Module – 3: Pulse Modulation Systems

Pulse Modulation Techniques

Pulse Modulation techniques are two types: Pulse Analog and Pulse Digital modulation
techniques

Pulse Analog Modulation Techniques:

There are three main types using which we represent the information carried by a sequence of
samples (three types of pulse modulations). Notice that the term “modulation” here is not
used in the sense of modulation that we used in the previous chapters, which the frequency of
a signal is shifted to a higher frequency for transmission. The term modulation here is used to
specify the process in which the information signal modifies some parameter of a sequence of
pulses. This parameter is used to transmit the desired information.
Pulse modulation consists essentially of sampling analog information signals and then
converting those samples into discrete pulses and transporting the pulses from a source to a
destination over a physical transmission medium. The three predominant methods of pulse
modulation: 1) Pulse amplitude modulation (PAM), 2) Pulse width modulation (PWM) and
3) Pulse position modulation (PPM)

(a)

(b)

(c)

(d)
Figure 7.5: (a) Message and Carrier (b) PAM (c)PWM (d)PPM

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 Module – 3: Pulse Modulation Systems

Pulse Amplitude Modulation (PAM)

 With PAM, the amplitude of a constant width, constant-position pulse is varied
according to the amplitude of the sample of the analog signal.
 The amplitude of a pulse coincides with the amplitude of the analog signal.
 PAM waveform resemble the original analog signal more than the PWM or PPM.

PAM Signal

Generation of PAM

 Pulse amplitude modulation is the basic form of pulse modulation in which the signal
is sampled at regular and each sample is made proportional to the amplitude of the
modulating signal at the sampling instant.
 The Fig 1 shows the generation of PAM signal from the sampler which has two inputs
i.e. modulating signal and sampling signal or carrier pulse.
 Thus the amplitude of the signal is proportional to the modulating signal through
which information is carried. This is Pulse amplitude modulation signal.
 Fig 2 shows the spectrum of pulse amplitude modulated signal along with the
message signal and the sampling signal which is the carrier train of pulses with the
help of the waveform plotted in time domain.
 Pulse Modulation may be used to transmitting analog information, such as continuous
speech signal or data.
 Instantaneous sampling of the message signal every Ts seconds, where the sampling
rate fs = 1/Ts is chosen in accordance with the sampling theorem.
 Lengthening the duration of the each sample so obtained to some constant value T.

Fig1. Generation of PAM signal

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Pulse Width Modulation (PWM) or (PDM)

 PWM is sometimes called pulse duration modulation (PDM), as the width (active
portion of the duty cycle) of a constant amplitude pulse is varied proportional to the
amplitude of the analog signal at the time the signal is sampled.
 The maximum analog signal amplitude produces the widest pulse, and the minimum
analog signal amplitude produces the narrowest pulse with same amplitude.

Fig. 1: PWM signal

Generation of PWM
 PWM signal can be generated by using a comparator, where modulating signal and
sawtooth signal form the input of the comparator.
 One input of the comparator is fed by the input message or modulating signal and the
other input by a sawtooth signal which operates at carrier frequency.
 The rising edges of the PWM signal coincides with the falling edge of the sawtooth
signal.
 When the sawtooth signal is at the minimum value which is less than the minimum of
the input signal, then the positive input of the comparator is at higher potential which
gives the comparator output as positive.
 When the sawtooth signal rises and is at the maximum value, the negative input of the
comparator is at higher potential, which will produce the comparator output to be –ve.
 Thus the input signal magnitude determines the comparator output and its potential,
which then decides the width of the pulse generated at the output.
 In other words we can say that the width of the pulse generated signal is directly
proportional to the amplitude of the modulating signal.

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Pulse Position Modulation

 With PPM, the position of a constant-width pulse within a prescribed time slot is
varied according to the amplitude of the sample of the analog signal.
 The higher the amplitude of the sample, the farther to the right the pulse is positioned
within the prescribed time slot.
 The highest amplitude sample produces a pulse to the far right, and the lowest
amplitude sample produces a pulse to the far left.

PPM Signal
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 Module – 3: Pulse Modulation Systems

Generation of PPM

Fig 7. PPM generation from PWM

 PPM signal can be generated with the help of PWM as shown in Fig7 below.
 The PWM signal generated above is sent to an inverter which reverses the polarity of
the pulses.
 This is then followed by a differentiator which generates +ve spikes for PWM signal
going from High to Low and -ve spikes for Low to High transistion. The spikes
generated are shown in the fourth waveform of Fig8.
 These spikes are then fed to the positive edge triggered pulse generator which
generates fixed width pulses when a +ve spike appears, coinciding with the falling
edge of the PWM signal.
 Thus PPM signal is generated at the output which is shown in the fifth waveform of
Fig8.where pulse position carry the message information.

Fig6. PWM and PPM signal generation

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 Module – 3: Pulse Modulation Systems

Pulse Digital Modulation Techniques (Waveform Coding Techniques)

A message signal can be originated from a digital or analog source. If the message signal is
derived from a digital source (e.g., digital computer), then from inception it is in the right
form for processing in digital communication system. If, however the message signal happens
to be analog in nature, as in a speech or video signal, then it has to be converted into digital
form before it can be transmitted by digital means. The block called formatter does this job.
The sampling process is the first process performed in analog to digital convertion. Two other
processes, Quantizing an encoding, are also involved in this conversion. Encoding schemes
like Pulse Code Modulation (PCM), Differential Pulse Code modulation (DPCM), Delta
Modulation (DM), Adaptive Delta Modulation (ADM) and so on are employed for this
purpose. This operation of encoding gives rise to digital signals. Though, almost all of these
coders carry the word “modulation” in their names, there is nothing in their bahaviour
resembling modulation. The are simply encoding techniques for analog sources. One should
conceptually be clear that even after passing through these so called “modulator” blocks the
digitised message sigal remains a base band signal, because it does not undergo any
frequency translation in any part of those blocks.

