UNIT IV PULSE MODULATION PART 1

Syllabus: Pulse Modulation: Types of Pulse modulation- PAM, PWM and

PPM. Comparison of FDM and TDM.

Fig. : Carrier for Continuous Wave and Pulse Modulation

Need of Pulse Modulation:

• Comparing to continuous wave modulation (like AM, FM), the
performance of all pulse modulation schemes except PAM in presence
of noise is very good.
• Due to better noise performance, it requires less power to cover large
area of communication.
• Due to better noise performance and requirement of less signal
power, the pulse modulation is most preferred for the communication
between space ships and earth.

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Sampling :
The signals we use in the real world, such as our voices, are called "analog“
signals. To process these signals in computers, we need to convert the
signals to "digital" form. While an analog signal is continuous in both time
and amplitude, a digital signal is discrete in both time and amplitude.
• To convert a signal from continuous time to discrete time, a
process called sampling is used. The value of the signal is
measured at certain intervals in time. Each measurement is referred
to as a sample.
• How many samples are necessary to ensure we are preserving the
information contained in the signal?
• If the signal contains high frequency components, we will need to
sample at a higher rate to avoid losing information that is in the
signal.

Ex(2):

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 In general, to preserve the full information in the signal, it is
necessary to sample at twice the maximum frequency of the signal. This is
known as the Nyquist rate. The Nyquist rate or frequency is the
minimum rate at which a finite bandwidth signal needs to be sampled to
retain all of the information.
Sampling theorem gives the criteria for minimum number of samples that
must be taken.
The Sampling Theorem states that a signal can be exactly reproduced if
it is sampled at a frequency F, where F is greater than twice the
maximum frequency in the signal. I,e Sampling frequency must be
twice the highest freq. fs=2fm Or
The original information signal can be reconstructed at the receiver
with minimal distortion if the sampling rate in the pulse modulation
system equal to or greater than twice the maximum information signal
frequency. fs >= 2fm (max)
 infinite bandwidth cannot be sampled.
 the sampling rate must be at least 2 times the highest frequency,
not the bandwidth.

If the required condition of the sampling theorem that fs>= 2f m maximum is

not met, then errors will occur in the reconstruction.
Aliasing occurs because signal frequencies can overlap if the
sampling frequency is too low. Frequencies "fold" around half the sampling
frequency - which is why this frequency is often referred to as the folding
frequency.

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 Sometimes the highest frequency components of a signal are simply
noise, or do not contain useful information. To prevent aliasing of these
frequencies, we can filter out these components before sampling the signal.
Because we are filtering out high frequency components and letting lower
frequency components through, this is known as low-pass filtering.
Aliasing effect:

Problem (1):
A complex low-pass signal has a bandwidth of 200 kHz. What is the
minimum sampling rate for this signal?
Solution: The bandwidth of a low-pass signal is between 0 and f, where f is
the maximum frequency in the signal. Therefore, we can sample this signal
at 2 times the highest frequency (200 kHz). The sampling rate is therefore
400,000 samples per second.
Problem (2):
A complex bandpass signal has a bandwidth of 200 kHz. What is the
minimum sampling rate for this signal?
Solution :
We cannot find the minimum sampling rate in this case because we do not
know where the bandwidth starts or ends. We do not know the maximum
frequency in the signal.

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Types of Sampling Methods:

5
Pulse Amplitude Modulation (PAM):
Definition: A modulation technique in which the amplitude of the pulsed
carrier signal is changed according to the amplitude of the message signal is
known as Pulse Amplitude Modulation (PAM) where width and position
remains constant.
(OR)
The amplitude of the pulsed carrier varies in accordance with the
instantaneous value of modulating signal, is called PAM.
Simply put, the transmission of data takes place by the variation of the
amplitude of pulse according to the modulating signal.
Block diagram of PAM

The figure below shows the block diagram of a PAM generator

FIGURE 1

Above figure consist of a low pass filter, a modulator along with a train
generator and a pulse reshaping circuit. The modulating signal is given to
the low pass filter in order to band limit the message signal.
The LPF at the beginning is placed in order to avoid aliasing of the
samples. The LPF passes only the low-frequency component of the signal
and eliminates the high-frequency signal component. The output of LPF is
then provided to a modulator, where it gets mixed with the rectangular
pulse train.

