UNIT II

Introduction To Digital Communication

 How to convert Analog message into Digital form?
We cannot give a continuous time varying signal to an ADC. It won’t be able
to convert it. So we have to convert the signal into discrete value.

A
Digital
VA D Outputs

VA is a Discrete value and it gets converted into Digital code of 8 or 16 bits

 INDRODUCTION
• If the information is represented by a finite set of
discrete values, then the result is called as digital signal.
• Analog information such as voice of human, any picture
can be converted into digital form using three basic
steps namely: Sampling, Quantizing and Encoding.
• In sampling operation, the sample values of analog
signal are obtained at regular intervals of time
• During quantizing, each of these sample value is made
equal to a predetermined level.
• During encoding, the quantized levels are represented
by a code containing specific number of bits.
Noise in an Information Carrying Channel
• For a given transmission and coding system there exists
a threshold noise level.
• As long as the noise does not exceed it, practically no
errors occur in transmission or reception.
• When binary coding system is used, noise must
compete with the full power of the transmitter to affect
the signal. Practically no errors occur in digital
communication when signal to noise ratio is equal to
30dB.
• But if we want to transmit information at a faster rate
(i.e. increase in signaling speed), then the number of
coding levels has to be raised.
Noise in an Information Carrying Channel
• This leads to increase in the number of bits
required to code the given information.
• This further leads to increase in the
bandwidth. Every time the bandwidth doubles
the noise power also doubles.
• This also changes the capacity of the channel.
• Hence in order to keep signal to noise ratio
constant the transmitter power has to be
raised considerably, whenever the signaling
speed is increased.
 Hartley Law
• It states that C=2δflog2N where C= Channel Capacity,
N= Number of coding levels & δf = bandwidth.
• Extending the same principle we have H=Ct, where
H = total number of bits of information sent in time t
seconds.
• The Hartley law states that the bandwidth required,
for transmitting a given information, at a given rate
in proportional to the rate at which the information
is generated.
• In statement form H = Ct = 2δf t log2N. This law holds
good only in the absence of noise.
 Shannon Hartley Theorem
• The Shannon Hartley theorem gives a formula for the capacity of a
channel when its bandwidth and noise levels are known. In other
words it overcomes the drawback of Hartley law.
• This theorem can be stated as
C = δf log2 (1+S/N)
Where S/N=Signal to noise ratio at the input of the receiver,
within a bandwidth δf.
• The importance of this theorem is that:
• There is a possibility of trading bandwidth for signal to noise ratio.
• Doubling the bandwidth for a given signal to noise ratio will only
increase the channel capacity but will not exactly double the
capacity of channel.
• There is a limit on the signaling speed in a noisy channel and
exceeding the limit will lead to distortion of the signal.
Advantages of Digital Pulse Communication

• Easy to process
• Transmit fast
• Eqpt simple
 Sampling Theorem
• More number of samples within a given period of time
always represents information in a better way.
• But increased number of samples increases the number of
bits required to code them and thereby increases the
channel capacity and bandwidth requirements.
• The number of samples taken and sent per second is called
sampling frequency.
• Statement: The sampling theorem states that the
minimum sampling frequency should be at least twice the
highest modulating signal frequency, so that signal can be
reconstructed in the receiver with minimum distortion.
 Sampling Theorem
• Statement: The sampling theorem states that the
minimum sampling frequency should be at least twice the
highest modulating signal frequency, so that signal can be
reconstructed in the receiver with minimum distortion.
• Stating mathematically, fs ≥ 2fm where fs=sampling
frequency and fm= highest modulating frequency
component.
• The signal recovery at the receiver is possible, only when
the sampling frequency is more than or equal to twice the
highest modulating frequency. This sampling frequency (i.e.
fs=2fm) is also called Nyquist rate.
 Pulse Communication

• Pulse Communication or Pulse Modulation is a

system used for transmission and reception of
continuous or analog information such as
speech or Video or data. In this system the
continuous wave forms are sampled at regular
intervals.
• There are basically two categories of pulse
modulation namely Analog and Digital.
 Types of Pulse Modulation
• In analog pulse modulation the amplitude of the
sample may vary with time.
• In digital pulse modulation a code that indicates the
amplitude of the sample to a nearest
predetermined level is sent.
• Pulse Amplitude modulation (PAM) and Pulse Time
Modulation (PTM) are types of analog pulse
modulation.
• Pulse code modulation (PCM) and Delta Modulation
(DM) are types of digital pulse modulation.
• PTM is further classified into PWM and PPM.
 Analog Pulse modulation -
Pulse Amplitude Modulation (PAM)

