DIGITAL PULSE MODULATION

Modulation is the process of varying one or more parameters of a carrier signal in accordance
with the instantaneous values of the message signal.

1. PULSE CODE MODULATION(PCM)

The message signal is the signal which is being transmitted for communication and the carrier
signal is a high frequency signal which has no data, but is used for long distance transmission.
There are many modulation techniques, which are classified according to the type of modulation
employed. Of them all, the digital modulation technique used is Pulse Code Modulation
(PCM).
A signal is pulse code modulated to convert its analog information into a binary sequence, i.e., 1s
and 0s. The output of a PCM will resemble a binary sequence. The following figure shows an
example of PCM output with respect to instantaneous values of a given sine wave.

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process is
called as digital. Each one of these digits, though in binary code, represent the approximate
amplitude of the signal sample at that instant.
In Pulse Code Modulation, the message signal is represented by a sequence of coded pulses. This
message signal is achieved by representing the signal in discrete form in both time and
amplitude.

11
Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists of Sampling, Quantizing and
 Encoding, which are performed in the analog-to-digital converter section. The low pass filter
prior to sampling prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of impaired signals, decoding, and
reconstruction of the quantized pulse train. Following is the block diagram of PCM which
represents the basic elements of both the transmitter and the receiver sections.

➢
Low Pass Filter
This filter eliminates the high frequency components present in the input analog signal which
is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.

➢ Sampler
This is the technique which helps to collect the sample data at instantaneous values of message
signal, so as to reconstruct the original signal. The sampling rate must be greater than twice the
highest frequency component W of the message signal, in accordance with the sampling
theorem.

➢ Quantizer
Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer reduces the redundant bits and compresses the value.

12
➢ Encoder

Encoder assigns code words to quantized sampled values. This coding techniques uses bits 0
and 1. If number of quantized levels are 16 then each sample is assigned with 4 bit code word.

➢ Regenerative repeater:
 The PCM has an ability to control the distortion and noise caused by the transmission of bits along
the channel. This ability is accomplished by several regenerative repeaters located at sufficient
placing along channel.

Regenerative repeaters have three functions.

1. Equalizing
2. Timing circuits
3. Decision making device

Equalizer shapes the received pulse so as to compensate amplitude and phase distortion caused by the
channel.

Timing circuits provides periodic pulse trains.

• Decision making device compares amplitude of equalized pulse plus noise to the pre-defined
threshold levels to make decisions whether the pulse is present or not.
• If the pulse is present (i.e. decision is yes), clean new pulse is generated and transmitted
through channel to next regenerative pulse. If the pulse is not present (i.e. the decision is no),
it generates clean base line to next regenerative repeater, provided the noise too large caused
bit error by taking the wrong decision

➢ Decoder

Decoder reboots all the received bits to make more words then it decodes as quantized PAM signals.

13
➢ Reconstruction Filter:
All coded words are passed through low pass filter so that analog signal can be reconstructed from
quantized PAM signal.The cut off frequency of low pass filter is fm Hz which is equal to band width
of message signal.

➢ Destination
It receives the signal from the reconstructive filter output is analog signal.
 Fig.PCM waveform

Bit rate and bandwidth requirements of PCM :

➢ The bit rate of a PCM signal can be calculated form the number of bits per sample × the
sampling rate. Bit rate =�������� The bandwidth required to transmit this signal
depends on the type of line encoding used.
➢ A digitized signal will always need more bandwidth than the original analog signal. Price
wepay for robustness and other features of digital transmission.

Important Relations

• Quantization Noise (����)=Δ2/2

• Signal to Noise ratio

(��������)=32.22�� ���� �������� ����
����=1.76+6.02��≅(1.8+6��)����

• ������ ��������=����.���� �������� ������

���������������������������� ��������=������
• Bandwidth for PCM signal =n.fm
Where,
n – No. of bits in PCM code
Fm – signal bandwidth
fs – sampling rate

14
SAMPLING, QUANTIZATION AND CODING
1. Sampling
• Definition: Sampling is defined as ―The process of measuring the instantaneous
values of continuous-time signal in a discrete form.‖
• Sample is a piece of data taken from the whole data which is continuous in the time domain.
When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e., High
or Low, the signal has to be discretized in time. This discretization of analog signal is called as
Sampling.
The following figure indicates a continuous-time signal x (t) and a sampled signal xs (t). When x (t)
is multiplied by a periodic impulse train, the sampled signal xs (t) is obtained.

Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be
termed as a sampling period Ts.
Sampling Frequency fs=1/Ts
Where,
Ts is the sampling time
fs is the sampling frequency or the sampling rate

Sampling frequency -is the reciprocal of the sampling period. This sampling frequency, can be
simply called as Sampling rate. The sampling rate denotes the number of samples taken per
second, or for a finite set of values.
15
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be
highly considered. The rate of sampling should be such that the data in the message signal should
neither be lost nor it should get over-lapped. Hence, a rate was fixed for this, called as Nyquist
rate

Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher than W Hertz. That
means, W is the highest frequency. For such a signal, for effective reproduction of the original
signal, sampling rate should be twice the highest frequency.
This means,
fs=2W
Where,
fs is the sampling rate
W is the highest frequency
This rate of sampling is called as Nyquist rate.
A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate.

Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient
sample rate in terms of bandwidth for the class of functions that are band limited.

The sampling theorem states that, ― a signal can be exactly reproduced if it is sampled at the
rate fs which is greater than twice the maximum frequency W.

To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal
whose value is non-zero between some –W and W Hertz.
Such a signal is represented as x(f)=0for|f|>W

For the continuous-time signal x (t), the band-limited signal in frequency domain, can be
represented as shown in the following figure.

16
.

We need a sampling frequency, a frequency at which there should be no loss of information, even
after sampling. For this, we have the Nyquist rate that the sampling frequency should be two
times the maximum frequency. It is the critical rate of sampling.

If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered, and if it
is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency domain.

The above figure shows the Fourier transform of a signal xs(t). Here, the information is
reproduced without any loss. There is no mixing up and hence recovery is possible.

The Fourier Transform of the signal

xs(t) is Xs (w)=1Ts∑n=−∞∞X(w−nw0)
Where Ts = Sampling Period and w0=2πTs

Let us see what happens if the sampling rate is equal to twice the highest
frequency (2W) That means,
Fs =2W
17
Where,
Fs is the sampling frequency
W is the highest frequency
 The result will be as shown in the above figure. The information is replaced without any loss.
Hence, this is also a good sampling rate.
Now, let us look at the condition,
Fs <2W
The resultant pattern will look like the following figure

We can observe from the above pattern that the over-lapping of information is done, which
leads to mixing up and loss of information. This unwanted phenomenon of over-lapping is
called as Aliasing

18
Aliasing

Aliasing can be referred to as ―the phenomenon of a high-frequency component in the

spectrum of a signal, taking on the identity of a low-frequency component in the spectrum of
its sampled version.‖
The corrective measures taken to reduce the effect of Aliasing are −
• In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the
sampler, to eliminate the high frequency components, which are unwanted. • The signal
which is sampled after filtering, is sampled at a rate slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier
design of the reconstruction filter at the receiver.
 Scope of Fourier Transform

It is generally observed that, we seek the help of Fourier series and Fourier transforms in
analyzing the signals and also in proving theorems. It is because −

• The Fourier Transform is the extension of Fourier series for non-periodic signals. •
Fourier transform is a powerful mathematical tool which helps to view the signals in
different domains and helps to analyze the signals easily.
• Any signal can be decomposed in terms of sum of sines and cosines using this Fourier
transform. The digitization of analog signals involves the rounding off of the values which are
approximately equal to the analog values. The method of sampling chooses a few points on the
analog signal and then these points are joined to round off the value to a near stabilized value.
Such a process is called as Quantization.

19
Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to create a series of digital values out of
the given analog signal. The following figure represents an analog signal. This signal to get converted into
digital has to undergo sampling and quantizing

The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels.
Quantization is representing the sampled values of the amplitude by a finite set of levels, which
means converting a continuous-amplitude sample into a discrete-time signal.
The following figure shows how an analog signal gets quantized. The blue line represents analog
signal while the brown one represents the quantized signal.