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 Basically, the pulsed carrier gets modulated by the message signal
here. The rectangular carrier pulse is generated by the pulse generator
circuit.
The modulator generates a pulse amplitude modulated signal. The
sampled pulses can be achieved either by natural or flat top sampling. The
output of the modulator is provided to the pulse reshaping circuit. This
basically shapes the pulses so that it can be easily detected at the receiver.
Why flat top sampling is preferred over natural sampling?
Regeneration of flat-top pulses by the repeater is somewhat easier in case of
long distance signal transmission.
Waveforms of PAM:

m(t)

c(t)

p(t)

t
p(t)

7
 Example (2)

Here as we can see the first image represents the analog message signal and
the next one shows the pulsed carrier wave that is to be modulated. The last
figure shows the PAM signal generated by the circuit.
Transmission bandwidth of PAM:
The bandwidth for the transmission of the PAM signal is greater than
the maximum frequency component of the message signal.
Demodulation of PAM:
For pulse amplitude modulated (PAM) signals, the demodulation is
done using a Holding circuit. below figure.1 shows the block diagram of a
PAM demodulator.

Figure 1

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In this method, the received PAM signal is allowed to pass through a Holding
circuit and a low pass filter (LPF) as shown in above figure.
The Sample and Hold circuit is an electronic circuit which creates
the samples of voltage given to it as input, and after that, it holds these
samples for the definite time. The time during which sample and hold circuit
generates the sample of the input signal is called sampling time. Similarly,
the time duration of the circuit during which it holds the sampled value is
called holding time. Now, fig.2 illustrates a very simple holding circuit.

Fig.2 : A zero-order holding circuit

Here the switch ‘S’ is closed after the arrival of the pulse and it is opened at
the end of the pulse. In this way, the capacitor C is charged to the pulse
amplitude value and it holds this value during the interval between the two
pulses. Hence, the sampled values are held as shown in fig.3.

Fig.3: the output of a Low Pass filter (LPF)

After this the holding circuit output is smoothened in Low Pass filter as
shown in fig.3. It may be observed that some kind of distortion is introduced
due to the holding circuit. In fact the circuit of fig.4 is known as zero-order
Holding circuit. This zero-order Holding circuit considers only the previous
sample to decide the value between the two pulses.

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 Fig.4: Output of holding circuit

Advantages of Pulse Amplitude Modulation:

 PAM is the simplest form of pulse modulation.
 Its implementation is quite easy.
Disadvantages of Pulse Amplitude Modulation
 The transmission bandwidth required is very large.
 Due to the variation in amplitude, the power required by the
generating unit also varies.
 Since PAM does not utilize constant amplitude pulses, output is
distorted due to additive noise so that it is infrequently used.
Applications of Pulse Amplitude Modulation
It is used in LED lighting, in microcontrollers in order to produce control
signals and in the Ethernet communication system.
Pulse Width Modulation (PWM)
When the width of pulsed carrier varies in accordance with the
instantaneous amplitude of modulating signal, is called PWM, where
amplitude and frequency remains constant. (OR)
A modulation technique where the width of the pulses of the pulsed
carrier wave is changed according to the modulating signal is known
as Pulse Width Modulation (PWM). It is also known as Pulse duration
modulation (PDM).
Basics of Pulse Width Modulation:

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It is a type of Pulse Time Modulation (PTM) technique where the timing of
the carrier pulse is varied according to the modulating signal.
Waveforms of PWM:

In pulse duration modulation (PDM), the amplitude and the frequency

of the PWM wave remain constant. Only the width changes, that is why the
information is contained in the width variation. This is similar to FM.
As the noise is normally additive noise, it changes the amplitude of
the PWM signal. At the receiver, it is possible to remove these unwanted
amplitude variations very easily by means of a limiter circuits.
As the information is contained in the width variation, it is unaffected
by the amplitude variations introduced by the noise. Thus, the PWM system
is more immune to noise than the PAM signal.
Generation of PWM signal:
The block diagram of a PWM signal generator is shown in fig.5 below. This
circuit can also be used for the generation of PPM signal.