• PAM is a system in which the modulating signal is

sampled at regular intervals and each sample is made
proportional to the amplitude of the signal at the instant
of sampling.
• These output pulses are then used to further modulate a
carrier.
• There are two types of PAM, namely Double polarity and
Single polarity.
• There is another way of classifying PAM system i.e.
Natural PAM and Flat Top PAM.
 Pulse Amplitude Modulation (PAM) - TYPES

• In double polarity PAM the samples are both positive

and negative.
• The single polarity PAM is generated by adding a
fixed DC level to the double polarity PAM signal, so
that all the pulses are either positive or negative.
• In natural PAM the amplitude of sample varies,
within the sampling duration.
• In flat top PAM, the sample amplitudes are constant
within the sampling duration.
PAM Waveforms - I

Modulating
Signal

Sampling
Signal

Double
Polarity
Natural PAM

Single Polarity
Natural PAM
PAM Waveforms - II
Modulating Signal

Sampling Signal

Double Polarity
Flat Top PAM

Single Polarity
Flat Top PAM
 Generation of PAM
+V

Modulating
Signal

PAM
Output

Clock / Sampling
Signal Figure 1
 Generation of PAM
• The circuit diagram of PAM modulator is shown in figure 1. It is a
simple emitter follower.
• The modulating signal is applied as input signal. Another input to the
base of the transmitter is the clock signal.
• The frequency of clock signal is made equal to the sampling
frequency.
• The amplitude of the clock signal is chosen such that the high level is
equal to zero volts and low level is equal to some negative value.
• Hence when the clock pulse is high the circuit behaves as an emitter
follower and the output is same as input. But when the clock signal is
low, the negative voltage drive the transmitter in to cut-off and the
output is zero.
• Thus the output waveform obtained will be the desired pulse
amplitude modulated waveform.
 Generation of PAM
• Another method of
generating PAM is shown in
figure 2.
• The modulating signal is Modulating
Input
applied as input signal.     PAM
• The clock signal is applied as a    
O/P

second input to the bridge

rectifier.
• When the clock pulse is low
Clock Input
the output follows the input
and when the clock pulse is
high the output is zero.
• Thus the output waveform is
the pulse amplitude
modulated waveform.
 Detection/Demodulation of PAM
• Demodulation of a natural PAM signal can be done
by an ideal low pass filter, with a cut off frequency at
fm.
• But for this the pulse top shape should be
maintained after transmission.
• This is very difficult due to the transmitter and
receiver noise.
• Therefore flat top PAM is preferred over natural PAM.
• There are two methods of demodulating a PAM
signal namely using an Equalizer and using holding
circuit.
Demodulation of PAM using Equalizer
 Demodulation of PAM using Equalizer
• Demodulation of flat top PAM can be done as shown
in figure.
• This circuit consists of a low pass filter and an
equalizer.
• The low pass filter is used to remove the carrier/clock
signal and pass only the modulating signal.
• Then an equalizer is used to compensate for the
aperture effect.
• This circuit containing a low pass filter and an
equalizer is known as composite filter.
 Demodulation using Holding Circuit
• Figure shows demodulation of PAM by using a holding circuit.
• The switch is closed and opened at frequency f s (sampling
frequency).
• It is opened and closed soon after the leading edge of the
sample pulse.
• During the sample interval the capacitor C charges to the
voltage proportional to the amplitude of the incoming pulse
and holds this voltage until the operation is repeated for the
next sample.
• The low pass filter following the capacitor smoothen the
output of the capacitor.
 Demodulation using Holding Circuit

PAM Holding Low Pass

I/P Circuit Filter Demodulated
O/P
 Practical Demodulator circuit
• Figure shows a practical PAM demodulator. It consists of a diode
detector and a second order low pass filter.
• In this the diode along with the R1 – C1 combination works as an
envelope detector. The PAM input is given to the envelope detector.
• The output of envelope detector is given to a second order active
low pass filter using an op-amp. The output will be the most
approximated value of the modulating signal