2.1 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. The essential operations in the transmitter of a PCM are sampling, Quantizing and
encoding as shown in Figure 2.1(a). The low-pass filter prior to sampling is included to
prevent aliasing of the message signal. The quantizing and encoding operations are usually
performed in the same circuit, which is called analog-to-digital converter. The basic
operations in the receiver are regeneration of impaired signals, decoding and reconstruction
of the train of quantized samples as shown in Figure 2.1(c). Regeneration also occurs at
intermediate points along the transmission path as necessary, as indicated in Figure 2.1(b).
The various operations of PCM are given as follows.

Figure 2.1: Basic elements of a PCM system (a) transmitter (b) transmission path (c) receiver

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Sampling
The sampling process is usually described in the time domain. As such, it is an operation that
is basic to digital signal processing and digital communications. Through use of the sampling
processes, an analog signal is converted into a corresponding sequence of samples that are
usually spaced uniformly in the time. Clearly, for such a procedure to have practical utility, it
is necessary that we choose the sampling rate properly, so that the sequence of samples
uniquely defined the original analog signal. This is the essence of the sampling theorem,
which is derived in what follows.
Consider an arbitrary signal x(t)of finite energy, which is specified for all time. A
segment of the signal x(t) is shown in Figure 7.1(a). Suppose that we sample the signal x(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 {x(nTs)}, where n takes
on all possible integer values. 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.

Figure 7.1: (a) x(t) (b) s(t) (c)xδ(t)

Let xδ(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
{x(nTs)}, as shown (see Figure 2.2(b))


x (t )   x  nT    t  nT 
n 
s s (2.1)

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 Module – 3: Pulse Modulation Systems

We refer to xδ(t) as the ideal sampled signal. The term δ(t – nTs) represents a delta function
positioned at t = nTs. From the definition of the delta function, we recall that such an
idealized function has unit area. We may therefore view the multiplying factor x(nTs) in Eq.
(2.1) as a “mass” assigned to the delta function δ(t – nTs). A delta function weighted in this
manner is closely approximated by a rectangular pulse of duration Δt and amplitude x(nTs)/
Δt; the smaller we make Δt the better will be the approximation. Using Fourier transform
pair, we may write


x (t )  f s  X  f  mT 
m
s (2.2)

Where G(f) is the Fourier transform of the original signal x(t), and fs is the sampling rate. Eq.
(2.2) states that the process of uniformly sampling a continuous–time signal of finite energy
results in a periodic spectrum with a period equal to the sampling rate.
Another useful expression for the Fourier transform of the ideal sampled signal xδ(t)
may be obtained by taking the Fourier transform of both sides of Eq. (2.1) and noting that the
Fourier transform of the delta function δ(t – nTs) is equal to exp(–j2πnfTs). Let Xδ(f) denote
the Fourier transform of xδ(t). We may therefore write


X ( f )   x  nT  exp   j2 nfT 
n 
s s (2.3)

This relation is called the discrete – time Fourier transform. It may be viewed as a complex
Fourier series representation of the periodic frequency function Xδ(f), with the sequence of
samples {x(nTs)} defining the coefficients of the expansion.
The relations, as derived here, apply to any continuous–time signal x(t) of finite
energy and infinite duration. Suppose, however, that the signal x(t) is strictly band limited,
with no frequency components higher than W hertz. That is, the Fourier transform X(f ) of the
signal x(t) has the property that X(f ) is zero | f | ≥ W, as illustrated in Figure 2.3(a); the shape
of the spectrum shown in this figure is intended for the purpose of illustration only. Suppose
also that we choose the sampling period Ts = 1/2W. Then the corresponding spectrum Xδ(f ) of
the sampled signal xδ(t) is shown in Figure 7.2. Putting Ts = 1/2W in Eq. (2.2) yields


 n   j nf 
X ( f )   x  2W  exp  
n  W 
 (2.4)

From Eq. (2.2), we readily see that the Fourier transform of xδ(t) may also be expressed as


X ( f )  fs X ( f )  fs  X ( f  mf )
m 
s (2.5)

Hence, under the following two conditions:

1. X(f ) = 0 for | f | ≥ W
2. fs = 2W

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we find from Eq. (2.5) that

1
X( f )  X  ( f ),  W  f  W (2.6)
2W

Substituting Eq. (2.4) into (2.6), we may also write

1 
 n   j nf 
X( f ) 
2W
 x  2W  exp  
n  W
,

W  f  W (2.7)

Figure 7.2: Spectrum of xδ(t) when fs = 2W

Therefore, if the sample values x(n/2W) of a signal x(t) are specified for all n, then the Fourier
transform X(f) of the signal is uniquely determined by using the discrete–time Fourier
transform of Eq. (2.7). Because x(t) is related X(f) by the inverse Fourier transform, it follows
that the signal g(t) is itself uniquely determined by the sample values x(n/2W) for – ∞ < n <
∞. In other words, the sequence {x(n/2W)} has all the information contained in x(t).
Consider next the problem of reconstructing the signal x(t) from the sequence of
sample values {x(n/2W)}. Substituting Eq. (2.7) in the formula for the inverse Fourier
transform defining g(t) in terms of X(f), we get


x (t )   X ( f ) exp( j 2 ft )df


W 1   n   j nf 

W 2W
 x
n   2W
 exp  
  W
 exp( j 2 ft )df


Interchanging the order of summation and integration:


 n  1 W   n 
x (t )   x  2W  2W 
n 
W
exp  j 2

f t    df
 2W  
(2.8)

The integral term in Eq. (2.8) is readily evaluated, yielding the final result


 n  sin  2 Wt  n 
x (t )   x  2W   2Wt  n 
n 

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 Module – 3: Pulse Modulation Systems


 n 
  x  2W  sinc  2Wt  n  ,
n 
t  (2.9)

Eq. (2.9) provides an interpolation formula for reconstructing the original signal x(t) from the
sequence of sample values {x(n/2W)}, with the sinc function sinc(2Wt) playing the role of an
interpolation function. Each sample is multiplied by a delayed version of interpolation
function, and all the resulting waveforms are added to obtain x(t).
We may state the sampling theorem for strictly band–limited signals of finite energy
in two equivalent parts, which apply to the transmitter and receiver of pulse modulation
system, respectively:
1. A band–limited signal of finite energy, which has no frequency components higher
than W Hertz, is completely described by specifying the values of the signal instants
of time separated by 1/2W seconds.
2. A band–limited signal of finite energy, which has no frequency components higher
than W Hertz, may be completely recovered from a knowledge of its samples taken at
the rate of 2W samples per second.
The sampling rate of 2W samples per second, for a signal bandwidth of W hertz, is called the
Nyquist rate and its reciprocal 1/2W (measured in seconds) is called the Nyquist interval.

