MODULE-IV

Analog Pulse Modulation System

 In the pulse modulation systems however the carrier is a train of discrete pulses rather than a sinusoidal as in
AM FM and PM.
 Any one characteristics of this pulse train, i.e. amplitude, width or position can be changed in proportion with
the amplitude of modulating signal. Such as PAM, PWM, PPM signal.
Applications of sampling Techniques
1) Pulse Amplitude modulation
(I) Naturally Sampled PAM
 In the PAM system, the amplitude of the pulse carrier is changed in proportion with instantaneous amplitude
of the modulating signal.
 The carrier is in the form of train of narrow pulses and is identical to natural sampling process.
 The PAM signal is then sent by either wire or cable or it is used to modulate another carrier.
Generation of PAM Signal

 The LPF will band limited the signal to W. i.e. all the frequencies higher than the frequency W are removed.
Band limiting is necessary to avoid the aliasing.
 Pulse Train generator generates a pulse train at a frequency fs, such that fs > 2W.All these pulses are equal
width.
 The uniform sampling takes place in the multiplier block to generate the PAM signal.
 The information in the modulating signal is contained in the amplitude variation of the pulse carrier.
Therefore, this system is similar to an AM system.
Types of PAM: - There are two types of PAM systems, Such as
(i) Single Polarity PAM
(ii) Double Polarity PAM

Detection/Demodulation of PAM Signal

 PAM signal can be demodulated by passing it through a LPF.

 The LPF cut – off frequency is adjusted to W so that all high frequency components are removed and the
original modulating signal is recovered back.
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Spectrum of Naturally Sampled PAM
Naturally sampled PAM is same as natural sampling and it can be expressed mathematically as
Naturally Sampled PAM:
 A  n 
s(t )   x(t )  Sinc e j 2 (nfs)t
n  
Ts  Ts 
We know that for a pulse signal of width „τ‟ and amplitude „A‟ and the time period „Ts‟, the exponential form of F.S.
  A
 n 
is c(t )    Sinc e j 2 (nfs)t
n  
Ts  Ts 
A  n 
Where Dn   Sinc 
To  To 
  A 
 n  A  n 
Then s (t) = x (t) ×c (t) = x(t )    Sinc e j 2 (nfs)t   x(t )  Sinc e j 2 (nfs)t
n  
Ts  Ts  n  
Ts  Ts 
Spectrum of Naturally Sampled PAM:-
  A  n  
S (f) = FT[s (t)] = FT   x(t )  Sinc e j 2 (nfs)t 
 Ts  Ts  
n   
  A
 n 
=   Sinc  FT  x(t )e j 2 (nfs)t 
n  
Ts  Ts   
  A 
 n   
=   Sinc    x(t )e j 2 (nfs)t e  j 2ft dt 
n  
Ts  Ts   
  A 
 n   
=   Sinc    x(t )e  j 2 ( f  nfs)t dt 
n  
Ts  Ts   

A    n 
=
Ts
 Sinc  X ( f  nfs)
 Ts 
n  
(I) Flat Top Sampled PAM
This is same as flat top sampling.

Flat Top sampling PAM: - s (t )   x(nTs)h(t  nTs)
n  

2
 
fs  X ( f  nfs)H ( f )
Spectrum of Flat Top Sampled PAM:- S (f) = FT[s (t)] = n 
H ( f ) = Spectrum of rectangular pulse h (t) which is a Sinc function
Transmission Bandwidth of PAM signal
Let τ = Width of the each pulse in a flat top sampled PAM signal
Ts = Duration between adjacent samples
τ is very small as compared to Ts
Therefore, τ << Ts
But Ts = 1/fs where fs = sampling frequency
1
If W is maximum frequency in x (t) then fs  2W. Hence, Ts 
2W
And τ << Ts
1 1
τ << ; W <<
2W 2
 Adequate pulse resolution i.e. to transmit and receive this PAM signal without much signal distortion, the
1
transmission band width (BT) needs to satisfy the following equation. BT   W OR BT  W
2
Q.1 For a PAM transmission of a voice signal with W = 3KHz.Calculate the transmission band width (B T) if
the width of each pulse, τ = 0.1Ts and the sampling frequency fs = 8 KHz.
1 1
Solution: - Ts =  = 0.125mSec
fs 8KHz
τ = 0.1Ts = 0.1×0.125mSec = 0.0125mSec.
1 1 1
The transmission bandwidth (BT) is given as BT  ; BT  =  40 KHz
2 2  0.0125mSec 0.025mSec
BT = 40 KHz
2) Pulse Width Modulation
 In PWM, the width of the modulated pulses varies in proportion with the amplitude of modulating signal.

