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Dr.B.

Nagajayanthi
SENSE

1
 Communication system (analog /digital) transfers an
information bearing signal from the source to the user
destination through a communication channel.
 Analog Communication: Information bearing signal
varies continuously in both amplitude and time. This
is used to vary the amplitude/frequency/phase of the
sinusoidal carrier(high frequency signal).
 Digital Communication: Information bearing
signal(both amplitude and time) is represented as
discrete values.

2
 Computer/facsimile/etc.,
 Flexibility and compatibility to adopt a common
digital format.
 Reliability –performs as per specifications
 Wideband channels provided by coaxial cables and
geostationary satellites.
 Increased complexity due to integrated solid state
electronics.

3
 Electrical communication systems are designed to send messages
or information from a source that generates the messages to one
or more destinations.

4
 In general, a communication system can be represented by the
functional block diagram shown in following Figure.

Information
Source and
Transmitter Noise
Input
Transducer

Channel

Output Output
Receiver
Signal Transducer

5
 A transducer is usually required to convert the output of a source
into an electrical signal that is suitable for transmission.

 For example, a microphone serves as the transducer that converts


an acoustic speech signal into an electrical signal, and a video
camera converts an image into an electrical signal.

6
 At the destination, a similar transducer is required to convert the
electrical signals that are received into a form that is suitable for
the user; e.g., acoustic signals, images, etc.

7
 The heart of the communication system consists of three basic
parts, namely, the transmitter, the channel, and the receiver.

Transmitter

Channel

Receiver

8
 The transmitter converts the electrical signal into a form
that is suitable for transmission through the physical
channel or transmission medium.

 For example, in radio and TV broadcast, the Federal


Communications Commission (FCC) specifies the
frequency range for each transmitting station.

 Hence, the transmitter must translate the information


signal to be transmitted into the appropriate frequency
range.

 Thus, signals transmitted by multiple radio stations do not


interfere with one another.
9
 Usually, modulation involves the use of the
information signal to systematically vary either the
amplitude, frequency, or phase of a sinusoidal carrier.

10
 The communications channel is the physical medium
that is used to send the signal from the transmitter to
the receiver.

 In wireless transmission, the channel is usually the


atmosphere (free space).

 On the other hand, telephone channels usually employ


a variety of physical media, including copper wires and
optical fiber cables.

11
 The most common form of signal degradation comes
in the form of additive noise, which is generated at the
front end of the receiver, where signal amplification is
performed.

 This noise is often called thermal noise.

12
 In wireless transmission, additional additive
disturbances are man-made noise, and atmospheric
noise picked up by a receiving antenna.

 Automobile ignition noise is an example of man-made noise.


 Electrical lightning discharges from thunderstorms is an
example of atmospheric noise.

 Interference from other users of the channel is another


form of additive noise that often arises in both wireless
and wireline communication systems.

13
 In the design of a communication system, the
system designer works with mathematical models
that statistically characterize the signal distortion
encountered on physical channels.

14
 The function of the receiver is to recover the message
signal contained in the received signal.

15
 Since the signal demodulation is performed in the
presence of additive noise and possibly other signal
distortion, the demodulated message signal is
generally degraded to some extent by the presence
of these distortions in the received signal.

 The fidelity of the received message signal is a


function of
 Type of modulation,
 Strength of the additive noise
 Type of any non-additive interference.

16
 Source generates messages e.g., human voice, picture,
temperature etc.,
 Message signal is converted using a transducer into
electrical waveforms.
 This is the baseband signal-the band of frequencies
representing the message signal generated by the
source.
 Analog data is converted into digital using sampling,
quantization and encoding.

17
 Up to this point we have described an electrical
communication system in rather broad terms based on the
implicit assumption that the message signal is a continuous
time varying waveform.

 We refer to such continuous-time signal waveforms as analog


signals and to the corresponding information sources that
produce such signals as analog sources.

 Analog signals can be transmitted directly via carrier


modulation over the communication channel and
demodulated accordingly at the receiver.