Figure 7.3: Spectrum of xδ(t) when fs < 2W

The derivation of the sampling theorem, as described herein, is based on the assumption that
the signal x(t) is strictly band limited. In practice, however, an information bearing signal is
not strictly band limited, with the result that some degree of under-sampling is encountered.
Consequently, some aliasing is produced by the sampling process. Aliasing refers to the
phenomenon of a high frequency component in the spectrum of the signal seemingly tacking
on the identity of a lower frequency in the spectrum of its sampled version, as illustrated in
Figure 7.3. The aliased spectrum, shown by the solid curve in Figure 7.3, pertains to an
“under-sampled” version of the message signal represented by the spectrum of Figure 7.3.
To combat the effect of aliasing in practice, we may use two corrective measures, as
described here:

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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 signal
2. The filtered signal is sampled at a rate slightly higher that the Nyquist rate.
The use of a sampling rate higher than the Nyquist rate also has the beneficial effect of easing
the design of the reconstruction filter used to recover the original signal from its sampled
version. Consider the example of a message signal that has been anti-alias (low-pass) filtered,
resulting in the spectrum shown in Figure 7.4. The corresponding spectrum of the
instantaneously sampled version of the signal is shown in Figure 7.4, assuming a sampling
rate higher than the Nyquist rate.
 The reconstruction filter is low-pass with a pass-band extending from –W to W, which
is itself determined by the anti-aliasing filter.
 The filter has a transition band extending (for positive frequencies) from W to fs – W,
where fs is the sampling rate.

Figure 7.4: (a) Spectrum of x(t) (b) Spectrum of xδ(t)when fs = 2W

The fact that the reconstruction filter has a well-defined transition band means that it is
physically realizable.

Quantization:
A continuous signal, such as voice, has a continuous range of amplitudes and therefore its
samples have a continuous amplitude range. In other word, within the finite amplitude range
of signals, we find an infinite number of amplitude levels. It is not necessary in fact to
transmit the exact amplitudes of the samples. Any human sense (the eye or the ear), as
ultimate receiver, can detect only finite intensity differences. This means that the original

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continuous signal may be approximated by a signal constructed of discrete amplitudes

selected on minimum error basis from an available set. The existence of a finite number of
discrete amplitude levels is a basic condition of pulse-code modulation. Clearly, if we assign
the discrete amplitude levels with sufficiently close spacing, we may make the approximated
signal practically indistinguishable from the original continuous signal.
Amplitude quantization is defined as the process of transforming the sample
amplitudes 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. We assume that the quantization process is
memory-less and instantaneous, which means that the transformation at time t = nTs is not
affected by earlier or later samples of the message signal.

Figure 1.10: Description of a memory-less quantizer

When dealing with a memory-less quantizer, we may simplify the notation by dropping the
time index. We may thus use the symbol m in place of m(nTs), as indicated in the block
diagram of a quantizer shown in Figure 1.10 (a). Then, as shown in Figure 1.10 (b), the signal
amplitude m is specified by the index k if it lies inside the partition cell

I k : mk  m  mk 1, k  1,2,..., L (1.6)

where L is the total number of amplitude levels used in the quantizer. The discrete amplitudes
mk, 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 Ik. These discrete amplitudes vk, k = 1, 2, …, L, are called
representation levels or reconstruction levels, and the spacing between two adjacent
representation levels is called a quantum or step size. Thus, the quantizer output v equals vk if
the input signal m belongs to the interval Ik. The mapping (see Figure 1.10 (a)) v = g(m) is the
quantizer characteristic, which is a staircase function by definition.
Quantizers can be of uniform or non-uniform type. In a uniform quantizer, the
representation levels are uniformly spaced; otherwise, the quantizer is non-uniform. The
quantizer characteristic can also be of midtread or midrise type. Figure 1.11 (a) shows the
input-output characteristic of a uniform quantizer of the midtread type, which is so called
because the origin lies in the middle of a thread of a staircase like graph. Figure 1.11 (b)
shows the corresponding input-output characteristic of the staircase like graph. Note that both
the midtread and midrise types of uniform quantizers illustrated in Figure 1.11 are symmetric
about the origin.

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 Module – 3: Pulse Modulation Systems

(a) (b)
Figure 1.11: Two types of quantization: (a) mid-thread and (b) mid-rise

Encoding:
In combining the process of sampling and quantization, the specification of a continuous
message (baseband) signal becomes limited to a discrete set of values, but not in the form
best suiyed to transmission over a telephone line or rado path. To exploit the adavantages of
samling and quantization for the porpose of making the transmitted signal more robust to
noise, interference and othe channel impairements, we require the use of an encoding process
to translate the discrete set of sample values to a more appropriate form of signal. During
encoding, each of the descrete sample is represented by a codewore, where each element in
the codeword is known as code element or binary symbol (bit). The stream of such
codewords is called as a code.