+
 

 
-
PAM I/P R1 C1 Demodulated
Output
 Frequency Spectrum of PAM
• Consider a complex modulating signal m(t) which contains different
frequency components.
• Let the frequencies present be from 0 to fm.
• Therefore the highest modulating frequency present is fm.
• In order to transmit this signal using PAM, we sample this signal using a
sampling signal.
• As per sampling theorem the frequency of the sampling signal should be
more than or equal to twice the highest modulating frequency. i.e. fs ≥ 2fm.
• If we consider a flat top sampled PAM wave, then the frequency spectrum
of modulating signal m(t) and the PAM output from the modulator m’(t)
are as shown in figure (a) and (b) below.
• As it can be seen, the PAM output wave consists of modulating signal and
infinite number of sidebands. The first pair of sidebands is centered at fs,
the second pair centered at 2fs and so on.
Frequency Spectrum of PAM
 Aperture Effect
• Aperture effect is defined as attenuation of high frequency
components of the modulating signal at the receiver due to
the characteristics of the filter used.
• At the receiver, the PAM signal is demodulated by passing it
through a Low pass filter having a cutoff frequency equal to f m.
• In a practical Low pass filter, the characteristics of the filter will
be such that the lower frequency components are passed and
the high frequency components are attenuated. Due to this
some of the high frequency components of the modulating
signal will be attenuated.
• The input and output of an ideal and a practical low pass filter
are shown in figure below.
PAM Demodulator Output
 Aperture Effect
• As seen from above figure, the low pass filter at the receiver is not only
cutting the high frequency sideband components but it is also
attenuating the higher frequency components of the modulating signal
itself. This is called as aperture effect and if not treated will lead to
errors in reception.
• In order to avoid Aperture effect, the LPF used should have a cutoff
frequency more than fm.
• In such a case if the cutoff frequency is very high then along with
modulating signal some part of the sideband component also may get
passed to the output. Hence care should be taken while designing the
LPF such that its cutoff frequency is only slightly more than f m.
• The equalizer circuit used in the receiver increases the level of high
frequency components from the output of LPF. Hence we can avoid
aperture effect by using equalizer circuit.
 Pulse Time Modulation (PTM)
• Pulse time modulation is a method in which
either the duration of the sampling signal or
the position of the sampling signal is varied in
accordance with the modulating signal.
• There are of two types as follows.
• Pulse Width Modulation (PWM)
• Pulse Position Modulation (PPM)
 Pulse Width/Duration Modulation
• If the duration of a pulse is varied in accordance with the
modulating signal, then it is called as pulse duration or
pulse width modulation.
• During Pulse Width Modulation, the amplitude of the
pulse is fixed.
• While doing pulse width modulation the starting time can
be fixed for each pulse, but the width of each pulse is
made proportional to the instantaneous amplitude of
modulating signal.
• Similarly, we can also fix the trailing edge of the pulse and
the width can be varied or the pulse can be fixed at the
center and its width can be varied.
 Indirect Method of PWM Generation

PAM o/p

Modulating Sampling PWM

Signal m(t) Adder Comparator
Circuit Output

Sampling Pulses
Ramp
s(t) Slicing Level
Generator
 Indirect Method of PWM Generation
• The block diagram for generating PWM wave using indirect
method is shown in the figure.
• In this method the modulating signal is first sampled to obtain
flat-top PAM signal using a Pulse Amplitude Modulator.
• This PAM signal is then added to the output of a ramp
generator using an Adder.
• The type of ramp waveform decides the type of PWM.
• If the ramp is negative then we get a PWM with starting edge
fixed.
• If the ramp is positive then we get a PWM with trailing edge
fixed.
• If the ramp is a triangular waveform then we get a center
fixed PWM wave.
 Indirect Method of PWM Generation
• The output of adder is given as one of the input to a
comparator.
• The second input to the comparator is a DC reference
level.
• For this method to give satisfactory result, the
amplitude of ramp voltage must be slightly greater
than maximum variation in the amplitude of PAM
signals.
• The DC reference level of the comparator should
intersect the sloping portion of the adder waveform.
• The comparator output will be the desired PWM
waveform.
 PWM Generation waveforms I