Both sampling and quantization result in the loss of information. The quality of a Quantizer
output depends upon the number of quantization levels used. The discrete amplitudes of the
quantized output are called as representation levels or reconstruction levels. The spacing
between the two adjacent representation levels is called a quantum or step-size.
20
The following figure shows the resultant quantized signal which is the digital form for the given
analog signal.

This is also called as Stair-case waveform, in accordance with its shape.

Types of Quantization
There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.
 1. The type of quantization in which the quantization levels are uniformly spaced is termed as a
Uniform Quantization.

2. The type of quantization in which the quantization levels are unequal and mostly the relation
between them is logarithmic, is termed as a Non-uniform Quantization.

There are two types of uniform quantization.

1. Mid-Rise type
2. Mid-Tread type.

21
The following figures represent the two types of uniform quantization

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of uniformquantization.
1. The Mid-Rise type is so called because the origin lies in the middle of a raising part
of the stair- case like graph. The quantization levels in this type are even in number. 2.
The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.

Both the mid-rise and mid-tread type of uniform quantizer are symmetric about the origin. Δ=
(������−������)��

����=������2��
Quantization Error

For any system, during its functioning, there is always a difference in the values of its input and output.
The processing of the system results in an error, which is the difference of those values.The difference
between an input value and its quantized value is called a Quantization Error.
A Quantizer is a logarithmic function that performs Quantization (rounding off the value). An
analog-to- digital converter (ADC) works as a quantizer.

22
The following figure illustrates an example for a quantization error, indicating the difference
between the original signal and the quantized signal.

Quantization Noise
It is a type of quantization error, which usually occurs in analog audio signal, while quantizing it
to digital. For example, in music, the signals keep changing continuously, where a regularity is
not found in errors. Such errors create a wideband noise called as Quantization Noise.
 23
➢ COMPANDING IN PCM SYSTEMS

The word Companding is a combination of Compressing and Expanding, which means that it does both.
This is a non-linear technique used in PCM which compresses the data at the transmitter and expands the
same data at the receiver. The effects of noise and crosstalk are reduced by using this technique

Fig. Companding
Companding means it amplifies the low level signals as well as attenuate high level at the
transmitter side. At the receiver side reverse operation done. It attenuates the low level signals and
amplifies the high level signals you get the original signal. Non-uniform quantization cannot be
applied directly by using companding technique.

Fig Companding curves for PCM

Companding is used to maintain constant Signal to Noise Ratio throughout dynamic quantization
ratio
 24
Fig. Non Uniform Quantization
There are two types of Companding techniques. They are –

1.A-law Companding Technique

i. Uniform quantization is achieved at A = 1, where the characteristic curve is linear and
no compression is done.

ii. A-law has mid-rise at the origin. Hence, it contains a non-zero value.
iii. A-law Companding is used for PCM telephone systems.

Y= A│x│ ; where 0≤ x ≤ 1/A 1+ln(A)

= 1+ln A│x│ ; 1/A≤ x ≤ 1 1+ln(A)

practically A=87.56

if A=1 we get uniform quantization

2.µ-law Companding Technique

i. Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and no
compression is done.

ii. µ-law has mid-tread at the origin. Hence, it contains a zero value. iii.
µ-law companding is used for speech and music signals.

Y= ±ln(1+µ│x│) ;│x│≤1 ln(1+µ)

Practically µ value is 256

For the samples that are highly correlated, when encoded by PCM technique, leave
redundant information behind. To process this redundant information and to have a better
output, it is a wise decision to take a predicted sampled value, assumed from its previous
 output and summarize them with the quantized values. Such a process is called as
Differential PCM (DPCM) technique.

25
➢ SAMPLING PROCESS

Due to the increased use of computers in all engineering applications, including signal
processing, it is important to spend some more time examining issues of sampling. In this
chapter we will look at sampling both in the time domain and the frequency domain.