Table2.1: Binary number system for n = 4 bits/sample

Ordinary Number of Level Number Expressed as Binary

Representation level sum of powers of 2 Number
0 0000
3 2 1 0
1 2 +2 +2 +2 0001
2 23 + 22 + 21 + 20 0010
3 23 + 22 + 21 + 20 0011
4 23 + 22 + 21 + 20 0100
5 23 + 22 + 21 + 20 0101
6 23 + 22 + 21 + 20 0110
7 23 + 22 + 21 + 20 0111
8 23 + 22 + 21 + 20 1000
9 23 + 22 + 21 + 20 1001
10 23 + 22 + 21 + 20 1010
11 23 + 22 + 21 + 20 1011
12 23 + 22 + 21 + 20 1100
13 23 + 22 + 21 + 20 1101
14 23 + 22 + 21 + 20 1110
15 23 + 22 + 21 + 20 1111

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In a binary code, each binary symbol may be either of two distinct values or kinds,
such as the presence or absence of a pulse. The two binary symbols of a binary code are
customarily denoted as 0 and 1. The binary represention is advantageous because a binary
symbol withstands a relatively high level of noise and is easy to regenerate. Suppose that, in a
binary code, each code word consists of n bits, where n denotes number of bits per sample.
Then using such a code, we may represent a total of 2n distinct number. For example, a
sample quantized into one of 256 levels may be represented by an 8-bit code word. There are
several ways of establishing a one-to-one correspondence between representation levels and
code words. A convenient method is to express the ordinal number of the representation level
as a binary number. In a binary number system, each digit has a place-value that is a power of
2, as given in Table 2.1 for the case of four bits per sample.
The use of an appropriate wave for for baseband representation of digital data is basic
to its transmission from source to a destination. Figure 3.1 show four different formats for the
represetation of binary data sequence 0110100011. In the unipolar format (also kown as on –
off signaling), symbol 1 is represeted by transmitting a pulse, whereas symbol 0 is
represented by switing of the pulse. When the pulse occupies the full duration of a symbol,
the unipolar format is said to be of the nonreturn to zero (NRZ) type. When it occupies a
fraction (usually one half) of the symbol duratio, it is said to be of return to zero (RZ) type.
The NRZ version of the unipolar format (based on a rectangular pulse) is shown in Figure 3.1
(a). The unipolar format offers simplicity of implementation. However, it contains a dc
component that is often found to be objectionable. In the polar format, a positive pulse is
transmitted for symbol, 1, and a negative pulse for symbol 0. It can be of the NRZ or RZ
type. The NRZ version of the polar format (using rectangular pulse) is depicted in Figure 3.1
(b). Unlike the unipolar waveform, a polar waveform has no dc component, provided that the
0s and 1s in the input data occur in equal proportion. In the bipolar format (also known as
pseudoternary signaling), positive and negative pulses are used alternately for the
transmission of 1s (with the alternation tasking place at every occurrence of a 1), and no
pulses for the transmission of 0s. It can be of the NRZ or RZ type. The NRZ version of the
bipolar format (based on rectangular pulses) is depicted in Figure 3.1 (c). Note that in this
representation, there are three levels: +1, 0 and – 1 (hence, the alternative name
“psedoternanry”). An attractive feature of the bipolar format is the absence of a dc
component, even though the input binary data may contain long string of 0s and 1s. This
property does not hold for the unipolar and polar formats. Also, the pulse alternation property
of the bipolar format provides a capability for in service performance monitoring in the sense
that any isolated error, whether it causes the deletion or creation of a pulse, will violate this
property. Moreover, the bipolar format eliminates ambiguity that may arise because of
polarity inversion during the course of transmission (this problem is a characteristic of
switched telephone networks). It is for these reasons that the bipolar format is adopted for use
in the T1 carrier system for digital telephony. The absence of dc transmission permits
repeaters, o the T1 carrier, to be transformer coupled. In the Manchester format (also known
as biphase baseband signaling), symbol 1 is represented by transmitting a positive pulse for
one half of the symbol duration, followed by a negative pulse for the remaining half of the
symbol duration; for symbol 0, these two pulses are transmitted in reverse order. The
Manchester format is depicted in Figure 3.1 (d). Clearly, it has no dc component.

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The NRZ versions of the unipolar, polar and bipolar formats make efficient use of
bandwidth. However, they do not offer a synchronization capability. On the other hand, the
Manchester format has a built-in synchronization capability, because there is a predictable
transition during each bit interval. But this capability is attained at the expense of a
bandwidth requirement twice that the NRZ unipolar, polar and bipolar formats.

Figure 3.1: Binary data formats (a) NRZ unipolar (b) NRZ polar format (c) NRZ bipolar format (d)
Manchester format

2.1.1 Regeneration
The most important feature of PCM system lies in the ability to control the effect of distortion
and noise produced by transmitting a PCM signal through a channel. This capability is
accomplished 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 Figure 2.3,
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 amplitudes and phase distortions produced by non-ideal transmission characteristics of the
channel. The timing circuitry provides a periodic pulse train, derived from the received
pulses, for sampling the equalized pulses at the instants of time where the signal-to-noise is a
maximum. Each sample so extracted is compared to a predetermined threshold in the
decision-making device. In each bit interval, a decision is then made whether the received
symbol is a 1 or a 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 completely removed, provided that
the disturbance is not too large to cause an error in the decision-making process. Ideally,
except for delay, the regenerated signal is exactly the same as the signal originally

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transmitted. 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 decision 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.

Figure 2.3: Block diagram of regenerative repeater

Decoding
The first operation in the receiver is to regenerate (i.e., reshape and clean up) the received
pulses one last time. These clean pulses are then regrouped into code words and decoded (i.e.,
mapped back) into a quantized signal. The decoding process involves generating a pulse the
amplitude of which is the linear sum of all the pulses in the code word, with each pulse being
weighted by its place value (20, 21, 22, …, 2n–1) in the code, where n is number of bits per
sample.

Reconstruction filters:
The final operation in the receiver is to recover the message signal by passing the decoder
output through a low-pass reconstruction filter whose cut-off frequency is equal to the
message bandwidth W. Assuming that the transmission path is error free, the recovered signal
includes no noise with the exception of the distortion introduced by the quantization process.