 The amplitude and frequency of PWM wave remains constant and only the width will changes. Information is
contained in the width variation and is similar to FM.
 Noise is normally additive noise; it changes the amplitude of the PWM signal.
 At the receiver, it is possible to remove these unwanted amplitude variations very easily by means of a limiter
circuit.
 As the information is contained in the width variation, it is unaffected by the amplitude variations introduced
by noise. Hence the PWM is more immune to noise than the PAM signal.
Generation of PWM and PPM Signal

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 (i) Generation of PWM Signal
 A saw tooth wave generator generates a saw tooth signal of frequency fs, as sampling signal. It is applied to
the inverting terminal of a comparator.
 The modulating signal is applied to the non – inverting terminal of the same comparator.
 The comparator output will remain high as long as the instantaneous amplitude of x (t) is higher than that of
the Ramp signal. This gives rise to a PWM signal at the output of comparator as shown in fig.
 The leading edge of the PWM waveform coincide the falling edge of the ramp signal. However, the
occurrence of its trailing edges will dependent on the instantaneous amplitude of x (t).Therefore, this PWM
signal is said to be Trail Edge Modulated PWM.
(ii) Generation of PPM Signal
 The PWM Pulses obtained at the output of the comparator are applied to a mono stable multi vibrator.
 This Mono stable Multi vibrator is negative edge triggered. Therefore, corresponding to each trailing edge of
the PWM signal, the mono stable output goes high.
 It remains high for a fixed time decided by its RC components. Thus, as the trailing edges of the PWM signal
keep shifting in proportion with the modulating signal x (t).
 All the PPM signals are of same amplitude and width.

Advantages & Disadvantages of PWM

Advantages:-
1) Less effect to noise i.e. very good noise immunity.
2) Synchronization between Rx and Tx is not essential.
3) It is possible to reconstruct the PWM signal from a noise contaminated PWM. Thus it is possible to separate
out signal from noise.
Disadvantages:-
1) Due to the variable pulse width, the pulses have variable power contents. So, the transmitter must be powerful
enough to handle the maximum width pulse.
2) The average power transmitted can be as low as 50% of this maximum power.
3) In order to avoid any waveform distortion, the BW for the PWM communication is large as compared to BW
of PAM.

Advantages & Disadvantages of PPM

Advantages:-
1) Due to constant amplitude of PPM pulses, the information is not contained in the amplitude. Hence, the noise
added to PPM signal does not distort the information. Thus it has good noise immunity.
2) It is possible to reconstruct PPM signal from the noise contaminated PPM signal.
3) Due to constant amplitude and width of the pulses, the transmitted power always remains constant.
Disadvantages: -
1) As the position of the PPM pulses is varied w.r.t. a reference pulse, a transmitter has send synchronizing
pulses to operate timing circuits in the receiver. Without them the demodulation will not possible to achieve.
2) Large BW is required to ensure transmission of undistorted pulses.

3) Pulse Code Modulation (PCM)

 PCM is a digital pulse modulation system, whose output is in the coded digital form whereas PAM, PWM,
PPM are analog pulse modulation system.