Single Polarity Flat Top PAM

obtained from modulating signal

Ramp Waveform

Comparator Reference level

Adder output

PWM Wave output

From Comparator
Starting edge Fixed
 PCM Transmitter
A to D Converter

Quantized
Signal
Analog Signal Output to
m(t) mq(t) Channel
Sampler Quantizer Encoder
 PCM Transmitter - Sampler
• The sampler block at the transmitter is similar
to PAM sampling circuit.
• It generates Flat Top Pulses whose amplitude
is proportional to the amplitude of the
modulating signal at the instant of sampling.
• The sampling rate should be greater than the
Nyquist rate i.e. fs ≥2fm.
 PCM Transmitter - Quantizer
• The operation of quantization can be
explained as below…
1. Consider a modulating signal m(t), having
maximum and minimum values in the range of
VH & VL.
2. This total range is divided into M number of
equal intervals each having a step size ‘S’, where
S = (VH – VL)/M.
3. The center of each of these steps is called as
quantization levels m0, m1, m2,…etc.
 PCM Transmitter - Quantizer
4. Whenever m(t) is in the range Δ0, the quantized
signal mq(t) will be maintained constant at level
m0.
5. Similarly whenever m(t)is in the range Δ1, the
quantized signal mq(t) will be constant at level
m1, and so on.
6. When m(t) crosses one range then mq(t) takes an
abrupt or “quantum” jump of step size S.
7. At every instant of time the difference between
the actual signal m(t) & the quantized signal
mq(t) is equal to S/2.
 Quantization

VH
m4
Δ4

m3 m(t)
Δ3
mq(t)

m2
Δ2
S = (VH – VL)/m

m1
Δ1

m0
Δ0
VL
 PCM Transmitter - Encoder
• After sampling & quantizing the signal arrives at the
Encoder.
• The function of encoder is to generate a specific binary
sequence for each quantized level.
• The advantage of using encoder is that it allows only
two voltage levels. That is +V volts and 0 Volts or –V
volts and +V volts. In this way the transmitter output
power is very high positive or very high negative.
• Now the noise in the channel has to compete with the
full power of the transmitter and hence problems of
noise are reduced.
 PCM Transmitter the entire Process
(5)VH
m4
4.5
Δ4
m3 m(t)
3.5
Δ3
m2
2.5
Δ2
m1
1.5
Δ1
0,5 m0
Δ0
(0)VL

Sampler o/p 1.2 V 0.6 V 3.3 V 4.4 V

Nr. Quanti
-zation level 1.5 V 0.5 V 3.5 V 4.5 V
Coding Level Δ1 Δ0 Δ3 Δ4
Binary signal 001 000 011 100
 Repeater
• The advantage of digital communication is that we can make
use of repeaters in the communication channel between the
transmitter & receiver.
• These repeaters are placed at such intervals, so that the
noise voltage level is always less than S/2.
• The signal mq(t) from the transmitter which has got added
noise voltages at each level will arrive at the repeater.
• The repeater block consists of a quantizer & amplifier. The
quantizer at the repeater re-quantizes the signals to the
same quantization level, thereby removing all the noise.
• Only when the noise voltage added to the signal is greater
than S/2 as shown in figure below, there will be an error in
level. The same principle works at the receiver also & hence
the first block of the receiver is quantizer.
 Working of Repeater – Re-Quantization
VH

m4 R
Noise quantized
Noise level E
m3 exceeds S/2
mq(t) P
m'q(t)
E
m2
A

m1 T

m0 R
VL
 Quantization Error
• Even though quantization at the transmitter & the receiver
and at the repeaters helps us in reducing the problems of
noise, the quantization by itself introduces an error known as
quantization error.
• It is because at no point of time, m(t) and m q(t) will have the
same magnitude.
• This difference between modulating signal m(t) & quantized
signal mq(t) is called as Quantization Error.
• This error can be reduced by reducing the step size or in other
words by increasing the number of quantization levels.
• It is also called as Quantization Noise. Its maximum value is
equal to S/2 and mean square value is S2/12.
 PCM Receiver

D to A Converter
Input from Analog Output
Comm. Channel m’(t)

Quantizer Decoder Filter

 PCM Receiver
• Quantizer
• The function of quantizer at the receiver is like that of a repeater. It re-
quantizes the signal so that any noise that was added during the
transmission of signal is removed.

• Decoder
• The function of decoder at the receiver is opposite to that of encoder. The
decoded output is a sequence of quantized levels. The decoder is also
known as D to A converter.