2.1.2 Bandwidth of PCM

Let us consider a PCM signal is transmitted using non return to zero unipolar wave and each
sample of duration Ts is represented by a code word of size n bits. Hence the bit duration is:

Tb = Ts/n (2.1)

or, correspondingly, the bit rate is

 bits samples  bits

rb  nrs   (2.2)
 sample sec  sec

where rs is the sampling rate. According to Nyquist theory, this rate should be equal to two
times the bandwidth of PCM. Thus, the bandwidth of PCM is: rb / 2.

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Signal to quantization noise ratio of midrise quantizer:

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 1.12 illustrates a
typical variation of the quantization as a function of time, assuming the use of a uniform
quantizer of the midtread type.

Figure 1.12: Illustration of the quantization process

Let the quantizer input m be the sample value of a zero mean random variable M. (If
the input has a nonzero mean, we can always remove it by subtracting the mean from the
input and then adding it back after quantization). A quantizer g(.) maps the input random
variable M of continuous amplitude into a discrete random variable V; their respective sample
values m and v are related by v = g(m). Let the quantization error be denoted by the random
variable Q of sample value q. We may thus write

q=m–v (1.7)

or, correspondingly,

Q=M–V (1.8)

With the input M having zero mean, and the quantizer assumed to be symmetric as in Figure
1.11, it follows that the quantizer output V and therefore the quantization error Q, will also
have zero mean. Thus for a particular statistical characterization of quantizer in terms of
output signal to (quantization) noise ratio, we need only find the mean-square value of the
quantization error Q.
Consider then an input m of continuous amplitude in the range (–Vmax, Vmax).
Assuming a uniform quantizer of the midrise type illustrated in Figure 1.11 (b), we find that
the step-size of the quantizer is given by

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2Vmax
 (1.9)
L

where L is the total number of representation levels. For a uniform quantizer, the quantization
error Q will have its samples values bounded by –Δ/2 ≤ q ≤ Δ/2. If the step size is sufficiently
small (i.e., the number of representation levels L is sufficiently large), it is reasonable 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. We may thus express the probability density function of the quantization error Q as
follows:

1  
 ,  q
fQ ( q)    2 2 (1.10)
 0, otherwise

For this to be true, we must ensure that the incoming signal does not overload the quantizer.

The area under probability density function of a uniformly distributed random

variable is equal to unity, because sum of probabilities of each sample should be unity. So,
the magnitude of probability density function of the quantization error Q is 1/Δ over an
interval –Δ/2 to Δ/2 as shown in Figure 1.13).

fQ ( q)
1/ 

q
 
2 2
Figure 1.13: Probability density function of Quantization error Q

Then, with the mean of the quantization error being zero, its variance  Q2 is the same as the
mean-square value:

 /2
 Q2  E Q 2    q 2 f Q ( q ) dq (1.11)
  /2

 /2
 q 2 f Q ( q ) dq
  /2

(Note: The power of a random variable (let x) equals to its mean-squared value i.e., E[x2].
Usually, the noise has zero mean (µ) and  n2 variance, where  n2  E (n   )2   E  n 2  . So,
noise power equal to its variance). Substituting Eq. (1.10) into Eq. (1.11), we get

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1  /2 2 2
  /2
 Q2  q dq  (1.12)
12

Let n denote the number of bits per sample used in the construction of binary code, then the
number of quantization levels can be L = 2n, or equivalently,

n  log 2 L (1.13)

Hence the step size given in Eq. (1.9) is:

2Vmax
 (1.14)
2n

Thus the use of Eq. (1.14) in Eq. (1.12) yields

1
 Q2  Vmax
2
2 2 n (1.15)
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  3P 
( SNR )O    2  22 n (1.16)
 2
Q  Vmax 

Eq. (1.16) shows that the output signal to noise ratio of the quantizer increases exponentially
with increasing number of bits per sample, n. Recognizing that an increase in n requires a
proportionate increase in the channel (transmission) bandwidth BT , we thus see that 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 presume that the FM and PPM systems are limited by receiver
noise, whereas the binary-coded modulation system is limited by quantization noise.
Consider the special case of a full-load sinusoidal modulating signal of amplitude Am,
which utilizes all the representation levels provided. The average signal power is (assuming a
load of 1 ohm)

A 
2
A 2
m / 2 Am2
P rms
  (1.17)
R R 2

The total range of the quantizer input is 2 Am, because the modulating signal swings between
– Am and Am. We may therefore set Vmax = Am, in which case the use of Eq. (1.15) yields the
average power (variance) of the quantization noise as

1
 Q2  Am2 2 2 n (1.18)
3

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The output signal to noise ratio of a uniform quantizer, for a full-load test tone, is

Am2 2 3
( SNR )O  2 2 n   2 2 n  (1.19)
Am 2 3 2

Expressing signal to noise ratio in decibels, we get

10log10 ( SNR )O  1.8  6n (1.20)

If n = 2, (SNR)dB required is 13.8, if n = 3, (SNR)dB required is 19.8 and if n = 4, (SNR)dB

required is 25.8. So, for increment of one bit to represent each sample, we require 6 dB of
addition transmitting power.

Non-uniform Quantization:
Speech communication is a very important and specialized area of digital communications.
Human speech is characterized by unique statistical property as illustrated in Figure 1.14. The
abscissa represents speech signal magnitudes, normalized to the root-mean-square (rms)
values, and the ordinate is probability. For most voice communication channels, very low
speech volumes predominate and large amplitude values are relatively rare. That means 50%
of the time, the voltage characterizing detected speech energy is less than one-fourth of the
rms value and only 15% of the time does the voltage exceed the rms value. As the
quantization noise depends on step size, a uniform quantizer would be wasteful for speech
signals, because many of the quantizing steps would rarely be used.
In a system that uses equally spaced quantization levels, the quantization noise is the
same for all signal magnitudes. Therefore, with uniform quantization, the SNR is worst for
low-level signals than for high-level signals. However, non-uniform quantization can provide
fine quantization of weak signals and coarse quantization for the strong signals. Thus in the
case of non-uniform quantization, quantization noise can be made proportional to signal size.
The effect is to improve the overall SNR by reducing the noise for the predominant weak
signals, at the expense of an increase in noise for the rarely occurring strong signals.