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  In PCM, the form of digital pulses of constant amplitude, width and position. The information in the form of
“code words”.
 A PCM system consists of a PCM encoder (Tx) and a PCM decoder (Rx).
(A) PCM Transmitter (Encoder) Tx

Operation:-
 The analog signal x (t) is passed through a band limited Low Pass Filter, which has a cut – off frequency fc =
W and which eliminate the possibility of aliasing.
 The band limited analog signal is then applied to a sample and hold block circuit where it is sampled at
adequately high sampling rate. Output of Sample and Hold block is a flat top PAM signal.
 The samples are then subjected to the operation called “Quantization” in the Quantizer. Quantization process
is the process of approximation is used to reduce the effect of noise.
 The Quantized PAM signal pulses are applied to an encoded which is basically an A to D converter.
 Each Quantized level is converted into an N bit word as M = 2N.
 The encoder output is converted into a stream of pulses by the parallel to serial converted block. Thus at the
PCM Transmitter output we get a train of digital pulses.

(B) PCM Receiver (Encoder) Rx

 The Quantizer at the receiver will separate the PCM pulses from noise and will reconstruct original PCM
signal.
 The pulse generator has to operate in synchronization with that at the Transmitter. Thus at the quantizer output
we get a clean PCM signal.
 The reconstructed PCM signal is passing through a serial to parallel convertor which is then applied to a
decoder (D to A Converter) which performs an inverse operation of encoder.
 The decoder output is a sequence of quantized multi-level pulses. The quantized PAM signal thus obtained at
the output of the decoder.
 This Quantized PAM signal is passed through a LPF to recover the analog signal.
Signaling rate and Transmission BW of PCM
We know, Q = 2N where Q = No. of Quantization levels
N = No. of bits per word
The input signal is sampled at the sampling rate fs, i.e. there is fs no. of samples per second. Each of these samples is
then converted into an N bit digital word.
 No. of bits per second = No. of samples per second × No. of bits per sample = fs × N
Signaling rate is nothing but the no. of bits per second.
 Signaling rate for PCM = Nfs
The Transmission bandwidth of PCM is equal to half the signaling rate.

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 Nfs
 Transmission BW of PCM signal =
2
Applications of PCM
(1) In telephony (In advent of fiber Optics Cable).
(2) In space communication where a space craft transmits signals to earth .Here the transmitted power very low
(10W to 15W) and the distance are huge (a few million km).still due to high noise immunity only PCM
systems can be used in such applications.

Advantages of PCM
(1) Very high noise immunity.
(2) Due to digital nature of the signal, repeaters can be placed between Tx and Rx. The repeaters actually
regenerate the received PCM signal. This is not possible in analog systems.
(3) It is possible to store the PCM signal due to the digital nature.
(4) It is possible to use various coding techniques so that only the desired person can decode the received signal.

Disadvantages of PCM
(1) The encoding, decoding and quantizing circuitry of PCM is complex.
(2) PCM requires the large BW as compared to the other systems.
Quantization Process
 Quantization is a Process of approximate or rounding off. Quantizer converts the sampled signal into an
approximate quantized signal which consists of only a finite number of predefined voltage levels.
 Each sampled value at the input of quantizer is approximated or rounded off to the nearest standard predefined
voltage levels. These standard levels are known as the “quantization levels”.

 The input signal x (t) is assumed to have a peak to peak swing of V L to VH volts. This entire voltage range has
been divided into „Q‟ equal intervals each of size “S”.
VH  VL
 “S” is called as the step size and its value is S = where Q = 8 = 2N in fig.
Q
 At the center of these steps, quantization levels q0, q1, ---q7 are located.
 xq (t) represents the quantized version of x (t), when x (t) is in the range Δ0. Then corresponding to each value
of x (t), the quantizer o/p will be equal to q .similarly, for others, the quantized signal as shown in fig.
 The quantized signal xq (t) is thus an approximate of x (t). The difference between them is called as
quantization error or quantization noise.
 This error as small as possible, to minimize the quantization error we need to reduce the step size “S” by
increasing the number of quantization levels Q.
Types of Quantization
(i) Uniform Quantization:-
 A quantizer is said to be a uniform quantizer if the step size remains constant throughout the input range.