 We call such a communication system an analog


communication system.
18
 Alternatively, an analog source output may be converted
into a digital form and the message can be transmitted via
digital modulation and demodulated as a digital signal at
the receiver.

19
 There are Some potential advantages to transmitting an
analog signal by means of digital modulation.

 The most important reason is that signal fidelity is better


controlled through digital transmission than analog
transmission.

20
21
 The process of efficiently converting the output of either
an analog or a digital source into a sequence of binary
digits is called source encoding.

 The sequence of binary digits from the source encoder,


which we call the information sequence is passed to the
channel encoder.

 The purpose of the channel encoder is to introduce, in a


controlled manner, some redundancy in the binary
information sequence which can be used at the receiver to
overcome the effects of noise and interference
encountered in the transmission of the signal through the
channel.
22
23
 In sampling ,sample values of the analog signal at
uniformly spaced discrete instants of time are
retained.
 In quantization, each sample value is rounded
approximated to the nearest value and forms finite set
of discrete levels.
 In encoding, the selected level is represented as a
codeword that has a prescribed number of code
elements.
 Sampling and quantization introduces errors.

24
 Source coding includes
 Formatting (input data)
 Sampling
 Quantization
 Symbols to bits (Encoding)
 Compression
 Decoding includes
 Decompression
 Formatting (output)
 Bits to symbols
 Symbols to sequence of numbers
 Sequence to waveform (Reconstruction)

25
 The process of efficiently converting the output of
either an analog or a digital source into a sequence of
binary digits is called source encoding.

26
 A major feature of digital data transmission is the
myriad techniques used to protect data or speech
through coding.

 Coding adds additional bits to the original payload to


provide a means of protecting the original information.

27
 The binary sequence at the output of the channel
encoder is passed to the digital modulator, which
serves as the interface to the communications channel.

28
 At the receiving end of a digital communications
system, the digital demodulator processes the
channel-corrupted transmitted waveform and
reduces each waveform to a single number that
represents an estimate of the transmitted data
symbol.

29
 Recovery

30
 As- a final step, when an analog output is desired, the
source decoder accepts the output sequence from the
channel decoder and, from knowledge of the source
encoding method used, attempts to reconstruct the
original signal from the source.

31
 To transform an analog waveform into a form that is
compatible with a digital communication, the
following steps are taken:
1. Sampling
2. Quantization and Encoding
3. Base-band transmission (PCM)

32
Strictly band limited
Band unlimited

33
• Process of converting analog signal into discrete signal.
• The signal is sampled at regular intervals such that each sample is proportional
to amplitude of signal at that instant
• Analog signal is sampled every 𝑇𝑠 𝑆𝑒𝑐𝑠, called sampling interval.
• 𝑓𝑠=1/𝑇𝑆 is called sampling rate or sampling frequency.
• 𝑓𝑠=2𝑓𝑚 is Min. sampling rate called Nyquist rate.
• Sampled spectrum (𝜔) is repeating periodically without overlapping.
• Original spectrum is centered at 𝜔=0 and having bandwidth of 𝜔𝑚.
• Spectrum can be recovered by passing through low pass filter with cut-off 𝜔𝑚.

34
 The sampling process is in time domain.
 An analog signal is converted into a corresponding
sequence of samples that are usually spaced uniformly
in time.
 Sampling rate is proportional to the bandwidth of the
message signal, so that the sequence of samples
uniquely defines the original analog signal.

35
 If Analog input-Conversion-Digital
 Sampling is a process in A/D conversion
 Sampling –Signal is discretized at periodic instants of
time.
 sampling rate properly in relation to the bandwidth of
the message signal, so that the sequence of samples
uniquely defines the original analog signal.

