCM) is a common method of digitizing or quantizing an
analog waveform.
 For any analog-to-digital conversion process, the quantization step produces an estimate of the
waveform sample using a digital codeword.
 This digital estimate inherently contains some level of error due to the finite number of bits
available.
 In practical terms, there is always tradeoff between the amount of error and the size of the
digital data samples.
 The goal in any system design is quantizing the data in smallest number of bits that results in a
tolerable level of error.
 In the case of speech coding, linear quantization with 13 bits sampled at 8 KHz is the
minimum required to accurately produce a digital representation of the full range of speech
signals.
 For many transmission systems, wired or wireless, 13 bits sampled at 8 KHz is an expensive
proposition as far as bandwidth is concerned. To address this constraint, a compandingsystem
is often employed.

 Companding is simply a system in which information is first compressed, transmitted through

a bandwidthlimited channel, and expanded at the receiving end.
 It is frequently used to reduce the bandwidth requirements for transmitting telephone quality
speech, by reducing the 13-bit codewords to 8-bit codewords.
 Two international standards for encoding signal data to 8-bit codes are A-law and  -law. A-
lawis the accepted European standard, while  -law is the accepted standard in the United
States and Japan.

5.9.1 Speech Companding

 The human auditory system is believed to be a logarithmic process in which high amplitude
sounds do not require the same resolution as low amplitude sounds.
 The human ear is more sensitive to quantization noise in small signals than large signals.
 A-law and  -law coding apply a logarithmic quantization function to adjust the data
resolution in proportion to the level of the input signal. Smaller signals are represented with
greater precision – more data bits – than larger signals.

25
 The result is fewer bits per sample to maintainan audible signal-to-noise ratio (SNR).

 Rather than taking the logarithm of the linear input data directly, which can be
computationally difficult, A-law/  -law PCM matches the logarithmic curve with a piece-
wise linear approximation.

 Eight straight-line segments along the curve produce a close approximation to the logarithm
function. Each segment is known as a chord.
 Within each chord, the piece-wise linear approximation is divided into equally size
quantization intervals called steps.
 The step size between adjacent codewords is doubled in each succeeding chord.
 Also encoded is the sign bit for the original integer.
 The result is an 8-bit logarithmic code composed of a 1-bit sign bit, a 3-bit chord, and a 4-bit
step.

5.9.2 A-Law Compander

 A-law is the CCITT recommended companding standard used across Europe.
 Limiting the linearsample values to 12 magnitude bits, the A-law compression is defined by
Equation 1, where A is the compression parameter (A=87.7 in Europe), and x is the
normalized integer to be compressed.

(Eq. 1), A-law definiton

 Table 1 illustrates an A-law encoding table. The sign bit of the linear input data is omitted
fromthe table.
 The sign bit (S) for the 8-bit code is set to 1 if the input sample is negative, and is set to0 if the
input sample is positive.

26
  After the input data is encoded through the logic defined in the table, an inversion pattern
isapplied to the 8-bit code to increase the density of transitions on the transmission line, a
benefit tohardware performance.
 The inversion pattern is applied by XOR‟ing the 8-bit code with 0x55.
 Decoding the A-law encoded data is essentially a matter of reversing the steps in the encoding.
 Table 2 illustrates the A-law decoding table, applied after reversing the inversion pattern.
 Theleast significant bits discarded in the encoding process are approximated by the median
value ofthe interval. This is shown in the output section by the trailing 1..0 pattern after the D
bit.

5.9.3  -Law Compander

 The United States and Japan use  -law companding. Limiting the linear sample values to
13magnitude bits, the  -law compression is defined by Equation 2, where m is the
compressionparameter (m =255 in the U.S. and Japan) and x is the normalized integer to be
compressed.

 The encoding and decoding process for  -law is similar to that of A-law. There are, however,
afew notable differences:
1)  -law encoders typically operate on linear 13-bit magnitude data, asopposed to 12-bit
magnitude data with A-law,
2) before chord determination a bias value of 33 isadded to the absolute value of the linear
input data to simplify the chord and step calculations,
3)the definition of the sign bit is reversed, and 4) the inversion pattern is applied to all bits in
the 8 bit code.

27
 Table 3 illustrates a  -law encoding table. The sign bit of the linear input data is omitted
from thetable.
 The sign bit (S) for the 8-bit code is set to 1 if the input sample is positive, and is set to 0 ifthe
input sample is negative.

 After the input data is encoded through the logic defined in the table, an inversion pattern
isapplied to the 8-bit code to increase the density of transitions on the transmission line, a
benefit tohardware performance. The inversion pattern is applied by XOR‟ing the 8-bit code
with 0xFF.
 Decoding the  -law encoded data is essentially a matter of reversing the steps in the
encoding.Table 4 illustrates the  -law decoding table, applied after reversing the inversion
pattern.
 Theleast significant bits discarded in the encoding process are approximated by the median
value ofthe interval. This is shown in the output section by the trailing 1..0 pattern after the D
bit.

Summary
 There is a wide array of audio transmission systems that employ A-law and/or  -law
companding for data rate reduction with good audio quality.
 The compression achieved by both A-law and  -law coding is the result of utilizing the
logarithmic characteristics of the human auditory system, where fewer bits of precision are
required for larger signals than smaller ones.
 The logarithmic transfer function is implemented with a piece-wise linear approximation
composed of a sign bit, a 3-bit chord, and a 4-bit segment.
 The encoding and decoding process is presented in table format, well suited for hardware
orsoftware implementation.