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 (ii) Non – Uniform Quantization:-
 If the quantizer characteristic is non – linear and the step size is not constant instead if it varies depending on
the amplitude of input, then the quantization is called as “Non – Uniform Quantization”.
Derivation of Expression for the Quantization Error
 The input signal x (t) varies between the voltage levels VH and VL.
Therefore, the total variation of input in amplitude = VH – VL
Let VH = V & VL = – V
Then total change in amplitude = 2V
V H  V L 2V
If this range is divided into „Q‟ levels of quantization, then the step size is S = 
Q Q
If VH = + 1volt and VL = – 1volt ; Then S =2/Q
 If the step size is assumed to be sufficiently small then the quantization error can be assumed to have
distributed uniformly and we can say that the quantization error is a random variable with “uniform
distribution”.
S S S
 Maximum quantization error is  .Therefore; we can say that over the range  to  , quantization error
2 2 2
is a uniformly distributed random variable.

 The probability density function (PDF) for the quantization error (Є) is defined as
 s
 0 for   2
 1 S S
f Є (Є) =  for    
S 2 2
 0 for  S
 2
 The mean value or average value of quantization error is zero.

Mean square value = E (Є ) =  2  f ( )d
 2
 2
=

S S
 PDF E (Є2) exists only over the range  to 
2 2

7
 S / 2

S
1  3 1 S3 S3  1 S3  S2

2 =  f  ( )d   d  
2 2
21
 =   =  =
S S 3  S  24 24  S  12  12
  S
2
  S / 2    

S2
 Mean Square Value of Quantization noise voltage is = Vn2
12
S2
 Normalized Quantization Noise Power, Nq = for linear quantization
12
Expression for the Maximum Signal to Quantization Error Ratio(S/Nq)
Si NormalizedSignalPower
 The signal to quantization noise ratio can be defined as 
Nq NormalizedNoisePower

 The power distribution in a resistance R by the signal voltage x (t) is given by Si =

rmsVoltage2
R
RMS voltage of x (t) = V
2

2
V 
  V2
 Signal Power Si =
 2

R 2R
 T 
 RMSVoltage  1 x (t )2 dt 
 T 0 
 
 T 2 T 
 1
  V 
T 0 2T 0
VSin t 2
dt  (1  Cos 2 t )dt
 
 
 V2 V 
 T  
 2T 2 
 
V2
 Normalized Signal Power can be obtained by R = 1, i.e. Therefore, Normalized Signal Power, Si =
2
V2
Si NormalizedSignalPower 6V 2 2V
   22  2 But S =
Nq NormalizedNoisePower S S Q
12
Si 6V 2 6V 2 3 2 3 N 2 3 2N
  2  2
 Q  (2 )  2
Nq S  2V  2 2 2
 Q 
3 2N
 Therefore, Maximum Signal to Quantization noise ratio  2
2
 Again, Signal to quantization Noise Ratio in dB
 Si 

3 2N  3
  10 log10   2   10 log10    10 log10 2
2  2
2N
 
 10  0.176  20 N log10 2
 N q  dB
 10  0.176  20N  0.301  1.76  6N  1.8  6 N dB (For sinusoidal input signal)
 The signal to quantization noise ratio for a non – sinusoidal input signal is give as
8
  Si 
Maximum    4.8  6 N dB
 N q  dB