36
Note

According to the Nyquist theorem, the


sampling rate must be
at least 2 times the highest frequency
contained in the signal.
 Consider an arbitrary signal g(t) of finite energy, which
is specified for all time t.
 Suppose that we sample the signal g(t) instantaneously
and at a uniform rate, once every Ts seconds.
 Consequently, we obtain an infinite sequence of
samples spaced Ts seconds apart and denoted by
{g(nTs)}, where n takes on all possible integer values,
positive as well as negative.
 Ts is the sampling period, and to its reciprocal fs =
1/Ts as the sampling rate.
 For obvious reasons, this ideal form of sampling is
called instantaneous sampling. 38
39
40
 Consider an arbitrary signal x(t) of finite energy, which is specified for
all time t.
 x(t)is instantaneously sampled at a uniform rate, once every Ts
seconds.
 This results in an infinite sequence of samples spaced Ts seconds apart
and denoted by {x(nTs)}, where n takes on all possible integer values,
positive as well as negative.
 We refer to Ts as the sampling period, and to its reciprocal fs = 1/Ts as
the sampling rate.
 This ideal form of sampling is called instantaneous sampling.
 Defining x(t)

41
 Let gδ(t) denote the signal obtained by individually weighting
the elements of a periodic sequence of delta functions spaced
Ts seconds apart by the sequence of numbers {g(nTs)}---
Sampled Signal.
 The term δ(t – nTs) represents a delta function positioned at
time t = nTs. delta function is an idealized function that has unit
area.

 The multiplying factor g(nTs) with the delta function


weighted in this manner is closely approximated by a
rectangular pulse of duration Δt and amplitude g(nTs)/ Δ t; the
smaller we make Δ t the better the approximation will be.
42
 The process of uniformly sampling a continuous-time
signal of finite energy results in a periodic spectrum
with a frequency equal to the sampling rate.

 where G(f) is the Fourier transform of the original


signal g(t) and fs is the sampling rate.
 The process of uniformly sampling a continuous-time
signal of finite energy results in a periodic spectrum
with a frequency equal to the sampling rate.
43
 Taking the Fourier transform of both sides of (1) and
noting that the Fourier transform of the delta function
δ (t – nTs) is equal to exp(–j2πnfTs).
 Letting G δ (f) denote the Fourier transform of g δ (t),
we may write

 Fourier transform of product in time domain becomes


convolution in frequency domain

44
 Suppose, however, that the signal g(t) is strictly band
limited, with no frequency components higher than W
hertz.
 Fourier transform G(f) of the signal g(t) has the
property that G(f) is zero for | f |> W, is shown in this
figure.
 Choose sampling period Ts = 1/2W. Then the
corresponding spectrum of the sampled signal is as
shown . Sub Ts = 1/2W yields
46
 is the desired formula for the frequency-domain
description of sampling.
 This formula reveals that if the sample values g(n/2W)
of the signal g(t) are specified for all n, then the
Fourier transform G(f) of that signal is uniquely
determined.
 Because g(t) is related to G(f) by the inverse Fourier
transform, it follows, therefore, that g(t) is itself
uniquely determined by the sample values g(n/2W) .
 In other words, the sequence {g(n/2W)} has all the
information contained in the original signal g(t).
47
 To reconstruct the signal g(t) from the sequence of
sample values {g(n/2W)}, substitute and
interchanging the order of summation and integration

 The definite integral includes the multiplying factor


1/2W, and is evaluated in terms of the sinc function, as
shown by

 Each sample, g(n/2W), is multiplied by a delayed


version of the basis function, sinc(2Wt – n), and all the
resulting individual waveforms in the expansion are
added to reconstruct the original signal g(t).
 Ideal
 Natural
 Practical
 Sample and Hold (Flat-top)

49
Ideal Sampling ( or Impulse Sampling)

x(t)x(t)
x(t)
Ts

 Is accomplished by the multiplication of the signal x(t) by the


uniform train of impulses
 Consider the instantaneous sampling of the analog signal x(t)

 Train of impulse functions select sample values at regular intervals

50
Ideal Sampling

51
Practical Sampling
 In practice we cannot perform ideal sampling
 It is not practically possible to create a train of impulses
 Thus a non-ideal approach to sampling must be used
 We can approximate a train of impulses using a train of very
thin rectangular pulses:

52
Natural Sampling
If we multiply x(t) by a
train of rectangular
pulses xp(t), we obtain a
gated waveform that
approximates the ideal
sampled waveform,
known as natural
sampling or gating

53
Natural Sampling
 Each pulse in xp(t) has width Ts and amplitude 1/Ts
 The top of each pulse follows the variation of the signal being
sampled
 Xs (f) is the replication of X(f) periodically every fs Hz
 Xs (f) is weighted by Cn  Fourier Series Coeffiecient
 The problem with a natural sampled waveform is that the tops of the
sample pulses are not flat
 It is not compatible with a digital system since the amplitude of each
sample has infinite number of possible values
 Another technique known as flat top sampling is used to alleviate
this problem; here, the pulse is held to a constant height for the
whole sample period
 This technique is used to realize Sample-and-Hold (S/H) operation
 In S/H, input signal is continuously sampled and then the value is
held for as long as it takes to for the A/D to acquire its value 54
Flat-Top Sampling

Time Domain

Frequency Domain

55
56
 Sampling theorem indicates the signal g(t) as strictly band limited.
 If message signal is not strictly band limited, there is undersampling and this
results in aliasing.
 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.
 To combat the effects of aliasing in practice, we may use two corrective measures:
1. Prior to sampling, a low-pass anti-aliasing filter is used to attenuate those high
frequency components of the signal that are not essential to the information being
conveyed by the message signal g(t).
 2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate. This
reconstruction filter is used to recover the original signal from its sampled version.
 The reconstruction filter is low-pass with a passband extending from –W to W,
which is itself determined by the anti-aliasing filter.
 The reconstruction filter has a transition band extending (for positive frequencies)
from W to (fs – W), where fs is the sampling rate.
58
59
• Output of Sampling (natural/S&H) is known as PAM
• Pulse-Amplitude Modulation (PAM)
▫ The amplitude of regularly spaced pulses are varied in
proportion to the corresponding sample values of a
continuous message signal.
▫ Two operations involved in the generation of the PAM
signal
 Instantaneous sampling of the message signal m(t) every Ts
seconds,
 Lengthening the duration of each sample, so that it occupies
some finite value T.

60
 Let s(t) denote the sequence of flat-top pulses generated
 PAM signal is a discrete convolution sum:

 where Ts is the sampling period and m(nTs) is the sample value of m(t)
obtained at time t = nTs.
 The h(t) is a Fourier-transformal pulse.
 The delayed version of the pulse shape h(t) is expressed as

61
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63
64
 Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a
min and a max.
 The amplitude values are infinite between the
two limits.
 We need to map the infinite amplitude values
onto a finite set of known values.
 This is achieved by dividing the distance
between min and max into L zones, each of
height 
 = (max - min)/L
4.65
66
 Nonuniform quantizers have unequally spaced levels
 The spacing can be chosen to optimize the Signal-to-Noise Ratio for
a particular type of signal
 It is characterized by:
 Variable step size
 Quantizer size depend on signal size

67
 M any signals such as speech have a nonuniform distribution
Basic principle is to use more levels at regions with large probability density
function (pdf)
 Concentrate quantization levels in areas of largest pdf
Or use fine quantization (small step size) for weak signals and coarse
quantization (large step size) for strong signals

68
69
Non-uniform Quantization

Non-uniform quantization is achieved by, first passing the input signal through a
“compressor”. The output of the compressor is then passed through a uniform
quantizer.

The combined effect of the compressor and the uniform quantizer is that of a non-
uniform quantizer.

At the receiver the voice signal is restored to its original form by using an
expander.

This complete process of Compressing and Expanding the signal before and after
uniform quantization is called Companding.

70
Non-uniform Quantization (Companding)

Compressor Uniform Quantizer Expander

m(t ) mˆ (t )

71
Non-uniform Quantization (Companding)

Compressor Uniform Quantizer Expander

m(t ) mˆ (t )

The 3 stages combine to give


the characteristics of a Non-
uniform quantizer.