We have already encountered the sampling theorem and, arguing purely from a trigonometric
identity point of view, have established the Nyquist sampling criterion for sinusoidal signals.
However, we have not fully addressed the sampling of more general signals, nor provided a
general proof. Nor have we indicated how to reconstruct a signal from its samples. With the tools
of Fourier transforms and Fourier series available to us we are now ready to finish the job that
was started months ago.

To begin with, suppose we have a signal x(t) which we wish to sample. Let us suppose further
that the signal is bandlimited to B Hz. This means that its Fourier transform is nonzero for −2πB
< ω < 2πB. Plot spectrum.

We will model the sampling process as multiplication of x(t) by the “picket fence” function

δT(t) = Xδ(t − nT).

We encountered this periodic function when we studied Fourier series. Recall that by its Fourier
series representation we can write

where . The frequency fs = ωs/(2π) = 1/T is the sampling frequency in samples/sec.

Suppose that the sampling frequency is chosen so that fs > 2B, or equivalently, ωs > 4πB.
THE SAMPLING THEOREM
If x(t) is bandlimited to B Hz then it can be recovered from signals taken at a sampling rate fs >
2B. The recovery formula is

26
where

Show what the formula means: we are interpolating in time between samples using the sinc
function.

We will prove this theorem. Because we are actually lacking a few theoretical tools, it will
take a bit of work. What makes this interesting is we will end up using in a very essential way
most of the transform ideas we have talked about.

1.The first step is to notice that the spectrum of the sampled signal,

is periodic and hence has a Fourier series. The period of the function in frequency is ωs, and the
fundamental frequency is

By the fourier.Series. we can write

where the cn are the F.S. coefficients

But the integral is just the inverse F.T., evaluated at t = −nT:

 ,

so

2.Let g(t) = sinc(πfst). Then

27

3.Let

We will show that y(t) = x(t) by showing that Y (ω) = X(ω). We can compute the F.T. of
y(t) using linearity and the shifting property:

Observe that the summation on the right is the same as the F.S. we derived in step 1:

.
Now substituting in the spectrum of the sampled signal (derived above)

since x(t) is bandlimited to −πfs < ω < πfs or −fs/2 < f < fs/2.
Fig. Sampling
28

Notice that the reconstruction filter is based upon a sinc function, whose transform is a rect
function: we are really just doing the filtering implied by our initial intuition.In practice, of course,
we want to sample at a frequency higher than just twice the bandwidth to allow room for filter
rolloff

➢ QUANTIZATION

Quantization approximates the sampled value to nearest discrete value from the set of finite
discrete levels. Quantizers are of two types.

1. Mid treed quantizer

2. Mid rise quantizer

Quantizing step size, Δ=(xmax-xmin)/q

Q=number of quantized level

Δ=(xmax-xmin)/2n
Where n is number of bits used to represent each level

1. Mid treed Quantizer

 Fig. Mid tred Quantizer

Error is ± Δ/2

29
Quantization error=quantized value-actual sampled value
Qe=xq(nTs)-x(nTs).
In mid treed quantization the input values lies between ± Δ/2, ± 3Δ/2, ± 5Δ/2, . . . in that output
values are quantized values at ± Δ,± 2Δ,±3Δ,……. Suppose the input (i.e. sampled value) lies
between ± Δ/2 which is approximated as zero. If the input values lies between Δ/2 to 3Δ/2 this
quantizer approximates sampled value as Δ. Here the origin of treed of stair case lies at midpoint
so the name is called mid treed quantizer. In that maximum quantization error is Δ/2 and
minimum quantization error is -Δ/2.

2. MidRise Quantizer:-

Fig. MidRise Quantizer

In mid rise quantizer the input values are ± Δ,± 2Δ,± 3Δ,….. the quantized values are ± Δ/2,±
3Δ/2,± 5Δ/2,…. The quantization error is ± Δ/2

30
Quantization error

Fig. Quantization effect in PCM

In mid treed quantization the input values lies between ± Δ/2, ± 3Δ/2, ± 5Δ/2, . . . in that output
values are quantized values at ± Δ,± 2Δ,±3Δ,……. Suppose the input (i.e. sampled value) lies
between ± Δ/2 which is approximated as zero. If the input values lies between Δ/2 to 3Δ/2 this
quantizer approximates sampled value as Δ. Here the origin of treed of stair case lies at midpoint
so the name is called mid treed quantizer. In that maximum quantization error is Δ/2 and
minimum quantization error is -Δ/2.
 31
2. DIFFERENTIAL PCM (DPCM)

DPCM Transmitter

The DPCM Transmitter consists of Quantizer and Predictor with two summer circuits.
Following is the block diagram of DPCM transmitter.