• Filter
• The output of the decoder contains quantized sample values. The output of
the sampler at the transmitter consists of the base band signal m(t) and
higher harmonics of fs (where fs is the sampling frequency). The filter
removes these harmonics & recovers the signal m’(t) which is identical to
m(t) except for the quantization noise.
 Companding
• The quantization of signals gives noise free reception provided that
the original signal lies in the range V H-VL. Consider two signals m1 (t)
and m2(t) as shown below.
• The signal m1(t) has maximum & minimum values beyond V H and VL
respectively. But this extra voltage will not be considered while
quantizing & hence an error occurs. The signal m 2(t) has variations
within a single quantization level. Hence it will be continuously
quantized to same level (Δ3) and at the receiver will yield a DC
output, which is again a serious error.
• Hence in order to overcome such problems, a process called as
Companding is used at both the transmitter & receiver.
• If the signal levels are beyond VH-VL then the signal is compressed
to fit into the range. If the signal levels are too low then the signal
is expanded so as to cross at least one quantization level. This
process is called as Companding.
 Compander

Voltage above VH
VH
m4
Δ4

m3
Δ3
m2(t)

m2
Δ2

m1 m1(t)
Δ1

Δ0 m0

VL
Voltage below VL
 Companding

Vomax
Transmitter No Companding

Vimin

Receiver

Vomin
 Disadvantages of PCM
• The number of bits required to code a quantization level
depends on the number of quantization levels.
• If we increase the number of quantization levels thereby
reducing the step size it leads to reduction in
quantization error.
• But this increases the number of bits required to encode
the information thereby increasing the bandwidth.
• When we increase the bandwidth for signal transmission,
the noise power also increases because P n i.e. noise
power is directly proportional to the bandwidth.
• Hence in practice, even though PCM yields a noise free
reception it is further modified before usage.
 Disadvantages of PCM
• In a PCM system, the sampled values are
quantized and transmitted.
• In order to yield less quantization error, in a
practical system we use 256 levels.
• Hence the number of bits required for
encoding will be (28 = 256) 8.
• This increases the band width requirement of
a PCM system.
 Differential PCM (DPCM)
• In DPCM system, instead of quantizing the samples
directly, we quantize the difference between two
successive samples.
• In other words rather than quantizing m(k-1) and
m(k) samples at (k-1)th and Kth time interval, we
quantize the difference between m(k-1) and m(k).
• The advantage of this scheme is that we will now
have sample values whose range is very less.
• Hence we require less number of levels for
quantizing. This leads to reduction in bandwidth.
 Advantage of DPCM over PCM
• For example: We have sampled a signal m(t), which is in the
range of 0 to 5 volts, and Let us say the sample values are S1,
S2, S3, S4 & S5 as in column 1 below.
• In order to quantize these values we divide the voltage range
into 5 steps and fix the quantization levels at 0.5V(m0),
1.5V(m1), 2.5V(m2), 3.5V(m3) & 4.5V (m4).
• Hence sample S1 is quantized to m1 and so on.
Sample Quantization
Value Level
S1 = 1.3V m1
S2 = 2.2V m2
S3 = 4.6V m4
S4 = 3.7V m3
S5 = 1.0V m0
 Advantage of DPCM over PCM
• But if we take difference between S2 and S1 then it is equal to
0.9V. Similarly we will have difference signals as 0.9V, 2.4 V,
-0.9V, 2.7V.
• Now the total voltage range is from -0.9 to +2.7 which is
significantly lower than the 0 to 5 V range. Hence with 5
quantization levels, the quantization error can be reduced.
• Alternatively we can also think of having less number of
quantization levels, which save on the bandwidth
 DPCM
Comparator
(Difference Generator)

m(t) Δ(t) Sample & Δ(k) Δq(k)

Hold Circuit Quantizer
m’(t)