28
5.10 Pulse Amplitude Modulation (PAM)
Discuss about the generation of PAM and its demodulation. [Nov/Dec 2010]
Introduction
 The amplitude of the pulse carrier is changed in proportion with the instantaneous amplitude
of the modulating signal.
Types of PAM
Depending upon the shape of the PAM pulse, there are two types of PAM. They are:
(i) Natural PAM
(ii) Flat top PAM
The flat top pulses have constant amplitude within the pulse interval.
Why flat top PAM is widely used?
 During the transmission, the noise interferes with the flat top of the transmitted pulses and this
noise can be easily removed.
 In natural samples PAM, the pulse has varying top in accordance with the signal variation.
 When such type of pulse is received by the receiver, it always seems to be contaminated by
noise.
 Then it becomes quite difficult to determine the shape of the top of the pulse and therefore
amplitude detection of those pulses is not exact.
 As a result of this, errors are introduced in the received signal.
 The electronic circuitry needed to perform natural sampling is somewhat complicated because
the pulse top shape is to be maintained. These complications are reduced by flat-top PAM.
Natural PAM
Generation of natural PAM
 The modulating signal x (t) is passed through a low pass filter which will band limit this signal
to fm.
 That means all the frequency components higher than the frequency fm are removed.
 Band limiting is necessary to avoid the “aliasing” effect in the sampling process.
 The pulse train generator generates a pulse train of frequency fs, such that fs > 2 fm. Thus the
Nyquist criterion is satisfied. This is nothing but sampling signal.

29
 Fig : Generation of PAM

Fig : Waveforms of natural PAM generation

 The continuous time signal x (t) is applied at the input of a multiplier.
 The other input is a sampling signal s (t), which is a periodic train of pulses with unit
amplitude and a period of “T s” seconds.
 The uniform sampling takes place at the multiplier block to generate the PAM signal.
 The information in the modulating signal is contained in the “amplitude variations” of the
pulsed carrier.
Detection of Natural PAM
 The PAM signal can be detected by passing it through a low pass filter, which is tuned to fm.
 So all high frequency ripples is removed and original modulating signal is recovered back.

Fig : Detection of natural PAM

30
 Fig : Waveforms of natural PAM detection
Flat top PAM
Generation of flat top PAM
 A sample and hold circuit is used to produce flat top sampled PAM. This consists of the two
field effect transistors (FET) switches and a capacitor.
 Flat top PAM signals are generated by applying the input modulating signal x (t) to charging
(sampling) switch.
 At the sampling instant, sampling switch is closed for a short duration by a short pulse applied
to a gate G1 of the transistor.
 During this period, the capacitor “C” quickly charged up to a voltage equal to the
instantaneous sample value of the incoming signal x (t).
 Now, the sampling switch is opened and capacitor „C‟ holds the charge.
 The discharge switch is then closed by a pulse applied to gate G2.
 Due to this, the capacitor “C” is discharged to zero volts.
 The discharges switch is then opened and thus capacitor has no voltage.

Fig (a): Circuit to generate flat top PAM

31
 Fig (b): Flat top PAM signal
Fig : Generation of flat top PAM
Detection of flat top PAM

Fig : Detection of flat top PAM

 Detector contains a low-pass reconstruction filter with cut off frequency slightly higher than
the maximum frequency present in the message signal x (t).
 The equalizer compensates for the aperture effect. It also compensates for the attenuation by a
low pass filter.
Transmission Bandwidth of PAM Signal
 In a PAM signal, the pulse duration τ is considered to be very small in comparison to the time

period (sampling period) Ts between any two samples.

τ << Ts ……..(1)

From sampling theorem,

fs ≥ 2 fm

1
 2 fm
Ts
1
Ts 
2 fm
From (1),
1
  Ts 
2 fm
 If the ON and OFF time of PAM pulse is same, then maximum frequency of the PAM pulse
will be,
1 1
f max  
  2

32
 τ τ
Fig: ON and OFF pulses of PAM
 Therefore, the bandwidth required for the transmission of a PAM signal would be equal to the

maximum frequency fmax.

BW  f max

1

2
1
But,  
2 fm
1
BW   f m
2
BW  f m

Advantage: Simple generation and detection

Disadvantages:
 Effect of additive noise is high in PAM.
 Transmission bandwidth required is too large.
 The transmission power is not constant due to the changes in amplitudes of PAM pulses.
**********************************************************************
5.11 Pulse Time Modulation (PTM)
Explain the concept of pulse time modulation in detail.
Explain the concept and method of generating of PWM. What are the advantages and application
of PTM? (May – 2013) [Nov/Dec 2013]
 In pulse time modulation, amplitude of pulse is held constant, whereas position of pulse is
made proportional to the amplitude of signal at the sampling instant.
 There are two types of pulse time modulation. They are:
 Pulse width modulation
 Pulse position modulation [Apr - 2019]

Explain the generation and detection of PWM with neat diagram. (April / May – 2011)
With neat diagram, explain the generation and detection of PPM.
5.11.1 Pulse Width Modulation (PWM)
Introduction

33
  The width of the carrier pulses varies in proportion with the amplitude of modulating signal.
 The amplitude and frequency of the PWM wave remains constant.
 Only the width changes.
 The information is contained in the width variation.
 The additive noise, changes the amplitude of the PWM signal.
 Using the limiter circuit at the receiver, unwanted amplitude variations are easily removed.

Fig: PWM signal

 Amplitude variations due to noise do not affect the performance. Thus PWM is more immune
to noise than PAM.
PWM signal generation
 A saw tooth signal acts as a sampling signal which is applied to inverting terminal of a
comparator.
 The modulating signal x (t) is higher than that of the saw tooth signal. This gives to a PWM
signal.

Fig : Block diagram of PWM and PPM generation

34
 Fig : PWM and PPM waveforms
PWM signal detection
 The PWM signal received at the input of the detection circuit contains noise.
 It is applied to pulse generator which regenerates the PWM signal and remove noises.
 The regenerated pulses are applied to a reference pulse generator.
 The reference pulse generator produces reference pulses with constant amplitude and pulse
width.
 These pulses are delayed by specific amount of delay.

Fig: Block diagram of PWM detection circuit

 The regenerated PWM pulses are also applied to a ramp generator.
 The ramp generator produces ramps for the duration of pulses such that height of ramp is
proportional to the widths of PWM pulses.
 The maximum ramp voltage is retained till the next pulse.
 The delayed reference pulses and the output of ramp generator is added with the help of adder.