72
 The midpoint of each zone is assigned a value from
0 to L-1 (resulting in L values)
 Each sample falling in a zone is then approximated
to the value of the midpoint.

4.73
 Assume we have a voltage signal with amplitutes
Vmin=-20V and Vmax=+20V.
 We want to use L=8 quantization levels.
 Zone width = (20 - -20)/8 = 5
 The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to
0, 0 to +5, +5 to +10, +10 to +15, +15 to +20
 The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5,
12.5, 17.5

4.74
 Each zone is then assigned a binary code.
 The number of bits required to encode the
zones, or the number of bits per sample as it is
commonly referred to, is obtained as follows:
nb = log2 L
 Given our example, nb = 3
 The 8 zone (or level) codes are therefore: 000,
001, 010, 011, 100, 101, 110, and 111
 Assigning codes to zones:
 000 will refer to zone -20 to -15
 001 to zone -15 to -10, etc.

4.75
Figure 4.26 Quantization and encoding of a sampled signal

4.76
 When a signal is quantized, we introduce an
error - the coded signal is an approximation of
the actual amplitude value.
 The difference between actual and coded value
(midpoint) is referred to as the quantization
error.
 The more zones, the smaller  which results in
smaller errors.
 BUT, the more zones the more bits required to
encode the samples -> higher bit rate

4.77
 Signals with lower amplitude values will suffer
more from quantization error as the error
range: /2, is fixed for all signal levels.
 Non linear quantization is used to alleviate this
problem. Goal is to keep SNQR fixed for all
sample values.
 Two approaches:
 The quantization levels follow a logarithmic curve.
Smaller ’s at lower amplitudes and larger’s at
higher amplitudes.
 Companding: The sample values are compressed at
the sender into logarithmic zones, and then
expanded at the receiver. The zones are fixed in
height. 4.78
 Basically, companding introduces a nonlinearity into the signal
 This maps a nonuniform distribution into something that more
closely resembles a uniform distribution
 A standard ADC with uniform spacing between levels can be used
after the compandor (or compander)
 The companding operation is inverted at the receiver

 There are in fact two standard logarithm based companding techniques


 US standard called µ-law companding
 European standard called A-law companding

79
 Companding is a method of reducing the number of bits required in ADC
while achieving an equivalent dynamic range or SQNR
 In order to improve the resolution of weak signals within a converter, and
hence enhance the SQNR, the weak signals need to be enlarged, or the
quantization step size decreased, but only for the weak signals
 But strong signals can potentially be reduced without significantly
degrading the SQNR or alternatively increasing quantization step size
 The compression process at the transmitter must be matched with an
equivalent expansion process at the receiver

80
 The signal below shows the effect of compression, where the
amplitude of one of the signals is compressed
 After compression, input to the quantizer will have a more uniform
distribution after sampling

 At the receiver, the signal is


expanded by an inverse operation
 The process of CO M pressing and
exPANDING the signal is called
companding
 Companding is a technique used
to reduce the number of bits
required in ADC or DAC while
achieving comparable SQNR

81
 Logarithmic expression Y = log X is the most commonly
used compander
 This reduces the dynamic range of Y

82
log e 1   (| x | / xmax 
y  ymax sgn( x)
log e (1   )

where
 x and y represent the input and output voltages
  is a constant number determined by experiment
 In the U.S., telephone lines uses companding with  = 255
 Samples 4 kHz speech waveform at 8,000 sample/sec
 Encodes each sample with 8 bits, L = 256 quantizer levels
 Hence data rate R = 64 kbit/sec
  = 0 corresponds to uniform quantization

83
 | x|
 A
xmax | x| 1
 ymax sgn( x), 0 
 (1  A) xmax A
y ( x)  
   | x| 
 1  log e  A 
  xmax  1 | x|
 ymax sgn( x),  1
 (1  log e A) A xmax
where
 x and y represent the input and output voltages
 A = 87.6
 A is a constant number determined by experiment

84
• Sampling and Quantization Effects
▫ Quantization (Granularity) Noise: Results when
quantization levels are not finely spaced apart enough to
accurately approximate input signal resulting in
truncation or rounding error.
▫ Quantizer Saturation or Overload Noise: Results
when input signal is larger in magnitude than highest
quantization level resulting in clipping of the signal.
▫ Timing Jitter: Error caused by a shift in the sampler
position. Can be isolated with stable clock reference.