Problems on PCM
Q.1 A voice signal band limited to 3.4 KHz is to be transmitted using PCM systems. The signaling rate of
PCM is not exceeding 36000bits/sec.
Find: - a) Approximate value of fs
b) The number of quantization levels Q
c) Number of Digits (bits) per word N
Solution: - It is given that, r (signaling rate)  36000bits/sec.
 Nfs = 36000
Minimum sampling frequency fs (min) = 2fm =2×3.4 KHz = 6.8 KHz
36000
N  5.29
6.8 KHz
N  5.29
Let N = 5
Maximum allowable value of sampling frequency fs (max)
36000 36000
fs (max) =   7.2 KHz
N 5
a) Approximate value of fs should be in between 6.8 KHz and 7.2 KHz
b) The number of quantization levels Q = 2N = 32
c) Number of Digits (bits) per word N=5
Q.2 An analog signal is quantized and transmitted by using a PCM system. If each sample at the receiving
end of the system to be known to within  0.5% of the peak to peak full scale value, how many binary
digits must each sample contain?
Solution: - Let 2A be the peak to peak value of the signal.
 The maximum allowable error =  0.5% of 2A
=  0.005 of 2A=  0.01A
S
Maximum error introduced in PCM system is Єmax =  where S = step size
2
S
 =  0.01A
2
S =  0.02A
2A 2A
We know, Q =   100
S 0.02 A
Q = 100 = 2N
log 10100 10
N=   6.64
log 102 log 102
As we want error to be less than  0.5%
Let the round off N to the next integer
 No. of binary digit per word, N = 7.
Q.3 In a Binary PCM system, the output signal to quantization noise ratio is to be held to a minimum of
40dB. First calculate the no. of binary digits per word, necessary to meet this requirement and then find
the actual value of the o/p signal to quantization noise ratio.
Solution: - Assume the signal to be sinusoidal,
 S 
Max   = 1.8 + 6N
 Nq 
  dB
The minimum value of signal to quantization noise ratio is 40dB
 S  40dB
N
(1.8 + 6N) dB  40dB
6NdB  40dB –1 .8dB = 38.2dB
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 N  6.36
Hence let us have N = 7
 S 
Actual value of   = 1.8 + 6N = 1.8 + 6×7 = 43.8dB
 Nq 
  dB
Q.4 An audio signal has spectral components presents in the range of 300Hz to 3300Hz. A PCM signal is
generated by sampling this audio signal at fs = 8 KHz. The minimum value of signal to quantization
noise ratio is 30dB.
Calculate:
(i) The minimum number of quantization levels Q and the number of binary digits per word N.
(ii) Signaling rate ‘r’.
(iii) Minimum Transmission BW.
 S 
Solution: - Given fs = 8 KHz and    30dB
 Nq 
  dB
 S 
Assume the input signal is sinusoidal, Then   = (1.8 + 6N) dB  30dB
 Nq 
  dB
N  4.7
Therefore, Number of digits per word, N = 5
Q = 2N = 25 = 32

Signaling Rate (r): - r = Nfs = 5 × 8 KHz = 40 KHz = 40 K Bits/sec

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Transmission Bandwidth: - BT = r = 20 KHz
2
Q.5 A PCM system uses a uniform quantizer followed by a 7bit encoder .The system bit rate is 50MBits/sec.
Calculate the maximum bandwidth of the message signal for which this system operates satisfactorily.
Solution: - r = 50 MBits /sec and N = 7
r = Nfs
r 50  10 6 fs 7.14MHz
fs =   7.14MHz Maximum Signal bandwidth = =  3.57 MHz
N 7 2 2
Q.6 The Bandwidth of a video signal is 4.5MHz.This signal is to be transmitted using PCM with the number
of quantization levels Q = 1024.The sampling rate should be 20% higher than the Nyquist rate.
Calculate the system bit rate.
Solution: - Bandwidth, W = 4.5MHz; Nyquist Rate = 2W = 2×4.5MHz = 9MHz
But fs should be 20% higher than the Nyquist rate, i.e. fs = 1.2× Nyquist Rate =1.2×9MHz =10.8MHz
We know that, Q = 2N = 1024 = 210 ; Therefore, N = 10
 The system bit rate, r = Nfs = 10×10.8MHz =108 MHz = 108 MBits/sec
Q.7 Derive the expression for the signal to quantization noise ratio of PCM system employing uniform
quantization technique. Assume that the input signal is of non – sinusoidal nature.
Si NormalizedSignalPower
Solution: - The signal to quantization noise ratio can be defined as 
Nq NormalizedNoisePower
S2 PeaktoPeakSignalAmplitude 2 x max 2 x max
 Normalized Noise Power = Nq = where S =  
12 Q Q 2N
2
 2 x max  4 x 2 max
2  
 Nq =
S
=
 2 N
  22N 
x 2 max
12 12 12 3  22N