35
 Fig : Waveform for PWM detection circuit
 The output of the adder is then clipped off at a threshold level to generate PAM signals at the
output of the clipper.
 A low pass filter is used to recover the original modulating signal from PAM signal.
Advantages
 In PWM noise is less because here amplitude is constant.
 No synchronization required between transmitter and receiver.
 It is easy to separate the signal from noise.
Disadvantages
 Variable pulse width causes variable power contents. So, transmission must be powerful
enough to handle the maximum width.
 Bandwidth requirement is higher than PAM.

5.11.2 Pulse Position Modulation (PPM)

 The amplitude and width of the pulses are kept constant but the position of each pulse is
varied in accordance with the amplitude of the sampled values of the modulating signal.

36
PPM signal generation

Fig : Generation of PPM signal

 To generate pulse position modulation, the PWM pulses obtained at the output of the
comparator are used as the trigger input to a monostable multivibrator.
 The monostable is triggered on negative (falling) edge of PWM.
 The output of monostable goes high. This voltage remains high for the fixed period then goes
low.
 As a result of shifting the trailing edges of PWM signal in proportion with the modulating
signal x (t), the PPM pulses also results in keep shifting.
PPM signal demodulation
 The received PPM signal is noise corrupted.
 The pulse generator develops a pulsed waveform at its output of fixed duration and applies
these pulses to reset pin (R) of a SR flip flop.

Fig : PPM demodulator circuit

 A fixed period reference pulse is generated from the incoming PPM.
 The SR flip flop is set by the reference pulses.
 Due to the set and reset signals applied to the flip-flop, a PWM signal is obtained in the output
which can be demodulated with a PWM demodulator.
Advantages
 Due to constant amplitude of pulses, the transmitted power always remains constant.
 It is easy to reconstruct PPM signal from the noise contaminated PPM signal.
Disadvantages
 Synchronization required between the transmitter and receiver.
 Large bandwidth requirement.

37
Difference Between PAM, PWM, and PPM
Difference Between PAM, PWM, and PPM

The below table gives the detailed difference between PWM, PAM, and PPM.

Sr. No. Parameter PAM PWM PPM

1 Type of Carrier Train of Pulses Train of Pulses Train of Pulses

Variable Characteristic
2 of the Pulsed Carrier Amplitude Width Position

Bandwidth
3 Requirement Low High High

4 Noise Immunity Low High High

Information Contained Amplitude Position

5 in Variations Width Variations Variations

6 Power efficiency (SNR) Low Moderate High

Varies with Varies with Remains

7 Transmitted Power amplitude of pulses variation in width Constant

Need to transmit
8 synchronizing pulses Not needed Not needed Necessary

Bandwidth Bandwidth
Bandwidth depends depends on the depends on the
on the width of the rise time of the rise time of the
9 Bandwidth depends on pulse pulse pulse

Instantaneous
Instantaneous Instantaneous transmitter
transmitter power transmitter power power remains
varies with the varies with the constant with
amplitude of the amplitude and the width of the
10 Transmitter power pulses width of the pulses pulses

Complexity of
generation and
11 detection Complex Easy Complex

Similarity with other

12 Modulation Systems Similar to AM Similar to FM Similar to PM

********************************************************

38
 5.12 Pulse-Code Modulation

15. Explain the operation of PCM in detail with proper block diagrams.
(May 2013, Nov 2013)(or)
Describe PCM waveform coder and decoder with neat sketch and list the merits
compared with analog coders. [Dec 2015] (or)
Explain in detail about temporal waveform encoding scheme. (or)
Explain pulse code modulation system with neat block diagram. [May 2016] [Apr - 2019]

 PCM is the most basic form of digital pulse modulation.

 In pulse-code modulation (PCM), a message signal is represented by a sequence of coded
pulses, this is accomplished by representing the signal in discrete form in both time and
amplitude.
 The basic operations performed in the transmitter of a PCM system are
 Sampling
 quantization, and
 encoding, as shown in Fig.;

Operations in the transmitter

 The low-pass filter, prior to sampling, is included just to prevent aliasing of the message
signal.
 In practice, an anti-alias (low-pass) filter is used at the front end of the sampler to reject
frequencies greater than Wbefore sampling,Figure14(a).

 The quantizing and encoding operations are usually performed in the same circuit, which is
called an analog-to-digital converter.

(i) Sampling
 The incoming message (baseband) signal is sampled with a train of rectangular pulses, narrow
enough to closely approximate the instantaneous sampling process.
 For perfect reconstruction of the message signal at the receiver, the sampling rate must be
greater than twice the highest frequency component Wof the message signal (in accordance
with the sampling theorem).
 Function of sampling: Sampling permits the reduction of the continuously varying message
signal (of some finite duration) to a limitednumber of discrete values per second.

39
 Figure14. The basic elements of a PCM system
(a) Transmitter, (b) transmission path,connecting the transmitter to the receiver, and (c) receiver.

(ii) Nonuniform Quantization

 The sampled version of the message signal is then quantized.
 It provides a newrepresentation of the signal that is discrete in both time and amplitude.

(iii) Encoding
 **The use of an encoding process to convert the discrete set of sample values to a more
suitable form of signal.
 **Code: Plan for representing 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.
 **Code word: A particular arrangement of symbols 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, such as a negative pulse or
positive pulse.

The two symbols of the binary code are customarily denoted as 0 and 1. In practice, a binary
code is preferred over other codes (e.g., ternary code) for two reasons:
1. 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.
2. The binary code is easy to generate and regenerate.

40
Regeneration along the Transmission Path

 This capability is attained by reconstructing the PCM signal by means of a chain of

regenerative repeaters located at sufficiently close spacing along the transmission route.
 As illustrated in Figure15, three basic functions are performed by a regenerative repeater:
 Equalization, Timing and Decision making.
 **Equalizer: 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.
 **Timing circuitry:The timing circuitry provides a periodic pulse train, derived from the
received pulses; this is done for renewed sampling of the equalized pulses at the instants of
time where the signal-to-noise ratio is a maximum.
 **Decision-making device:The sample so extracted is compared to a predetermined threshold
in the decision-making device. In each bit interval, a decision is then made on 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, another clean new pulse representing symbol 0 is transmitted.