Accumulator Predictor

Duplicate/ Dummy receiver

 Difficulties of DPCM
• If the difference signal is greater than VH or VL then
it cannot be accommodated due to reduced
number of levels.
• Suppose m(t) is changing very slowly with respect
to time, then the difference between m(t1) and
m(t2), m(t2) & m(t3) … etc may be very small and
the output of quantizer may yield a DC level.
• Suppose during a particular time interval m(t)
decreases suddenly and increases again then
these changes may not be possible to
accommodate.
 DPCM
• In this process, instead of quantizing the difference directly,
the accumulator accumulates these differences and
generates an approximate wave form m’(t).
• Then this signal m’(t) is compared with m(t) and their
difference is transmitted.
• In order to generate m’(t) we make use of a Predictor whose
function is to logically predict whether m(t) will be rising or
falling and accordingly adjust the accumulator output.
• To get the same quality of output at the receiver, the DPCM
has much more complicated circuitry than a simple PCM and
the advantage of band width is not up to the mark.
 Delta Modulation (DM)
• PCM : As No. of steps Increases
= Bits required for encoding Increases
= Bandwidth required for communication Increases
= Noise power Increases
• DPCM : Circuitry is complicated.
• Delta modulation is a type of DPCM scheme in
which the difference signal Δ(t) is encoded into
just a single bit. This single bit indicates whether
to increase or decrease the estimated value of
m’(t).
 DM Transmitter
Pulse
Generator

Δ(t) = +V; m(t)>m’(t)

m(t) Δ(t) = -V; m(t)<m’(t)
Comparator Modulator

m’(t)

Integrator
 DM Transmitter
• Comparator: It has two inputs
– The modulating signal m(t) is applied to the non-inverting
input of the comparator.
– A feedback signal m’(t) is applied to the inverting input of
the comparator.
The comparator output
Δ(t) = Positive (i.e. HIGH) if m(t) > m’(t)
= Negative (i.e. LOW) if m(t) < m’(t).
Therefore the polarity of the comparator output depends
on the polarity of the difference signal.
• Pulse Generator: This block generates periodic train of
pulses Pi(t) with a fixed frequency fs.
 DM Transmitter
• Modulator: The comparator output is applied to
the modulator along with Pulse generator
output.
The output of modulator is train of pulses P o(t),
whose polarity depends on the polarity of
difference signal Δ(t).
– Po(t) = Positive pulse (i. e. logic 1) if Δ(t) is
positive
– Po(t) = Negative pulse (i.e. logic 0) if Δ(t) is
negative
 DM Transmitter
• Integrator: The output of modulator, which is either a positive
or negative polarity pulse is applied to the integrator.
It generates an output m’(t), which raises or falls by a fixed
step height depending on the polarity of the input pulse.
– Hence the output of Integrator is a staircase waveform. If
the step size of the staircase is small, then m’(t) will be an
approximation to the signal m(t).
– The step size can be adjusted by changing the gain of the
integrator.
– The signal Po(t), which gives information about the
difference signal Δ(t) is transmitted. It is a single bit having
a value of either 1 or 0.
 DM Receiver

From Channel
1/0
Quantizer Integrator LPF
m(t)
 DM Receiver
• At the receiver the signal is received along with
noise. It is sent through a quantizer, which removes
most of the noise.
• The output of quantizer will be same as
transmitted form of Po(t). This is fed to the
integrator, which produces a staircase waveform
similar to m(t). The output of integrator is
smoothed by the low pass filter to give a waveform
similar to the original modulating signal m(t).
 DM Transmitter
2nd Method

Δ(t) = +V; m(t)>m’(t)

m(t) Δ(t) = -V; m(t)>m’(t)
Sample & So(t)
Comparator
Hold Circuit
m’(t)

D to A Up – Down
Convertor Counter Count Direction
Command
Clock
 DM Transmitter -II Method
• At the power ON m’(t) is zero and m(t) may be some positive value.
• Accordingly the output of the comparator is V(H) or V(L).
• This signal is sampled at regular intervals using Sample & Hold (S&H)
circuit.
• The So(t) is same as Po(t) as shown in previous diagram. The
positive or negative output is given as count direction command to
the Up-Down counter. The counter output will either increase or
decrease with the sampling interval. The D to A converter gives an
analog equivalent of digital input i.e. m’(t) which is given as input to
the comparator.
• In this block diagram S & H circuit does the function of modulator,
Up-Down counter along with D to A converter gives equivalent of
integrator of the previous diagram for DM Transmitter.
 Drawbacks of DM – a) Slope Overload
• The waveform m’(t) needs to closely follow the
waveform m(t). Only then the recovered signal will be
similar to m(t).
• But due to the fixed step size in m’(t) an error known
as Slope Overload Error can occur.
• The slope overload occurs if the change in
modulating signal between two sampling intervals is
greater than the step size of the integrator.
• In DM, the integrator will not be able to follow any
signal, with a very fast rate of, rise or fall.
• Since this error is dependent on the slope of the
modulating signal, it is known as Slope Overload error
 Slope Overload
• There are two types of Slope Overload.
• Let us say the signal m(t) rises faster than the
step waveform m’(t) during a signaling interval
P to Q. Therefore m’(t) rises constantly and it
catches up with m(t), at point Q. Thus an
error will be caused in the demodulated signal
for portion PQ. This is called the positive slope
overload error.
• If the slope of m(t) is more negative than the
slope of m’(t) for portion PQ, then it is known
as negative slope overload
 Slope Overload
Positive Slope Negative Slope
Overload Overload
Q
P