The signals at each point are named as −

i. x(nTs) is the sampled input
ii. x^(nTs) is the predicted sample
iii. e(nTs) is the difference of sampled input and predicted output, often called as
prediction error

iv. v(nTs) is the quantized output

v. u(nTs) is the predictor input which is actually the summer output of
 the predictor output and the quantizer output

The predictor produces the assumed samples from the previous outputs of the transmitter
circuit. The input to this predictor is the quantized versions of the input signal x(nTs).
Quantizer Output is represented as −

v(nTs)=Q[e(nTs)]
=e(nTs)+q(nTs)
Where,
q (nTs) is the quantization error

32
Predictor input is the sum of quantizer output and predictor output,
u(nTs)=xˆ(nTs)+v(nTs)
u(nTs)=xˆ(nTs)+e(nTs)+q(n
Ts) u(nTs)=x(nTs)+q(nTs)

The same predictor circuit is used in the decoder to reconstruct the original

input DPCM Receiver

The block diagram of DPCM Receiver consists of a decoder, a predictor, and a summer circuit.
Following is the diagram of DPCM Receiver.

The notation of the signals is the same as the previous ones. In the absence of noise, the
encoded receiver input will be the same as the encoded transmitter output.As mentioned before,
the predictor assumes a value, based on the previous outputs. The input given to the decoder is
processed and that output is summed up with the output of the predictor, to obtain a better output.
The sampling rate of a signal should be higher than the Nyquist rate, to achieve better
sampling. If this sampling interval in Differential PCM is reduced considerably, the sampleto
 sample amplitude difference is very small, as if the difference is 1-bit quantization, then the
step-size will be very small i.e., Δ (delta).

Advantages of DPCM

1) Bandwidth Requirement Of DPCM Is Less Compared To PCM

2) Quantization Error Is Reduced Because Of Prediction Filter.

3) Numbers Of Bits Used To Represent .One Sample Value Are Also Reduced Compared To PCM

33
3. DELTA MODULATION

The sampling rate of a signal should be higher than the Nyquist rate, to achieve better sampling.
If this sampling interval in Differential PCM is reduced considerably, the sample-to-sample
amplitude difference is very small, as if the difference is 1-bit quantization, then the step-size
will be very small i.e., Δ (delta).

The type of modulation, where the sampling rate is much higher and in which the step size after
quantization is of a smaller value Δ, such a modulation is termed as delta modulation.

Fig. Block diagram of delta modulator and demodulator

Features of Delta Modulation

Following are some of the features of delta modulation.
• An over-sampled input is taken to make full use of the signal correlation. •
The quantization design is simple.

• The input sequence is much higher than the Nyquist rate.

• The quality is moderate.
• The design of the modulator and the demodulator is simple.
 • The stair-case approximation of output waveform.
• The step-size is very small, i.e., Δ (delta).
• The bit rate can be decided by the user.
• This involves simpler implementation.

Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit DPCM scheme.
As the sampling interval is reduced, the signal correlation will be higher.
34
➢ Delta Modulator
The Delta Modulator comprises of a 1-bit quantizer and a delay circuit along with two summer
circuits. Following is the block diagram of a delta modulator.

The predictor circuit in DPCM is replaced by a simple delay circuit in DM.