 In this way, the accumulation of distortion and noise in a repeater span is removed.
 In practice, however, 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 regeneratedsignal.
2. If the spacing between received pulses deviates from its assigned value, a jitter is
introduced into the regenerated pulse position, thereby causing distortion.

Figure15. Block diagram of Regenerative repeater

41
 Operations in the Receiver

(i) Decoding and Expanding

 The first operation in the receiver is to regenerate (i.e., reshape and clean up) the received
pulses.
 These clean pulses are then regrouped into code words and decoded (i.e., mapped back) into
a quantized PAM signal.

 Decoding: The decoding process involves generating a pulse whose amplitude is the linear
sum of all the pulses in the code word; each pulse is weighted by its place value
(20 ,21 ,2 2 ,.........,2 R1 ) in the code, where R is the number of bits per sample.

 The sequence of decoded samples represents an estimate of the sequence of compressed

samples produced by the quantizer in the transmitter.
 In order to restore the sequence of decoded samples to their correct relative level, a subsystem
is used in the receiver called an expander (complementary to the compressor, used in the
transmitter).
 The combination of a compressorand an expander is referred to as a compander.

(ii) Reconstruction
 The final operation in the receiver is to recover the message signal.
 This operation isachieved by passing the expander output through a low-pass
reconstruction filterwhose cutoff frequency is equal to the message bandwidth.
 Recovery of the messagesignal is intended to signify estimation rather than exact
reconstruction.

5.13 Time Division Multiplexing:

16. Explain in detail about the process of Time division multiplexing. [May 2010, Nov 2011] (or)
What is TDM? Explain the difference between analog TDM and digital TDM. [May 2016]

 Concept of TDM:The transmission of the message samples 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.

42
 ***Thetime-division multiplex (TDM) system, enables the joint utilization of a common
communication channel by a plurality of independent message sources without mutual
interferenceamong them.

FIGURE 16 Block diagram of TDM system.

 The concept of TDM is illustrated by the block diagram shown in Fig. 16.

Transmitting system:

Low-pass (anti-aliasing) filter:

 Each input message signal is first restricted in bandwidth by a low-pass anti-aliasing filter.
 It removes the frequencies that are non-essential to a satisfactory signal representation.

Commutator:

 The low-pass filter outputs are then applied to a commutator.

 It is usually implemented using electronic switching circuitry.
 The function of the commutator is twofold(dual):
(1) to take a narrow sample of each of the N input messages at a rate that is slightly higher
than 2W, where W is the cutoff frequency of the anti-aliasing filter, and
(2) to sequentially interleave these N samples inside the sampling interval Indeed, this latter
function is the essence of thetime-division multiplexing operation.

Pulse modulator:
 Next to the commutation process, the multiplexed signal is applied to a pulse modulator.
 Pulse modulator transforms the multiplexed signal into a form suitable for transmission over
the common channel.
 The use of time-division multiplexing introduces a bandwidth expansion factor N, because the
scheme must squeeze N samples derived from N independent message sources into a time slot
equal to one sampling interval.

43
Receiving System
Pulse Demodulator:
 At the receiving end of the system, the received signal is applied to a pulse demodulator,
which performs the reverse operation of the pulse modulator.

Decommutator:
 The narrow samples produced at the pulse demodulator output are distributed to the
appropriate low-pass reconstruction filters through a decommutator.
 Decommutator operates in synchronism with the commutator in the transmitter.
 This synchronization is essential for a satisfactory operation of the system.
 Synchronization depends on the method of pulse modulation used to transmit the multiplexed
sequence of samples.

Equalization:
 The TDM system is highly sensitive to dispersion in the common channel.
 A non-constant magnitude response of the channel and a nonlinear phase response, both being
measured with respect to frequency.
 Accordingly, equalization of both magnitude and phase responses of the channel is necessary
to ensure a satisfactory operation of the system; in effect, equalization compensates for
dispersion in the channel.
 However, unlike frequency-division multiplexing (FDM), to a first-order
approximation TDM is immune to nonlinearities in the channel as a source of cross-
talk.
 The reason for this behavior is that different message signals are not simultaneously
applied to the channel.

Synchronization
 For a PCM system with time-division multiplexing to operate satisfactorily, it is necessary that
the timing operations at the receiver, except for the time lost in transmission and regenerative
repeating, follow closely the corresponding operations at the transmitter.
 In a general way, this amounts to requiring a local clock at the receiver to keep the same time
as a distant standard clock at the transmitter, except that the local clock is delayed by an
amount equal to the time required to transport the message signals from the transmitter to the
receiver.

44
5.14 Frequency-Division Multiplexing (FDM)
[Apr - 2019]
Explain in detail about Frequency-Division Multiplexing (FDM) .

 Frequency-Division Multiplexing (FDM) is a scheme in which numerous signals are

combined for transmission on a single communications line or channel.
 It is analog multiplexing technique. Each signal is assigned a different frequency (sub
channel) within the main channel. its requires channel synchronization.
 FDM multiplexing technique is based on orthogonality of sinusoids.
 FDM requires that the bandwidth of a link should be greater than the combined bandwidths of
the various signals to be transmitted.
 Thus each signal having different frequency forms a particular logical channel on the link and
follows this channel only.
 These channels are then separated by the strips of unused bandwidth called guard bands.

These guard bands prevent the signals from overlapping as shown in Fig.

In FDM, signals to be transmitted must be analog signals. Thus digital signals need to be converted to
analog form, if they are to use FDM.

A typical analog Internet connection via a twisted pair telephone line requires approximately three
kilohertz (3 kHz) of bandwidth for accurate and reliable data transfer.

Twisted-pair lines are common in households and small businesses. But major telephone cables,
operating between large businesses, government agencies, and municipalities, are capable of much
larger bandwidths.

Advantages of FDM:
1. A large number of signals (channels) can be transmitted simultaneously.
2. FDM does not need synchronization between its transmitter and receiver for proper operation.
3. Demodulation of FDM is easy.
4. Due to slow narrow band fading only a single channel gets affected.