Si
P
x 2 max = P  3  2 2 N  3P  2 2 N
Nq 3  22N x 2 max x 2 max
Si 3P
Therefore, Signal to quantization noise ratio for non – sinusoidal Signal  22N
Nq 2
x max
If the input signal x (t) is normalized, then x max  1
10
  3  2 2 N  P and if the signal power “P” is also normalized at the destination. Then P  1,
Si
Hence
Nq
Si
  3  22N  P  3  22N
Nq
Si
  3 2 2 N
Nq
 Si 
  
 Nq 

 10 log10 3  2 2 N dB 
  dB
 Si 
  
 Nq 
 
 10 log10 3dB  10 log10 2 2 N dB  4.8  6 N dB
  dB
 Hence, maximum signal to quantization noise ratio for normalized power P and 1/P amplitude x (t):
 Si 
   4.8  6 N dB
 Nq 
  dB

Companding (Companded PCM)

 Practically it is difficult to implement the non – uniform quantization because it is not known in advance about
the changes in the signal level.
 Therefore, the weak signals are amplified and strong signals are attenuated before applying them the uniform
quantization. This process is called as “Compression” and the block that provides is called as a “Compressor”.
 At the receiver exactly the opposite process is followed which is called “Expansion”. The block which
provides the expansion is called “Expander”.
 The Compression at the transmitter and Expansion at the receiver is combined to be call as “Companding”.
Need of Companding: -

 For the input signals of smaller amplitudes, the signal power is low. However due to the uniform quantization,
the step size and hence the quantization noise power is constant.
 This results in reduction in the signal to quantization noise ratio for weak signals which is not desirable.
 Companding is used to improve the SNRq of the weak signals.
 Companding = Compressing + Expanding
Compression & Expander Characteristics:-

 Compressor provides a higher gain to the weak signals and smaller gain to the stronger signals. Thus weak
signals are artificially boosted to improve the SNRq ratio.
 The characteristic of the expander is the inverse of the compressor characteristics. This insures that the overall
characteristic is a straight line (dotted line). This indicates that all the boosted signals are brought down to
their original amplitudes, at the receiver.

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Types of Compressor Characteristics

 Ideally we need a linear compressor characteristic for small amplitudes of the input signal and a logarithmic
characteristic elsewhere. Practically this is achieved by using two methods:
(i) μ – law Companding
(ii) A – law Companding
(i) μ – law Companding

 In the μ – law Companding, the compressor characteristics is continuous. It is approximately linear for small
value of input levels and logarithmic for high input levels.
ln(1   x )
 The μ – law Compressor characteristics is mathematically expressed as Z (x) = (sgn x) , x 1
Ln(1   )
 The practically used value of μ is 255
 The characteristics corresponding to μ = 0 corresponds to the uniform quantization.
 μ –Law Companding is used for speech and music signal.
(ii) A – law Companding
 In the A – Law Companding, the compressor characteristics is piecewise, made up of a linear segment for low
level inputs and a logarithmic segment for high level inputs.

 When A = 1, the characteristic is linear which corresponds to uniform quantization.

 The practically used value of A = 87.56 and is used for telephone systems.
 Compressor characteristics for A – Law compressor is given by

12
  Ax
 x
 x max , for0 
 1  log e A x max  1
z ( x)  
x max 1  log  A x 

 e x max x
   , for 1


 1  log e A A x max  1
Inter Symbol interference, Eye patterns, Equalization
 In telecommunication, Inter Symbol interference (ISI) is a form of distortion of a signal in which one symbol
interferes with subsequent symbols. This is an unwanted phenomenon as the previous symbols have similar
effect as noise, thus making the communication less reliable.