Figure 1.14: Statistical distribution of single-talker speech signal magnitudes

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One way of achieving non-uniform quantization is to use a non-uniform quantizer

characteristic, shown in Figure 1.15 (a). But, this type of quantizer with varying step-size is
difficult to implement. More often, non-uniform quantization is achieved by first distorting
the original signal with a logarithmic compression characteristic, as shown in Figure 1.15 (b),
and then using a uniform quantizer. For small magnitude signals the compression
characteristic has a much steeper slop than for large magnitude signals. Thus, a given signal
change at small magnitudes will carry the uniform quantizer through more steps than the
same change at large magnitudes. The compression characteristic effectively changes the
distortion of the input signal magnitudes so that there is not a preponderance of low
magnitude signals at the output of the compressor. After compression, the distorted signal is
used as the input to a uniform (linear) quantizer characteristic, shown in Figure 1.15 (c). At
receiver, an inverse compression characteristic, called expansion, is applied so that the
overall transmission is not distorted. The processing pair (compression and expansion) is
usually referred to as companding.

(a) (b) (c)

Figure 1.15: (a) Non-uniform quantizer characteristic (b) Compression characteristic (c) Uniform
quantizer characteristic

The early PCM systems implemented a smooth logarithmic compression function.

Today, most PCM systems use a piecewise linear approximation to the logarithmic
compression characteristics. America and Europe both agreed on the need for compander in
voice telephony systems, but could not agree on the details. Hence two logarithmic
compression laws have been standardised. America and Japan use µ-law compander and
Europe and rest of the world’s national systems and international systems use A-law. Figure
1.16 shows that both the µ-law and A-law transfer functions are logarithmic. µ-law
characteristics can be defined by:

1  x  x
y  y max ln  1   , 0 1 (1.21)
ln(1   )  xmax  xmax

where µ is a positive constant, x and y represents input and output voltages, and xmax and ymax
are the maximum positive excursion of the input and output voltages, respectively. The
compression characteristic is shown in Figure 1.16 (a) for several values of µ. In North
America, the standard value for µ is 255. Notice that µ = 0 corresponds to uniform
quantization. Similarly, the A-law characteristic defined as:

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 A  x  x 1
 ymax  , 0 
 1  ln( A)  xmax  xmax A
y (1.22)
y 1   x  1 x
 1  ln  A   ,  1
 max
1  ln( A) 
  xmax   A xmax

where A is a positive constant and x and y are as defined in Eq. (1.21). The A-law
compression characteristic is shown in Figure 1.16 (b) for several values of A. A standard
value for A is 87.6.

(a) (b)
Figure 1.16: Compression characteristics (a) µ-law characteristics (b) A-law characteristics

2.2 Delta Modulation

In delta modulation (DM), an incoming signal is oversampled (i.e., at a rate much higher than
the Nyquist rate) to purposely increase the correlation between adjacent samples of the signal.
This is done to permit the use of a simple quantizing strategy for constructing the encoded
signal.
In its basic form, DM provides a staircase approximation to the oversampled version
of the message signal, as illustrated in Figure 2.5 (a). The difference between the input and
the approximation is quantised into only two levels, namely, ± ∆, corresponding to positive
and negative differences. Thus if the approximation falls below the signal at any sampling
epoch, it is increased by ∆. If on the other hand, the approximation lies above the signal, it is
diminished by ∆. Providing that the signal does not change too rapidly from the sample to
sample, we find that the staircase approximation remains within ± ∆ of the input signal.

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Figure 2.5: Illustration of delta modulation

Let m(t) denote the input (message) signal, and mq(t) demote its staircase approximation. For
convenience of presentation, we adopt the following notation that is commonly used in the
digital signal processing literature:

m[n] = m(nTs), n = 0, ± 1, ± 2,… (2.9)

Where Ts is the sampling period and m(nTs) is a sample of the signal m(t) taken at time t =
nTs, and likewise for the samples of the other continuous time signals, We may then
formalize the basic principles of the delta modulation in the following set of discrete time
relations:

e[n] = m[n] – mq[n – 1] (2.10)

eq[n] = ∆ sgn(e[n]) (2.11)

mq[n] = mq[n – 1] + eq[n] (2.12)

Where e[n] is an error signal representing the difference between the present sample m[n] of
the input signal ad the latest approximation mq[n – 1] to it, eq[n] is the quantized version of
e[n], and sgn(.) is the signum function. Finally, the quantizer output mq[n] is coded to
produce the DM signal.
Figure 2.6 (a) illustrate the way in which the staircase approximation mq(t) follows
variation in the input signa m(t) ain accordance with Eq. (2.10) – (2.12), and Figure 2.6 (b)
displays the corresponding binary sequence at the delta modulator output. It is apparent that
in a delta modulation system the rate of information transmission is simply equal to the
sampling rate fs = 1/Ts.
The pronciple virtue of DM is its simplicity. It may be generated by applying the
sampled version of the incomig message signal to a modulator that onvolves a comparator,
quantizer and accumulator interconnected as shown in Figure 2.7 (a). The block labeled z –1
iside the accumulator represents a unit delay, that is, a delay equal to one sampling period.
(The variable z is commonly used in the z – transform, which is basic to the analysis of

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discrete time signals and syatems). Details of the modulator follows directly from Eq. (2.10)
– (2.12). The comparator computes the difference between its two inputs. The quantizer
consists of a hard limiter with an input – output relation that is a scaled version of the signum
function. The quantizer output is then applied to an accumulator, producig the result

n n
mq [n ]    sgn  e[i ]   eq [i ] (2.13)
i 1 i 1

Which is obtained by solving Eq. (2.11) and (2.12) for mq[n]. Thus, at the sampling instant
nTs, the accumulator increments the approximation by a step ∆ in a positive or negative
direction, depending on the algebraic sign of the error sample e[n]. If the input sample m[n] is
greater than the most recent approximation mq[n], appositive increment + ∆ is applied to the
approximation. If, on the other hand, the input sample is smaller, a negative increment – ∆ is
applied to the approximation. In this way, the accumulator does the best it can track the input
samples by one step (of amplitude + ∆ or – ∆) at a time. In the receiver show in Figure 2.6
(b), the staircase approximation mq(t) is reconstructed by passing the sequence of positive and
negative pulses, produced at the decoder output through an accumulator in a manner similar
to that used in the transmitter. The out of band quantization noise in the high frequency
staircase waveform mq(t) is rejected by passing it through a low-pass filter, as in Figure 2.7
(b), with a bandwidth equal to the original message bandwidth.