Disadvantages of FDM:
1. The communication channel must have a very large bandwidth.
2. Intermodulation distortion takes place.
3. Large number of modulators and filters are required.

45
4. FDM suffers from the problem of crosstalk.
5. All the FDM channels get affected due to wideband fading.

Applications of FDM
1. FDM is used for FM & AM radio broadcasting. Each AM and FM radio station uses a different
carrier frequency. In AM broadcasting, these frequencies use a special band from 530 to 1700 KHz.
All these signals/frequencies are multiplexed and are transmitted in air. A receiver receives all these
signals but tunes only one which is required. Similarly FM broadcasting uses a bandwidth of 88 to
108 MHz

2. FDM is used in television broadcasting.

3. First generation cellular telephone also uses FDM.

PROBLEMS

17. A PCM sinusoidal has a uniform quantizer followed by a ‘v’ bit encoder. Show that the rms
signal to noise ratio is approximately given by 1.8 + 6 v dB, assuming a sinusoidal input.
[April/May 2018]
Solution:
Assume that the modulating signal be a sinusoidal voltage, having peak amplitude Am. Let the
signal cover the complete excursion os representation levels.

The power of the signal will be,

v2
P
r
2
A 
 m 
 2
When R=1, the power P is normalized, i.e.,
Normalized power:
Am2
P
2 , with R=1 in the above equation
Therefore, The signal to quantization noise ratio is given by
P
( SNR) O 
 Q2
3P 2 R
 2
2
mmax

Substitute:
Am2
P mmax  Am
2 ,

46
 Am2
3
3P 2 R 3
( SNR) O  2
2  22 2 2 R  2 2 R  1.5  2 2 R
mmax Am 2

Expressing signal to noise ratio in dB,

(SNR)dB  10 log110.5  10 log10  1.76  (2v  10  0.3)
2R
2

( SNR)dB in PCM  1.8  6v; for sinusoidal signal

18. Show that the signal to noise power ratio of a uniform quantizer is PCM system increases
significantly with increase in number of bits per sample. Also determine the signal to
quantization noise ratio of an audio signal S t   4 sin(2 500t ) , which is quantized using a
10 bit PCM. [April/May 2018, Nov 2017]
Given:
S t   4 sin(2 500t )

Solution:
For 10 bit PCM
L  2n
n  10
 Number of levels = 1024
The amplitudeAmof sinusoidal waveform means that mp = 4 volts.
The total signal swing possible (-mpto +mp )will be 2mp= 8 volts.
The average signal power is
  Am 2   42 
Pave      8 watts
 2   2 
The interval,
2mp
V 
L
8

1024 levels
 7.81103 volt
Quantization noise,

Nq 
V 2
12
SNR:
 S   Pave 
SNR      8  12
 N   N  V 2
 q  q 

96

6.10  105

 15,73,770

SNRdB  10 log1573770
10  61.96dB

47
 UNIT V - SAMPLING & QUANTIZATION
TWO MARKS
1. What is Communication system?
The Communication System is the system which is used to transport an
information bearing signal from a source to a user destination via a communication
channel.
2. What are different categories of Communication Systems?
 Analog Communication Systems are designed to transmit analog information
using analog modulation methods.
 Digital Communication Systems are designed for transmitting digital information
using digital modulation schemes, and
 Hybrid Systems that use digital modulation schemes for transmitting sampled and
quantized values of an analog message signal.
3. How can BER of an system be improved? [NOV/DEC2012]
Increasing the transmitted signal power Employing modulation and demodulation
technique Employing suitable coding and decoding methods Reducing noise interference with
help of improved filtering.
4. Which parameter is called figure of merit of a digital communication systemand why?
[NOV/DEC 2010]
The ratio Eb/No or bit energy to noise power spectral density is called figure of merit
of a digital communication system
5. Define half power bandwidth. [NOV/DEC2011]
Half power bandwidth is the bandwidth where PSD of the signal drops to half (3dB) of
its maximum value.It is called 3dB bandwidth.
6. What is channel? Give examples. [Nov/Dec 2013]
A channel is used to convey an information signal, for example a digital bit stream,
from one or several senders (or transmitters) to one or several receivers. A channel has a
certain capacity for transmitting information, often measured by its bandwidth in Hz or its
data rate in bits per second.
Ex: Physical transmission medium such as a wire, logical connections over
multiplexed medium such as a radio channel.
7. Draw a typical digital communication system. [Nov/Dec 2012], [Nov/Dec 2011]

48
 8. What are the Advantages of Digital Communication? [Nov/Dec 2013]
 The effect of distortion, noise and interference is less in a digital
communication system.
 Regenerative repeaters can be used at fixed distance along the link, to identify and
regenerate a pulse before it is degraded to an ambiguous state.
 Digitalcircuits are more reliableand cheaper compared to analog circuits.
 Signal processing functions like encryption, compression can be employed to maintain
the secrecy of the information.
 Error detecting and Error correcting codes improve the system performance by
reducing the probability of error.
9. What are Disadvantages of Digital Communication? (or)
State the demerits of digital communication. [May/June 2014]
 Large System Bandwidth:- Digital transmission requires a large system
bandwidth to communicate the same information in a digital format as compared
to analog format.
 System Synchronization:- Digital detection requires system synchronization
whereas the analog signals generally have no such requirement.
10. What is sampling process?
 SAMPLING: A message signal may originate from a digital or analog source. If the
message signal is analog in nature, then it has to be converted into digital form before
it can transmit by digital means.
 The process by which the continuous-time signal is converted into a discrete–time
signal is called Sampling.

SAMPLING THEOREM FOR LOW-PASS SIGNALS:-

11. Define sampling.rate. [Apr - 2019]
The bandpass signal g (t ) whose maximum bandwidth is 2W can be completely
represented into and recovered from its samples if it is sampled at the minimum rate of twice
the bandwidth.