85
WAVEFORM CODING TECHNIQUES
DR.B.NAGAJAYANTHI SENSE

86
87
Pulse Code Modulation (PCM)

 Pulse Code Modulation refers to a digital baseband


signal that is generated directly from the quantizer and
encoder output
 Sometimes the term PCM is used interchangeably with
quantization

88
89
90
 PCM (Pulse-Code Modulation)
 A message signal is represented by a sequence of coded pulses, which is
accomplished by representing the signal in discrete form in both time and
amplitude
 The basic operation
 Transmitter : sampling, quantization, encoding
 Receiver : regeneration, decoding, reconstruction

 Operation in the Transmitter


1. Sampling
1. The incoming message signal is sampled with a train of rectangular pulses
2. The reduction of the continuously varying message signal to a limited number
of discrete values per second
2. Nonuniform Quantization
1. The step size increases as the separation from the origin of the input-output
amplitude characteristic is increased, the large end-step of the quantizer can
take care of possible excursions of the voice signal into the large amplitude
ranges that occur relatively infrequently. 91
3. Encoding
1. To translate the discrete set of sample vales to a more
appropriate form of signal
2. A binary code
 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.
 The binary code is easy to generate and regenerate

92
93
 Regeneration Along the Transmission Path
 The ability to control the effects of distortion and noise produced by
transmitting a PCM signal over a channel
 Equalizer
 Shapes the received pulses so as to compensate for the effects of amplitude and
phase distortions produced by the transmission
 Timing circuitry
 Provides a periodic pulse train, derived from the received pulses
 Renewed sampling of the equalized pulses
 Decision-making device
 The sample so extracted is compared o a predetermined threshold
 ideally, except for delay, the regenerated signal is exactly the same as the
information-bearing signal
1. The unavoidable presence of channel noise and interference causes the repeater
to make wrong decisions occasionally, thereby introducing bit errors into the
regenerated signal
2. If the spacing between received pulses deviates from its assigned value, a jitter is
introduced into the regenerated pulse position, thereby causing distortion.

94
95
• Operations in the Receivers
1. Decoding and expanding
1. Decoding : regenerating a pulse whose amplitude is the linear
sum of all the pulses in the code word
2. Expander : a subsystem in the receiver with a characteristic
complementary to the compressor
1. The combination of a compressor and an expander is a compander

2. Reconstruction
1. Recover the message signal : passing the expander output
through a low-pass reconstruction filter

96
 The bit rate of a PCM signal can be calculated form the
number of bits per sample x the sampling rate
Bit rate = nb x fs
 The bandwidth required to transmit this signal depends
on the type of line encoding used. Refer to previous
section for discussion and formulas.
 A digitized signal will always need more bandwidth
than the original analog signal. Price we pay for
robustness and other features of digital transmission.

4.97
Example

We want to digitize the human voice. What is the


bit rate, assuming 8 bits per sample?

Solution
The human voice normally contains frequencies from 0
to 4000 Hz. So the sampling rate and bit rate are
calculated as follows:

4.98
 To recover an analog signal from a digitized
signal we follow the following steps:
 We use a hold circuit that holds the amplitude value
of a pulse till the next pulse arrives.
 We pass this signal through a low pass filter with a
cutoff frequency that is equal to the highest
frequency in the pre-sampled signal.
 The higher the value of L, the less distorted a
signal is recovered.