(a)

(b)
Figure 2.7: DM system (a) Transmitter (b) Receiver

Delta modulation is subject to two types of quantization error: slope overload

distortion and granular noise.

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Slope overload distortion: From Eq. (2.12), the digital equivalent of integration in the sense
that it represents the accumulation of positive and negative increments of magnitude ∆. Also,
denoting the quantization error by q[n], as shown by:

mq[n] = m[n] + q[n] (2.14)

we observe from Eq. (2.10) that the input to the quantizer is

e[n] = m[n] – m[n – 1] – q[n – 1] (2.15)

Thus except for the quantization error q[n – 1], the quantizer input is a first backward
difference of the input signal, which may be viewed as a digital approximation to the
derivative of the input signal or, equivalently, as the inverse of the digital integration process.
If we consider the maximum slope of the original input waveform m(t), it is clear that in order
for the sequence of samples {mq[n]} to increase as fast as input sequence of samples {m[n]}
in a region of maximum slope of m(t), we require that the condition

 dm(t )
 max (2.16)
Ts dt

be satisfied. Otherwise, we find that the step-size ∆ is too small for the staircase
approximation mq(t) to follow a steep segment of the input waveform m(t), with the result that
mq(t) falls behind m(t), as illustrated in Figure 2.7. This condition is called slope overload,
and the resulting quantization error is called slope-overload distortion (noise). Note that since
the maximum slope of the staircase approximation mq(t) is fixed by the step size ∆, increases
and decreases in mq(t) tend to occur along straight lines. For this reason, a delta modulator
using a fixed step size is often referred to as a linear delta modulation
Granular noise: In contrast to slope-overload distortion, granular noise occurs when step size
∆ is too large relative to the local slope characteristics of the input waveform m(t), thereby
causing the staircase approximation mq(t) to hunt around a relatively flat segment of the input
waveform; this phenomenon is also illustrated in Figure 2.7. Granular noise is analogous to
quantization noise in a PCM system.
We thus see that there is a need to have a large step-size to accommodate a wide
dynamic range, whereas a small step size is required for the accurate representation of
relatively low level signals. It is therefore clear that the choice of the optimum step size that
minimizes the mean square value of the quantization error in a linear delta modulator will be
the result of a compromise between slope-overload distortion and granular noise. To satisfy
such requirement, we need to make the delta modulator “adaptive” in the sense that the step
size is made to vary in accordance with the input signal.

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Figure 2.7: Illustration of the two different forms of quantization error in delta modulation

Multiplexing
Multiplexing, or muxing, is a way of sending multiple signals or streams of information over
a communications link at the same time in the form of a single, complex signal. When the
signal reaches its destination, a process called demultiplexing, or demuxing, recovers the
separate signals and outputs them to individual lines.
Multiplexing is a method used by networks to consolidate multiple signals -- digital or
analog -- into a single composite signal that is transported over a common medium, such as
a fiber optic cable or radio wave. When the composite signal reaches its destination, it is
demultiplexed, and the individual signals are restored and made available for processing.
Networks use a variety of multiplexing techniques, but at a conceptual level, they all
operate in a similar manner. The individual network signals are input into a multiplexer
(mux) that combines them into a composite signal, which is then transmitted through a shared
medium. When the composite signal reaches its destination, a demultiplexer (demux) splits
the signal back into the original component signals and outputs them into separate lines for
use by other operations.

Uses of Multiplexing: Multiplexing is used in a wide range of industries to facilitate both

analog and digital communications. It was first introduced in the 1870s to support telegraphy
but has since become a mainstay in telecommunications, such as radio, television and
telephone. It is also used in computer networks, often to transmit multiple signals across a
wide area network (WAN).
Organizations implement multiplexing on their networks for two reasons: to enable network
devices to communicate with each other without needing a dedicated connection between
each device pair, although multiplexing still requires shared media; and to better utilize
scarce or expensive network resources. For example, multiplexing can be used to transmit

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multiple signals across a satellite uplink or on a cable or fiber strand running between major
metropolitan areas.

Types of Multiplexing:
Organizations can select from multiple forms of multiplexing. Their choices will depend in
large part on the types of signals being transmitted -- analog vs. digital -- and the media used
to carry those transmissions, such as coaxial cable, fiber optic cable or microwave link. The
following is an overview of several common multiplexing techniques.
Frequency Division Multiplexing (FDM): In FDM, multiple transmitted signals use a
common channel but the total available bandwidth is utilized among the various signals. This
implies that over a complete channel a particular frequency slot is allotted to only one signal.
Basically, in FDM, a different frequency band is used to modulate different data
signal. This means that different carrier frequency modulates the various signals that are to be
transmitted over the channel. Further, the modulated signals are mixed and transmitted over a
single communication link. The figure below shows the process of frequency division
multiplexing:

From the figure shown above, it is clear that a single channel is divided into multiple
parts. And each part is nothing but a separate channel carrying a signal or data stream. The
technique of FDM is also known as analog multiplexing as it is used for multiplexing of
analog signals.
So, to reduce the chances of interference between the simultaneously transmitted
signals, guard bands are provided between the frequency slots. Guard bands are nothing but
unused frequency slot in the entire band that avoids overlapping of one frequency channel
with the other.
Once the signal gets transmitted over the channel, then at the receiver end,
demultiplexing of the signal is performed in order to separate various signals from one
another.
Time Division Multiplexing (TDM): It is a multiplexing technique that allows transmission
of multiple signals over a common channel but in different time slots. Each signal will get
transmitted very quickly over the channel but at a time only one signal will be transmitted. So
basically in TDM, the overall transmission time is divided according to the multiple signals
required to be transmitted over the common link.