12. Why prefilterring done before sampling? [AUC NOV/DEC 2010]

The signal must be limited to some highest frequency W Hz before sampling. Then the
signal is sampled at the frequency of fs=2W of higher. Hence the single should be prefiltered
at higher that W Hz.If the signal is not prefiltered, then frequency component higher that W
Hz will generate aliasing in the sampled signal spectrum.

49
13. Draw the circuit theoretic representation of ideal sampling process.
This circuit-theoretic interpretation of g (t ) is depicted in Fig. (2)

Figure:Circuit-theoretic interpretation of the ideal sampling process as impulse modulation

where, g (t ) - Modulatiing wave, T s
-Carrier wave and g (t ) -Instantaneously sampled

value

14. Draw the spectrum of (a) analog signal g (t ) (b) Spectrum of sampled signal g (t ) for a

sampling rate f s  2W . (c) Ideal amplitude response of reconstruction filter.

Figure: (a) Spectrum of signal g (t ) . (b) Spectrum of sampled signal g (t ) for a sampling

rate f s  2W . (c) Ideal amplitude response of reconstruction filter.

15. Write about sinc function.

The sinc function exhibits an important property known as the interpolatory property,
which is describes as follows:
1, x  0
sincx = 
0, x  1,2,....

50
16. Draw the block diagram of Reconstruction filter.
Reconstruction filter.

Figure: Reconstruction filter

17. Give the complete statement of sampling theorem. (or)

State sampling theorem for lowpass signals. [May/June 2009] (or)
State sampling theorem. [May/June 2014], [May/June 2012](or)
Define Band pass sampling. [April/May 2018]
Statement of the Sampling Theorem:
1. If a finite-energy signal contains no frequencies higher than W hertz, it is completely
determined by specifying its ordinates at a sequence of points spaced 1/2W seconds apart.
2. If a finite-energy signal contains no frequencies higher than W hertz, it may be completely
recovered from its ordinates at a sequence of points spaced 1/2W seconds apart.
18. Define Nyquist rate and Nyquist interval.
The minimum sampling rate of 2W samples per second, for a signal bandwidth of W
hertz, is called the Nyquist rate.Correspondingly, the reciprocal, 1/2W, is called the Nyquist
interval.
19. What are the different types of sampling?
Types of sampling (Practical Sampling):
1. Ideal sampling
2. Natural sampling
3. Flat top sampling
20. Wirte in short about Ideal Sampling (or)Instantaneous sampling (or) Impulse sampling.
Ideal Sampling (or)Instantaneous sampling (or) Impulse sampling:

Fig (a) Functional diagram of a Fig (b) Message x(t ) and sampled x (t ) signals
switching sampler

51
  Ideal sampling is same as instantaneous sampling.
 Fig. (a)shows the switching sampler.
 If closing time 't' of the switch approaches zero the output x (t ) gives only
instantaneous value. The waveforms are shown in Fig. (b).
 Since the width of the pulse approaches zero, the instantaneous sampling gives train of
impulses in x (t ) . The area of each impulse in the sampled version is equal to
instantaneous value of input signal x(t ) .

21. Write about Natural Sampling (or) Chopper Sampling.

Natural Sampling (or) Chopper Sampling:

Fig. 5.3. Natural sampling

 Although instantaneous sampling is a convenient model, a more practical way of
sampling a band-limited analog signal m(t) is performed by high-speed switching
circuits.
 An equivalent circuit employing a mechanical switch and the resulting sampled signal
are shown in Fig. 5-3(a) and (b),respectively.
 The sampled signal xns (t ) can be written as
xns (t )  m(t ) x p (t ) (5.4)
Where x p (t ) is the periodic train of rectangular pulses with period Ts, and each
rectangular pulse in x p (t ) has width d and unit amplitude.
 The sampling here is termed natural sampling, since the top of each pulse in xns (t )
retains the shape of its corresponding analog segment during the pulse interval.
22. Write about Flat top sampling (or) rectangular pulse shaping.
Flat-Top Sampling (or) Rectangular Pulse Shaping:

Fig. 5.4.Flat-top Sampling

 The simplest and thus most popular practical sampling method is actually performed
by a functional block termed the sample-and-hold (S/H) circuit [Fig. 5-4(a)].
 This circuit produces a flat-top sampled signal xs (t ) [Fig. 5-4(b)].

52
23. Compare Instantaneous, Natural and flat top sampling techniques.
Comparison of Various Sampling Techniques:

24. What is aliasing in sampling process? [May/June 2016], [Nov/Dec 2012]

Aliasing Phenomenon

Fig. (a) Spectrum of a signal. (b) Spectrum of an undersampled version of the signal,
exhibiting the aliasing phenomenon.

53
  Aliasing refers to the phenomenon of a high-frequency component in the spectrum of the
signal seemingly taking on the identity of a lower frequency in the spectrum of its sampled
version, as illustrated in Fig.

25. What are the corrective measures of aliasing effects?[Nov/Dec 2012]

To combat the effects of aliasing in practice, we may use two corrective measures:
1. Prior to sampling, a low-pass anti-alias filter is used to attenuate the high-
frequencycomponents of a message signal that are not essential to the information..
2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

26. Draw the spectrum of (a) Anti-alias filtered spectrum of an information-bearing signal.
(b) Spectrum of instantaneously sampled version of the signal, assuming the use of a
sampling rate greater than the Nyquist rate. (c) Idealized amplitude response of the
reconstruction filter.

Fig 5.4 (a) Anti-alias filtered spectrum of an information-bearing signal. (b) Spectrumof
instantaneously sampled version of the signal, assuming the use of a sampling rate greaterthan the
Nyquist rate. (c)Idealized amplitude response of the reconstruction filter.
Reconstruction of a message process from its samples:

27. Write about the nature of reconstruction of samples.

If a stationary message process contains no frequencies higher than W hertz, it may be
reconstructed from its samples at a sequence of points spaced 1/2Wseconds apart with zero
mean squared error (i.e., Zero error power).

54
Quantization
28. What is meant by amplitude quantization?
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 a discrete amplitude v(nTs) taken from a
finite set of possible amplitudes.
The discrete amplitudes mk,k = 1, 2, ... , L, at the quantizer input are called decision
levels or decision thresholds.