From the above diagram, we have the notations as −

• x(nTs) = over sampled input
• ep(nTs) = summer output and quantizer input
• eq(nTs) = quantizer output = v(nTs)
• x^(nTs) = output of delay circuit
• u(nTs) = input of delay circuit

Using these notations, now we shall try to figure out the process of delta modulation.
ep(nTs)=x(nTs)−xˆ(nTs) ------------------(1)
=x(nTs)−u([n−1]Ts)
 =x(nTs)−[xˆ[[n−1]Ts]+v[[n−1]Ts]] ---------------(2) Further,

v(nTs)=eq(nTs)=S∑.[ep(nTs)]------------------ (3)
u(nTs)=xˆ(nTs)+eq(nTs)
Where,
xˆ(nTs) = the previous value of the delay circuit

35
eq(nTs) = quantizer output = v(nTs)
Hence,
u(nTs)=u([n−1]Ts)+v(nTs)---------------------- (4)
The present input of the delay unit = (The previous output of the delay unit) + (the present
quantizer output)
Assuming zero condition of Accumulation,
u(nTs)=S∑j=1n∑ [ep(jTs)]
Accumulated version of DM output = ∑j=1nv(jTs)--------------------(5)
Now, note that
xˆ(nTs)=u([n−1]Ts)=∑j=1n−1v(jTs) ------------(6)
Delay unit output is an Accumulator output lagging by one sample

From equations 5 & 6, we get a possible structure for the demodulator.

A Stair-case approximated waveform will be the output of the delta modulator with the step-size as
delta (Δ). The output quality of the waveform is moderate

Delta Demodulator
The delta demodulator comprises of a low pass filter, a summer, and a delay circuit. The
predictor circuit is eliminated here and hence no assumed input is given to the demodulator.
Following is the diagram for delta demodulator.
From the above diagram, we have the notations as −
• v^(nTs) is the input sample

36
• u^(nTs) is the summer output
• x¯(nTs) is the delayed output
A binary sequence will be given as an input to the demodulator. The stair-case approximated
output is given to the LPF.
Low pass filter is used for many reasons, but the prominent reason is noise elimination for out
of-band signals. The step-size error that may occur at the transmitter is called granular noise,
which is eliminated here. If there is no noise present, then the modulator output equals the
demodulator input.

Advantages of DM Over DPCM

• 1-bit quantizer
• Very easy design of the modulator and the demodulator However, there exists some noise in DM. •
Slope Over load distortion (when Δ is small)
• Granular noise (when Δ is large)

Advantages of Delta Modulation

• In Delta modulation electronic circuit requirement for modulation at transmitter and for
demodulation at receiver is substantially simpler compare to PCM.

• In delta modulation, amplitude of speech signal does not exceed maximum sinusoidal
amplitude.

• Signaling rate and bandwidth of DPCM or delta modulation is less than PCM technique.
Disadvantages of Delta Modulation

• If changes in signal is less than the step size, then modulator no longer follow signal.
Thus produces train of alternating positive and negative pulses.

• Modulator overloads when slope of signal is too high.

• High bit rate.

• It requires predictor circuit and hence it is very complex.

• Its practical usage is limited.

37
Delta modulation has two major drawbacks that are
1. Slope overload distortion

This distortion arises because of large dynamic range of input signal.

Fig.1: Quantization Errors in Delta Modulation

We can observe from fig.1 , the rate of rise of input signal x(t) is so high that the staircase signal
can not approximate it, the step size ‗Δ‘ becomes too small for staircase signal u(t) to follow the
step segment of x(t).Hence, there is a large error between the staircase approximated signal and
the original input signal x(t).This error or noise is known as slope overload distortion .To reduce
this error, the step size must be increased when slope of signal x(t) is high. Since the step size of
delta modulator remains fixed, its maximum or minimum slopes occur along straight lines.
Therefore, this modulator is known as Linear Delta Modulator (LDM).

2. Granular noise

Granular noise occurs when step size is too large compared to small variations in the input
signal. This means that for very small variations in the input signal, the staircase signal is
changed by large amount because of large step size. The error between the input and
approximated signal is called granular noise. The solution to this problem is to make step size
 small. Adaptive Delta Modulation
To overcome the quantization error due to slope overload distortion and granular noise, the step
size (Δ) is made adaptive to variations in input signal x(t). Particularly in the step segment of
the x(t) , the step size is increased. Also, if the input is varying slowly, the step size is reduced.
Then this method is known as Adaptive Delta Modulation (ADM).
The adaptive delta modulators can take continuous changes in the step size or discrete changes
in the step size
38
4. ADAPTIVE DELTA MODULATION

In digital modulation, we have come across certain problem of determining the step-size, which
influences the quality of the output wave.