(A) INTER SYMBOL INTERFERENNCE

 Among the different types of distortions in communication Inter Symbol Interference is a phenomenon that
causes the heavy data loss in the communication.
 Usually the digital information that is transmitted will be in the form of square waveform representing the 1‟s
and 0‟s.
 When this square waveform mixes with the noises and non linearities in the channel, the square waveform
starts to spread and merge with the adjacent symbol sequence, making the data there to be unreadable.
 At the receiver end this data is wrongly decoded as the receiver cannot predict the correct level of the square
waveform leading to the loss of information.
 ISI is usually caused due to multipath propagation of the signal in band limited channel and the non-linear
frequency response of the channel.
 The Fig 1 represents the graphical representation of ISI.The presence of ISI in the system introduces errors in
the decision device at the receiver output.
 Therefore, in the design of the transmitting and receiving filters, the objective is to minimize the effects of ISI,
and thereby deliver the digital data to its destination with the smallest error rate possible.

CAUSES OF INTER SYMBOL INTERFERENCE:-

(1) Multipath propagation:-
 One of the causes of Inter Symbol Interference is multipath propagation in which a wireless signal from a
transmitter reaches the receiver via many different paths.
 The causes of this include reflection refraction (such as through the foliage of a tree) and atmospheric effects
such as atmospheric ducting and ionospheric reflection.
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  Since all of these paths are different lengths - plus some of these effects will also slow the signal down - this
results in the different versions of the signal arriving at different times.
 This delay means that part or all of a given symbol will be spread into the subsequent symbols, thereby
interfering with the correct detection of those symbols.
 Additionally, the various paths often distort the amplitude and/or phase of the signal thereby causing further
interference with the received signal.
(2) Band-limited channels:-
 Another cause of Inter Symbol Interference is the transmission of a signal through a band-limited channel, i.e.,
one where the frequency response is zero above the cutoff frequency.
 Passing a signal through such a channel results in the removal of frequency components above this cutoff
frequency; in addition, the amplitude of the frequency components below the cutoff frequency may also be
attenuated by the channel.
(3) Noises and Fading Channels:-
 Noise may be defined as any unwanted signal that interferes with the communication, measurement or
processing of an information-bearing signal.
 Noise is present in various degrees in almost all environments.
 For example, in a digital cellular mobile telephone system, there may be several variety of noise that could
degrade the quality of communication, such as acoustic background noise, thermal noise, electromagnetic
radio-frequency noise, co-channel interference, radio channel distortion, echo and processing noise.
 Noise can cause transmission errors and may even disrupt a communication process.

A Communication Channel is known as fading channel if in which it has to face different fading phenomenon, during
signal transmission. In real world environment, the radio propagation effects combine together and multipath is
generated by these fading channels. Due to multiple signal propagation paths, multiple signals will be received by
receiver and the actual received signal level is the vector sum of the all signals.

Among the noises, the following three are the noises that widely affect the digital communication. They are Additive
White Gaussian Noise (AWGN), Rayleigh Fading, and Rician Fading. These noises and fading cause huge loss of data
in digital communication. And fading causes a major error called Inter Symbol Interference (ISI).

(B) EYE PATTERNS

 In telecommunication, an eye pattern, also known as an eye diagram, is an oscilloscope display in which a
digital signal from a receiver is repetitively sampled and applied to the vertical input, while the data rate is
used to trigger the horizontal sweep.