29. Compare uniform and non uniform quantization. [AUC NOV/DEC 2011]
S.NO UNIFORM QUANTIZATION NON QUANTIZATION
1 The quantization step size remains The quantization step size varies with the
samethroughout the dynamic range amplitude of the input signal
of the signal
2 SNR ratio varies with input signal amplitude SNR ratio can be maintained constant

30. What are the types of quantizers?

Quantizers can be of a uniform or nonuniform type. In a uniform quantizer,
therepresentation levels are uniformly spaced; otherwise, the quantizer is nonuniform.

Uniform & non-uniform quantization:

31. Write about uniform quantizer.

a. Uniform quantizer
b. Nonuniform quantizer

FIGURE.Two types of quantization: (a) midtread and (b) midrise.

 In a uniform quantizer, the representation levels are uniformly spaced; otherwise, the
quantizer is nonuniform.

32. Write a note on uniform quantizer.

Uniform Quantization
 The quantizer characteristic can also be of a midtread or midrise type.

55
 Figure (above) shows the input–output characteristic of a uniform quantizer of the midtread
type,which is called as uniform, because the origin lies in the middle of a tread of the
staircase-like graph.

33. Write about Nonuniformquantizer.

Nonuniform Quantization
 The nonuniformquantizer with the feature that the step size increasesas the separation from the
origin of the input–output amplitude characteristic isincreased, the large end-step of the
quantizer can take care of possible excursions ofthe voice signal into the large amplitude
ranges that occur relatively infrequently.
34. Write about   law .
A particular form of compression law that is used in practice is the so called   law defined
by

(01)
where the logarithm is the natural logarithm; m and are respectively the normalized input and
output voltages, and  is a positive constant.

35. Write about A  law .

 One of the compression law that is used in practice is the A-law, defined by

(03)
which is shown plotted in Fig. 5.12(b). Typical values of A used in pratice tend to be in the
vicinity of 100. The case of uniform quantization corresponds to A  1 .

 The reciprocal slope of this second compression curve is given by the derivative of m with

respect to v as shown by

(04)

56
Quantization noise

36. Draw the diagram to illustrate Quantization process.

Fig. 3.11. Illustration of Quantization process

37. Write about Quantization noise.
 The use of quantization introduces an error defined as the difference between the input signal
m and the output signal v. The error is called quantization noise.
 Figure 3.11 illustrates a typical variation of the quantization noise as a function of time,
assuming the use of a uniform quantizer of the midtread type.
38. Write about the quantizer step-size.
 Consider then an input m of continuous amplitude in the range (-mmax, mmax).
 Assuming a uniform quantizer of the midrise type, we find that the step-size of the quantizer is
given by

2mmax
 (04)
L

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.
39. What will happen if the quantization step size is smaller?
If the step-size  is sufficiently small (i.e., the number of representation levels L is
sufficiently large), it isreasonable to assume that the quantization error Q is a uniformly
distributed random variable, and the interfering effect of the quantization noise on the
quantizer input is similar to that of thermal noise.

40. Write the expression for probability density function. (or)

Derive the expression for quantization noise of a PCM system. [Nov 2017]
The expression for the probability density function of the quantization error Q as follows:

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 41. Write the expression for the output SNR of a uniform quantizer.
Let P denote the average power of the message signal m(t). We may then express the
output signal-to-noise ratio of a uniform quantizer as
P
( SNR)O =
 Q2
3P 2 R
= 2
2
mmax
Logarithmic Companding of speech signal

42. What is companding? [May/June 2016]

 Companding is simply a system in which information is first compressed, transmitted through
a bandwidthlimited channel, and expanded at the receiving end.
 It is frequently used to reduce the bandwidth requirements for transmitting telephone quality
speech, by reducing the 13-bit codewords to 8-bit codewords.

Pulse Amplitude Modulation

43. What is PAM?write its types.

 The amplitude of the pulse carrier is changed in proportion with the instantaneous amplitude
of the modulating signal.
Types of PAM
Depending upon the shape of the PAM pulse, there are two types of PAM. They are:
(iii) Natural PAM
(iv) Flat top PAM

44. Why flat top PAM is widely used? [Dec – 2016]

 During the transmission, the noise interferes with the flat top of the transmitted pulses and this
noise can be easily removed.
 In natural samples PAM, the pulse has varying top in accordance with the signal variation.
 When such type of pulse is received by the receiver, it always seems to be contaminated by
noise.
 Then it becomes quite difficult to determine the shape of the top of the pulse and therefore
amplitude detection of those pulses is not exact.
 As a result of this, errors are introduced in the received signal.
 The electronic circuitry needed to perform natural sampling is somewhat complicated because
the pulse top shape is to be maintained. These complications are reduced by flat-top PAM.

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 45. What are the advantages and disadvantages of PAM?
Advantage: Simple generation and detection
Disadvantages:
 Effect of additive noise is high in PAM.
 Transmission bandwidth required is too large.
 The transmission power is not constant due to the changes in amplitudes of PAM
pulses.
Pulse-Time Modulation
46. What is Pulse-Time Modulation and its types?
 In pulse time modulation, amplitude of pulse is held constant, whereas position of
pulse is made proportional to the amplitude of signal at the sampling instant.
There are two types of pulse time modulation. They are:
 Pulse width modulation
 Pulse position modulation
47. Define Pulse Width Modulation (PWM)
 The width of the carrier pulses varies in proportion with the amplitude of modulating signal.
 The amplitude and frequency of the PWM wave remains constant.
 Only the width changes.
 The information is contained in the width variation.
 The additive noise, changes the amplitude of the PWM signal.
 Using the limiter circuit at the receiver, unwanted amplitude variations are easily removed.
48. Draw the waveform of PWM.

Fig: PWM signal

 Amplitude variations due to noise do not affect the performance. Thus PWM is more immune
to noise than PAM.

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49. Draw the block diagram and waveform of PWM and PPM.