A larger step-size is needed in the steep slope of modulating signal and a smaller step size is
needed where the message has a small slope. The minute details get missed in the process. So, it
would be better if we can control the adjustment of step-size, according to our requirement in
order to obtain the sampling in a desired fashion. This is the concept of Adaptive Delta
Modulation.

The performance of a delta modulator can be improved significantly by making the step size of
the modulator assume a time-varying form. In particular, during a steep segment of the input
signal the step size is increased. Conversely, when the input signal is varying slowly, the step size
is reduced.

In this way, the size is adapted to the level of the input signal. The resulting method is called
adaptive delta modulation (ADM).

There are several types of ADM, depending on the type of scheme used for adjusting the step
size. In this ADM, a discrete set of values is provided for the step size.
39
 A large step size was required when sampling those parts of the input waveform of steep slope. But a
large step size worsened the granularity of the sampled signal when the waveform being sampled was
changing slowly. A small step size is preferred in regions where the message has a small slope. This
suggests the need for a controllable step size - the control being sensitive to the slope of the sampled
signal.

The Implementation of ADM Modulator

The audio signal will pass through a low-pass filter, which can remove all the unwanted signal and only
obtain the audio signal. The input signals of the comparator are the audio signal and triangle wave signal, and

40
then the output of the comparator is the square wave signal. The D type flip flop is used as sampling

and then the output signal of the flip flop is the modulated ADM signal. After that the signal will
feedback to tunable gain amplifier and level adjuster. In accordance with the different between the input
signal x(t) and the reference signal X (t), we can change the magnitude of the gain of the tunable
amplifier. If the different of the input signal and the reference signal is very large, then the level adjuster
will change the gain of the t unable amplifier so that the value of Δ(t ) will become large. On the other
hand, if the different of the input signal and the reference signal is very small, then the level adjuster
will change the gain of the tunable amplifier so that the value of Δ( t ) will become small. With this
advantage, when the frequency variation of the input signal is large, then we can increase the value of
Δ(t) to prevent the occurrence slope overload. And when the frequency variation of the input signal is
small, then we can decrease the value of Δ(t) to reduce the error.

Fig. Waveforms of ADM

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COMPARISON OF PCM AND DM SYSTEMS

When the analog signal is sampled, it can be quantized and encoded by any one of the
following techniques
b. Pulse code modulation (PCM)
c. Delta Modulation (DM)
d. Differential pulse code modulation (DPCM)

a. PCM: The analog speech waveform is sampled and converted directly into a multi bit digital
code by an A/D converter. The code is stored and subsequently recalled for playback b. DM: Only
a single bit is stored for each sample. This bit 1 or 0, represents a greater than or less than
condition, respectively as compared to the previous sample. An integrator is then used on the
output to convert the stored nit stream to an analog signal.
c. DPCM: Stores a multibit difference value. A bipolar D/A converter is used for playback to
convert the successive difference values to an analog waveform.
.
These techniques convert an analog pulse to its digital equivalent. The digital information is then
transmitted over the channel. The major difference among the techniques are given below-
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Noise in PCM and DM systems

Signal to Quantization Noise ratio in PCM:

The signal to quantization noise ratio is given as:

The number of quantization value is equal to:

Putting this value in eqn (6), we get

Substitute this value in eq, we get

Let the normalized signal power is equal to P then the signal to quantization noise will be given

by

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COMPARISON OF PCM AND DM SYSTEMS
S.No Parameter Pulse Code Modulation Delta Modulation

1 Number of bits Very high, It can use 4,8 It uses one bit per
or 16 bits per sample sample

2 Quantization levels It depends on number of One bit quantizer is

bits q=2v used

3 Type of error Quantization error Slope overload

error and granular
noise

4 Signal to Noise Ratio Very high Moderate

5 Bandwidth Highest bandwidth is Lowest bandwidth

needed since the number is enough
of bits are high

6 Complexity Complex system to Simple to

implement implement
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