 The eye diagram takes its name from the fact that it has the appearance of a human eye. It is created simply
by superimposing successive waveforms to form a composite image.
 The eye diagram is used primarily to look at digital signals for the purpose of recognizing the effects of
distortion and finding its source.
(C) EQUILIZATION
 Equalization or equalisation is the process of adjusting the balance between frequency components within an
electronic signal.
 The most well-known use of equalization is in sound recording and reproduction but there are many other
applications in electronics and telecommunications.
 The equalizer is a device that attempts to reverse the distortion incurred by a signal transmitted through a

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 channel.
 In digital communication its purpose is to reduce inter symbol interference to allow recovery of the transmit
symbols. It can be a simple linear filter or a complex algorithm.
TDM-PCM

 Time-division multiplexing(TDM) is a method of transmitting and receiving independent signals over a

common signal path by means of synchronized switches at each end of the transmission line so that each
signal appears on the line only a fraction of time in an alternating Pattern.

 PCM-TDM - signal by interlacing two PCM signals. De- multiplexing of same.

 Each frame contained a coded version of a 'flat top' sample of an analog signal (obtained with a sample-and-
hold operation), together with a frame synchronization bit.

 Time-division multiplexing is used primarily for digital signals, but may be applied in
analog multiplexing in which two or more signals or bit streams are transferred appearing simultaneously as
sub-channels in one communication channel, but are physically taking turns on the channel.
 Time-division multiplexing (TDM) is a method of putting multiple data streams in a single signal by
separating the signal into many segments, each having a very short duration.
 The circuit that combines signals at the source (transmitting) end of a communications link is known as
a multiplexer.
 After multiplexing, these signals are transmitted over a shared medium and reassembled into their original
format after de-multiplexing.
 Time slot selection is directly proportional to overall system efficiency. Time division multiplexing (TDM)
is also known as a digital circuit switched.
 Time division multiplexing systems are more flexible than frequency division multiplexing. Time division
multiplexing circuitry is not complex. Problem of cross talk is not severe.

LINE CODES
 In telecommunication, a line code is a pattern of voltage, current, or photons used to represent digital data
transmitted down a transmission line.
 Common line encodings are unipolar, polar, bipolar, and Manchester code.

 Digital Line Coding is a special coding system chosen to allow transmission to take place in a
communications system.
 The chosen code or pattern of voltage used to represent binary digits on a transmission medium is called line
encoding.
 The types of line encoding are polar, unipolar and bipolar.
 The common types of line encoding are unipolar, polar, bipolar and Manchester encoding.
 Line codes are used commonly in computer communication networks over short distances.
 A line code is the code used for data transmission of a digital signal over a transmission line.
 This process of coding is chosen so as to avoid overlap and distortion of signal such as inter-symbol
interference.
Return to zero (RZ) –

15
Biphase (Manchester and Differential Manchester ) –

Bipolar schemes –
(i) Alternate Mark Inversion (AMI) –
(ii) Pseudoternary –

Polar schemes –In polar schemes, the voltages are on the both sides of the axis.
NRZ-L and NRZ-I –

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Unipolar scheme –
In this scheme, all the signal levels are either above or below the axis.
Non return to zero (NRZ)

Some of the more common binary line codes include:

Signal Comments 1 state 0 state

Non-return-to-zero level. This

is the standard positive logic
NRZ–L forces a high level forces a low level
signal format used in digital
circuits.

does nothing (keeps sending the

NRZ–M Non-return-to-zero mark forces a transition
previous level)
does nothing
NRZ–S Non-return-to-zero space (keeps sending the forces a transition
previous level)
goes high for half
RZ Return to zero the bit period and stays low for the entire period
returns to low
Manchester. Two consecutive
forces a negative
bits of the same type force a forces a positive transition in the
Biphase–L transition in the
transition at the beginning of a middle of the bit
middle of the bit
bit period.
Variant of Differential
Manchester. There is always a
Biphase–M forces a transition keeps level constant
transition halfway between
the conditioned transitions.
Differential Manchester used
in Token Ring. There is always keeps level
Biphase–S forces a transition
a transition halfway between constant
the conditioned transitions.
Differential Need a Clock, always a
is represented by is represented by a transition at the
Manchester transition in the middle of the
no transition. beginning of the clock period.
(Alternative) clock period
forces a positive or
The positive and negative
Bipolar negative pulse for keeps a zero level during bit period
pulses alternate.
half the bit period

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