4.99
Components of a PCM decoder

4.100
amplitude
x(t)
111 3.1867

110 2.2762 Quant. levels


101 1.3657

100 0.4552

011 -0.4552 boundaries

010 -1.3657

001 -2.2762 x(nTs): sampled values


xq(nTs): quantized values
000 -3.1867
Ts: sampling time
PCM t
codeword 110 110 111 110 100 010 011 100 100 011 PCM sequence
101
Example

We have a low-pass analog signal of 4 kHz. If we


send the analog signal, we need a channel with a
minimum bandwidth of 4 kHz. If we digitize the
signal and send 8 bits per sample, we need a
channel with a minimum bandwidth of 8 × 4 kHz =
32 kHz.

4.102
 In uniform quantization the variance of quantization
noise is independent of the variance of the input
signal.
 PCM accommodates transmission of speech signals
with varying power levels.
 A robust quantizer has equal SNR for varying input
power levels .
 This requires a non-uniform quantizer. Step size
increases as the separation from the origin increases.
 Compressor +Uniform Quantization. To restore
expander is used. Compressor and Expander forms
companding. 103
 Signal is amplified at low signal levels and attenuated
at high signal levels. Dotted lines show uniform
quantization.

104
 Except some samples the step size will be same. Smaller amplitudes are
more. Step size is small. Larger amplitude signals has larger step size.
 So reduce the quantization error for smaller amplitudes.
 Increase the number of levels which is increase the bits of
representation.

105
106
1

 Telephones in the U.S., Canada


and Japan use -law companding:
Output |x(t)|

ln(1   | x(t )|)


| y (t ) |
ln(1   )
 Where  = 255 and |x(t)| < 1
 Input is normalized by the
maximum value.
 To avoid negative values
absolute is used.
0 1
Input |x(t)|
Eeng
360
107
108
 If μ=0 compression factor is zero then output is linear.
 If it is increasedmore compression so concentrate for
increased the number of step sizes.

109
 More sharp A is used to represent the amount of
compression. If A is 1 there is no compression.

110
111
 As A increases linearity reduces.

112
 PCM-Analog to digital
 As fs increases the samples are close and are almost
same.
 Redudant information is transmitted so BW is wasted
as correlated samples are transmitted.
 If input level is more L is more then number of bits is
less so more compression is achieved.
 Find difference between samples , are quantized ,
encode and transmit.
 L is less number of bits is less.
113
114
 Staircase approximated signal is passed through LPF.

115
116
117
 This scheme sends only the difference between
pulses, if the pulse at time tn+1 is higher in
amplitude value than the pulse at time tn, then
a single bit, say a “1”, is used to indicate the
positive value.
 If the pulse is lower in value, resulting in a
negative value, a “0” is used.
 This scheme works well for small changes in
signal values between samples.
 If changes in amplitude are large, this will result
in large errors.
4.118
The process of delta modulation

4.119
Delta modulation components

4.120
Delta demodulation components

4.121
 Instead of using one bit to indicate positive and
negative differences, we can use more bits ->
quantization of the difference.
 Each bit code is used to represent the value of
the difference.
 The more bits the more levels -> the higher the
accuracy.

4.122
123
 PDM (Pulse-duration modulation)
 Pulse-width or Pulse-length modulation.
 The samples of the message signal are used to vary the
duration of the individual pulses.
 PDM is wasteful of power

 PPM (Pulse-position modulation)


 The position of a pulse relative to its un-modulated time of
occurrence is varied in accordance with the message signal.

124
125
126
• The input to the line encoder is the output
of the A/D converter or a sequence of
values an that is a function of the data bit
• The output of the line encoder is a
waveform: 
s (t )  a
n 
n f (t  nTb )

where f(t) is the pulse shape and Tb is the bit period (Tb=Ts/n for n bit
quantizer)
 This means that each line code is described by a symbol mapping
function an and pulse shape f(t)
 Details of this operation are set by the type of line code that is being used

127
Goals of Line Coding
A line code is designed to meet one or more of the following goals:
 Self-synchronization
 The ability to recover timing from the signal itself
 That is, self-clocking (self-synchronization) - ease of clock lock or
signal recovery for symbol synchronization
 Long series of ones and zeros could cause a problem
 Low probability of bit error
 Receiver needs to be able to distinguish the waveform associated
with a mark from the waveform associated with a space
 BER performance
 relative immunity to noise
 Error detection capability
 enhances low probability of error