Fig : Block diagram of PWM and PPM generation

Fig : PWM and PPM waveforms

50. Draw the Block diagram of PWM detection circuit.

Fig: Block diagram of PWM detection circuit

51. List the advantages and disadvantages of PWM.
Advantages
 In PWM noise is less because here amplitude is constant.
 No synchronization required between transmitter and receiver.
 It is easy to separate the signal from noise.
Disadvantages
 Variable pulse width causes variable power contents. So, transmission must be powerful
enough to handle the maximum width.
 Bandwidth requirement is higher than PAM.

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 52. Define Pulse Position Modulation (PPM).
The amplitude and width of the pulses are kept constant but the position of each pulse is
varied in accordance with the amplitude of the sampled values of the modulating signal.
53. Draw PPM demodulator circuit.

Fig : PPM demodulator circuit

54. List the advantages and disadvantages of PPM.
Advantages
 Due to constant amplitude of pulses, the transmitted power always remains constant.
 It is easy to reconstruct PPM signal from the noise contaminated PPM signal.
Disadvantages
 Synchronization required between the transmitter and receiver.
 Large bandwidth requirement.
55. Difference Between PAM, PWM, and PPM.

The below table gives the detailed difference between PWM, PAM, and PPM.

Sr. No. Parameter PAM PWM PPM

1 Type of Carrier Train of Pulses Train of Pulses Train of Pulses

Variable Characteristic
2 of the Pulsed Carrier Amplitude Width Position

Bandwidth
3 Requirement Low High High

4 Noise Immunity Low High High

Information Contained Amplitude Position

5 in Variations Width Variations Variations

6 Power efficiency (SNR) Low Moderate High

Varies with Varies with Remains

7 Transmitted Power amplitude of pulses variation in width Constant

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Pulse-Code Modulation
56. What is Pulse code modulation?
Pulse code modulation:
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.
57. What are the basic operations performed in PCM?
The basic operations performed in the transmitter of a PCM system are sampling,
quantization, and encoding; the low-pass filter prior to sampling is included merely to prevent
aliasing of the message signal.
58. Write about quantization process in PCM.
The quantizing and encoding operations are usually performed in the same circuit,
which is called an analog-to-digital converter.

59. What are the operations performed in PCM receiver?

 The basic operations in the receiver are regeneration of impaired signals, decoding,
and reconstruction of the train of quantized samples.
 Regeneration also occurs at intermediate points along the transmission path as
necessary.
60. Write about sampling in PCM.
Sampling in PCM
 The incoming message (baseband) signal is sampled with a train of rectangular pulses, narrow
enough to closely approximate the instantaneous sampling process.
 Thus the application of sampling permits the reduction of the continuously varying message
signal (of some finite duration) to a limitednumber of discrete values per second.
61. What is the purpose of ternary code used in PCM?
The two symbols of the binary code are customarily denoted as 0 and 1. In practice, a binary
code is preferred over other codes (e.g.,ternary code) for two reasons:
1. Binary symbol withstands a relatively high level of noise.
2. The binary code is easy to generate and regenerate.
62. Draw the block diagram of Pulse code modulation.

FIGURE. The basic elements of a PCM system

(a) Transmitter, (b) transmission path,connecting the transmitter to the receiver, and (c) receiver.

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63. In a PCM system, the output of the transmitting quantizer is digital. Then why is it
further encoded. [Nov 2017, May 2018]
In a PCM system, the output of the transmitting quantizer is digital. It is required to
translate the discrete set of sample values to a more appropriate form of the signal. So it is
further encoded.

64. Write about the regeneration in the transmission path of PCM.

Regeneration along the Transmission Path
Three basic functions are performed by a 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.

65. What are reasons for the regenerated signal departs from the original signal?
 In practice, however, 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, thereby introducing bit errors into the regeneratedsignal.
2. If the spacing between received pulses deviates from its assigned value, a jitter is
introduced into the regenerated pulse position, thereby causing distortion.

Fig. 5.13. Block diagram of Regenerative repeater

66. Explain the reconstruction process in PCM.
Reconstruction
 The final operation in the receiver is to recover the message signal.
 This operation isachieved by passing the expander output through a low-pass
reconstruction filterwhose cutoff frequency is equal to the message bandwidth.
 Recovery of the messagesignal is intended to signify estimation rather than exact
reconstruction.

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Time Division Multiplexing:
67. What is the need for TDM system? [Apr - 2019]
A time-division multiplex (TDM) system, which enables the joint utilization of a common
communication channel by a plurality of independent message sources without mutual
interference among them.
68. Draw the block diagram of TDM system.

FIGURE 5.21 Block diagram of TDM system.

69. What is the function of commutator?

The function of the commutator is twofold:
(1) to take a narrow sample of each of the N input messages at a rate that is slightly higher
than 2W, where W is the cutoff frequency of the anti-aliasing filter, and
(2) to sequentially interleave these N samples inside the sampling interval Indeed, this latter
function is the essence of thetime-division multiplexing operation.

70. Define Frequency-Division Multiplexing (FDM) .

 Frequency-Division Multiplexing (FDM) is a scheme in which numerous signals are
combined for transmission on a single communications line or channel.
 It is analog multiplexing technique. Each signal is assigned a different frequency (sub
channel) within the main channel. its requires channel synchronization.
71. Draw the block diagram of FDM.

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72. List the advantages and disadvantages of FDM.
Advantages of FDM:
1. A large number of signals (channels) can be transmitted simultaneously.
2. FDM does not need synchronization between its transmitter and receiver for proper
operation.
3. Demodulation of FDM is easy.
4. Due to slow narrow band fading only a single channel gets affected.

Disadvantages of FDM:
1. The communication channel must have a very large bandwidth.
2. Intermodulation distortion takes place.
3. Large number of modulators and filters are required.
4. FDM suffers from the problem of crosstalk.
5. All the FDM channels get affected due to wideband fading.

73. Mention the applications of FDM.

Applications of FDM
1. FDM is used for FM & AM radio broadcasting.
2. FDM is used in television broadcasting.
3. First generation cellular telephone also uses FDM.

****

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