m(t) m’(t)
m’(t) m(t)

P
Q
 Drawbacks of DM – b)Granular Noise
• It is the second limitation of Delta Modulation.
• If the variation in m(t) are such that they are
within the step size then the signal m’(t) is a
square wave.
• It alternates above and below m(t) or “hunts”
around m(t).
• This when demodulated at the receiver yields a DC
signal whereas the original signal m(t) is not DC.
• Hence in this case too, there is distortion and the
noise is known as Granular Noise.
 Granular Noise

m(t) m’(t)

Δ
 Drawbacks of Delta Modulation
• Slope Overload – can be taken care by increasing step
size of integrator output
• Granular noise - can be taken care by reducing the step
size.
• A small step size can lead to slope overload more easily
and a large step size can lead to Granular noise.
• Slope Overload and granular noise, both can be
reduced by improving the sampling rate, well above
the NYQUIST rate (i.e. fs should be very much greater
than 2fm). But this leads to increased bandwidth.
 ADM Transmitter
Pulse
Generator

Δ(t) = +V; m(t)>m’(t)

m(t) Δ(t) = -V; m(t)<m’(t)
Comparator Modulator
m’(t)
Variable Gain
Amplifier
Integrator
Gain Control I/P
Square Law
Device
 Adoptive Delta Modulation (ADM)
• Slope Overload error occurs, when the slope
of the signal m(t) is too high, while granular
noise occurs when the changes in the signal
m(t) are less than the step size.
• Both these limitations of DM can be overcome
in Adoptive Delta Modulation, where the step
size is adjusted in accordance with the signal.
• Thus when the signal changes are small the
step size is reduced and it is increased when
the signal changes are large.
 ADM Transmitter
m’(t)

Increased Step Size

m(t)

Decreased Step Size

 ADM Transmitter
• ADM is a type of DM in which both the limitations of DM
can be overcome, by adjusting the step size in accordance
with the signal being transmitted.
• Thus when the signal changes are small, the step size is
reduced, while to avoid slope overload, the step size is
increased.
• In the block diagram of ADM transmitter, a variable gain
amplifier is used.
• One input to variable gain amplifier is from the modulator
i.e. Po(t).
• The second input to variable gain amplifier is called gain
control input and it is obtained by integrating Po(t) using an
RC network and then passing it through a square law device.
 ADM Transmitter
• Under the Slope overload condition, Po(t) is a long
sequence of positive or negative pulses.
• Hence RC network output is a large positive or
negative voltage.
• The square law device will always produce a positive
voltage irrespective of whether its input is positive or
negative.
• Thus under Slope Overload conditions, gain of the
amplifier increases, thus resulting in an increased
step size.
• The output of variable gain amplifier goes to the
integrator which generates the step waveform.
 ADM Receiver

From Channel Variable Gain

Quantizer Integrator LPF
Po(t) Amplifier m(t)
Gain Control I/P

Square Law
Device
 ADM Receiver
• On the receiver side, the quantizer eliminates the channel noise.
• The output of the quantizer is fed to a variable gain amplifier
whose gain control input is derived from an RC integrator and a
square law device. Thus the adaptive adjustment of step size is
obtained at the receiver,
• The output of variable gain amplifier is fed to integrator.
• The integrator generates a staircase waveform having variable
step size.
• This staircase when passed through a LPF leads to undistorted
reception of the original modulating signal.
• Since the slope of the signal can be continuously varied, ADM
method is also known as CVSDM (Continuously Variable Slope
Delta Modulation.)