128
 Spectrum Suitable for the channel
 Spectrum matching of the channel
 e.g. presence or absence of DC level
 In some cases DC components should be avoided
 The transmission bandwidth should be minimized
 Power Spectral Density
 Particularly its value at zero
 PSD of code should be negligible at the frequency near zero
 Transmission Bandwidth
 Should be as small as possible
 Transparency
 The property that any arbitrary symbol or bit pattern can be
transmitted and received, i.e., all possible data sequence should
be faithfully reproducible

129
 Categories of Line Codes
 Polar – Send positive pulse or negative of pulse
 Uni-polar - Send pulse or a 0
 Bipolar (a.k.a. alternate mark inversion, pseudoternary)
 Represent 1 by alternating signed pulses
 Generalized Pulse Shapes
 NRZ -Pulse lasts entire bit period
 Polar NRZ
 Bipolar NRZ
 Unipolar NRZ
 RZ - Return to Zero - pulse lasts just half of bit period
 Polar RZ
 Bipolar RZ
 Unipolar RZ
 Manchester Line Code
 Send a 2-  pulse for either 1 (high low) or 0 (low high)
 Includes rising and falling edge in each pulse
130
 No DC component
 When the category and the generalized shapes are combined, we have the
following:
 Polar NRZ:
 Wireless, radio, and satellite applications primarily use Polar
NRZ because bandwidth is precious
 Unipolar NRZ
 Turn the pulse ON for a ‘1’, leave the pulse OFF for a ‘0’
 Useful for noncoherent communication where receiver can’t
decide the sign of a pulse
 fiber optic communication often use this signaling format
 Unipolar RZ
 RZ signaling has both a rising and falling edge of the pulse
 This can be useful for timing and synchronization purposes

131
 Bipolar RZ
 A unipolar line code, except now we alternate
between positive and negative pulses to send a ‘1’
 Alternating like this eliminates the DC component
 This is desirable for many channels that cannot
transmit the DC components

132
133
134
135
 Self-synchronization
 Manchester codes have built in timing information because they
always have a zero crossing in the center of the pulse
 Polar RZ codes tend to be good because the signal level always
goes to zero for the second half of the pulse
 NRZ signals are not good for self-synchronization
 Error probability
 Polar codes perform better (are more energy efficient) than Uni-
polar or Bipolar codes
 Channel characteristics
 We need to find the power spectral density (PSD) of the line
codes to compare the line codes in terms of the channel
characteristics

136
 Different pulse shapes are used
 to control the spectrum of the transmitted signal (no DC value,
bandwidth, etc.)
 guarantee transitions every symbol interval to assist in symbol timing
recovery
1. Power Spectral Density of Line Codes (see Fig. 2.23, Page 90)
 After line coding, the pulses may be filtered or shaped to further improve
there properties such as
 Spectral efficiency
 Immunity to Intersymbol Interference
 Distinction between Line Coding and Pulse Shaping is not easy
2. DC Component and Bandwidth
 DC Components
 Unipolar NRZ, polar NRZ, and unipolar RZ all have DC components
 Bipolar RZ and Manchester NRZ do not have DC components

137
138
(a) Original binary data. (b) Differentially encoded data, assuming
reference bit 1. (c) Waveform of differentially encoded data using
unipolar NRZ signaling.

139
• Encoding
▫ encoded(k) = encoded(k – 1) XOR original(k)
▫ where k starts from 0
▫ Encoded(-1) is called the reference bit which can be
either 1 or 0
• Decoding
▫ original(k) = encoded (k – 1) XOR encoded(k)
▫ where k starts from 0
▫ Reference bit remains same for both encoding and
decoding process

140
• Channel Effects
▫ Channel Noise (AWGN, White Noise, Thermal etc)
▫ Intersymbol Interference (ISI)

• Sampling and Quantization Effects


▫ Quantization (Granularity) Noise
▫ Quantizer Saturation or Overload Noise
▫ Timing Jitter

141

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