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Each transmitted source is allotted with one or more time slot in each frame for the
transmission of the signal. The figure below shows the transmission of 4 different source
signals using time division multiplexing:

Here as we can see that signals from 4 different sources are multiplexed together. So, the 4
separate TDM frames are generated that gets transmitted over the channel in different time
slots.
Code-division multiplexing (CDM): A sequence of bits called the spreading code is
assigned to each signal to distinguish one signal from another. The spreading code is
combined with the original signal to produce a new stream of encoded data, which is then
transmitted on a shared medium. A demux that knows the code can then retrieve the original
signals by subtracting out the spreading code, a process called dispreading. CDM is widely
used in digital television and radio broadcasting and in 3G mobile cellular networks --
4G and 5G primarily use OFDM. CDM can also support multiple signals from multiple
sources, a technique known as code-division multiple access.
Space-division multiplexing (SDM): Signal paths are spatially separated through the use of
multiple conductors, such as optical fibers or electrical wires. The conductors are bundled
into a single transport medium but are physically separated, with each conductor handling a
transmitted channel. Individual conductors can be further multiplexed through the use of
FDM, TDM or other techniques. SDM is often used in submarine cable systems to help
increase capacity, but it can also be used for wireless communications.

Problems
1. Find the Nyquist rate and Nyquist interval for the given signal.
 1 
x (t )    cos(200 t ) cos(300 t ) .
 2 
2. A waveform, x(t) =10 cos (1000t+π/3) +20 cos (2000t + π /6) is to be uniformly sampled for
digital transmission.
(a) What is the maximum allowable time interval between sample values that will ensure
perfect signal reproduction?
(b) If we want to reproduce 1 hour of this waveform, how many sample values need to be
stored?

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3. A message signal m(t) = 10 cos (4π103t) is transmitted through a channel using 4 bit
PCM. The sampling rate is 50% higher than Nyquist rate calculate bit rate and
maximum bandwidth of PCM system.
4. A video signal is band limited to 4.5 MHz and transmitted through a channel using
PCM. (a) Calculate the sampling rate if the signal is to be sampled at least 20% higher
than the Nyquist rate. (b) If the signal is quantized into 1024 levels, calculate bitrate and
maximum bandwidth of the PCM signal.
5. A sinusoidal signal is band limited to 5 kHz and it is transmitted through a channel
using PCM. The sampling rate is twice the Nyquist rate. The maximum quantization
error should be 0.1% of peak signal amplitude. Calculate bit rate and maximum
bandwidth of PCM signal.
6. A message signal m(t) = 4 cos (8π103t) is transmitted through a channel using 3 bit
PCM. The sampling rate is twice the Nyquist rate. (a) Calculate bit rate maximum
bandwidth of PCM system. (b) If the sampled values are 3.9, 2.5, 0.2, –1.2, –3.8, –3.3,
determine the quantizer output, encoder output and quantization error per each sample.
(c) Sketch the transfer characteristics of the quantizer.
7. In a PCM system, the number of bits increased from 6 to 8. Then by what factor the
quantization noise power decreases.
8. A 40 MB hard disk is used to store PCM data. The signal is sampled at 8 kHz and
encoded PCM is to have an average signal to noise ratio of at least 30 dB. For how
many minutes the PCM date can be stored on the hard disk.
9. A compact disc (CD) records audio signals digitally using PCM. Assume the audio
signal is band limited to be 3 KHz and has maximum signal amplitude 1 V. The
sampling rate is 50% higher than the Nyquist rate. If the speech signal is quantized into

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4096 levels, then determine the quantization step, the signal-to-quantization noise ratio
and the bit rate required to encode the given audio signal.
10. A 7 bit PCM system employing uniform quantizer having an overall signalling rate of
56 kbps. (a) Calculate (SQNR) for sinusoidal signal (b) calculate sampling frequency fs.
(c) find the theoretical maximum frequency that this system can handle.
11. An Analog signal whose voltage is varying between 0V–2V and it is given as input to
2bit PCM with Midrise type quantizer of Δ = 0.5. Let x[k]={0.2,0.7,1.3,1.7} be the
sample values of the given analog signal. (a) Find the Quantized signal xq[k]. (b) Find
the Encoder output in terms of Binary digits. (c) Draw NRZ Unipolar and Polar signal
for the Encoder output.
12. Find the Transmitter and Receiver output of the one bit version of DPCM system for the
following signal by considering δ = ±1. x(k)= {0.3, 1.5, 0.7, 1, 2.3, 3.7, 2.8, 3.4, 2.8, 0}.
13. The message input to a delta modulator is 5 cos 2π(1000t). The pulse rate is 50,000
pulses/sec. Determine the optimum value of the step size.
14. The message input of a delta modulator is 6 sin 2π(100t). The step size is 0.314.
Determine the pulse rate which will prevent both the distortions.
15. The input to a delta modulator is a sinusoidal signal whose frequency can be vary from
200 Hz to 4000 Hz. The input signal is sampled at 8 times the Nyquist rate. The peak
amplitude of sinusoidal signal is 1 Volt. (a) Determine the optimum value of the step
size when the signal frequency is 800 Hz. (b) Determine the peak amplitude of the
signal when the frequency is 500 Hz.
16. The input signal to the delta modulator is: x(t) = 250t [u(t) – u(t – 1)] + (500 – 250t)[u(t
–1) + u(t – 2)]. If the sampling rate is 25k samples per second then determine the
optimum value of step size to eliminate slope overload and granular noise distortions.

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