Figure 5

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 A sawtooth generator generates a sawtooth signal of frequency fs, and
this sawtooth signal in this case is used as a sampling signal. It operates
at carrier frequency and is applied to the inverting terminal of a
comparator.
 The modulating signal x (t) is applied to the non-inverting terminal of
the same comparator.
 The Comparator compares the two signals and generates a PWM
signal as its output waveform. This gives rise to a PWM signal at the
comparator output as shown in fig.6

If the value of the Sawtooth triangle signal is more than the

modulation signal x(t) then the PWM output signal is at “High” else
it’s in “Low” state. Thus, the value of the input signal magnitude
determines the comparator output which defines the width of the pulse
generated at the output.
Block Diagram of PWM Demodulation

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 • The PWM signal received at the input of the detection circuit is
contaminated with noise. This signal is applied to pulse generator
circuit which regenerates the PWM signal. Thus, some of the noise is
removed and the pulses are squared up.
• The regenerated pulses are applied to a reference pulse generator. It
produces a train of constant amplitude, constant width pulses. These
pulses are synchronized to the leading edges of the regenerated PWM
pulses but delayed by a fixed interval.
• The regenerated PWM pulses are also applied to a ramp generator. At
the output of it, we get a constant slope ramp for the duration of the
pulse. The height of the ramp is thus proportional to the width of the
PWM pulses.
• At the end of the pulse, a sample and hold amplifier retains the final
ramp voltage until it is reset at the end of the pulse.
• The constant amplitude pulses at the output of reference pulse
generator are then added to the ramp signal.
• The output of the adder is then clipped off at a thereshold level to
generate a PAM signal at the output of the clipper.
• A low pass filter is used to recover the original modulating signal back
from the PAM signal. The waveforms for this circuit have been shown
in above fig.
Waveform Representation of PWM Signal Detection:

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Advantages of Pulse Width Modulation (PWM):
1. PWM technique helps in preventing overheating of LED’s while
maintaining its brightness.
2. Pulse Width Modulation provides accuracy and quick response time.
3. It provides high input Power Factor.
4. Initial cost is low.
5. PWM technique helps the motors to generate maximum torque even
when they are running at lower speeds.
Disadvantages of Pulse Width Modulation (PWM):
The disadvantages are:
As the PWM frequency is high, switching losses is considerably high.
It induces Radio Frequency Interference (RFI).
Applications of Pulse Width Modulation (PWM):
 PWM Techniques are used in Telecommunications for encoding
purposes.
 Pulse Width Modulation helps in voltage regulation and thus finds its
use in controlling Brightness in Smart Lighting Systems and also
controls the speed of motors.
 Computer Motherboard requires PWM Signals that controls the heat
generated in the board.
 It is also used in Audio/Video Amplifiers.
Pulse Position Modulation (PPM):
Definition: A modulation technique that allows variation in the position
of the pulses according to the amplitude of the sampled modulating signal
is known as Pulse Position Modulation (PPM). It is another type of PTM,
where the amplitude and width of the pulses are kept constant and only the
position of the pulses is varied.
Simply put, the pulse displacement is directly proportional to the
sampled value of the message signal.
To understand the generation of the PPM signal, it is necessary to
understand Pulse Width Modulation (PWM or PDM). PWM is the first type of
PTM.

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Basics of Pulse Position Modulation
The basic idea about the generation of a PPM waveform is that here, as the
amplitude of the message signal increases, the pulse shifts according to the
reference.
In PWM, is that due to the variable width of the pulses, the transmission
power also varies accordingly. However, this is not the case with PPM as
here the width of the pulses remains constant and only their position varies.
Thus, transmission power does not show variation.
How the position of the pulses show variation?
A PPM signal is generated in reference to a PWM signal. Thus, the trailing
edge of the PWM signal acts as the beginning point of the pulses of PPM
signal.
The figure below shows the waveform representation of the PPM signal:

Here, the first image shows the modulating signal, and the second one
shows a carrier signal. The next one shows a PWM signal which is
considered as reference for the generation of PPM signal shown in the last
image.
In the above figure that the point of ending the PWM pulse and the
beginning of PPM pulse is coinciding, which can be clearly seen from the
dotted line.

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Generation of PPM Signal
The PPM signal can be generated from PWM signal as shown in below fig.2
(a).

Fig.2 : Generation of PPM signal

The PWM pulses obtained at the comparator output are applied to a
monostable multivibrator. The monostable is negative edge triggered.
Hence, corresponding to each trailing edge of PWM signal, the monostable
output goes high. It remains high for a fixed time decided by its own RC
components.
Thus, as the trailing edges of the PWM signal keep shifting in proportion
with the modulating signal x(t), the PPM pulses also keep shifting, as shown
in fig.3.

Fig.3
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Detection (Demodulation) of PPM signal:
The figure below shows the block diagram for the detection of a PPM signal
at the receiver:

In the above figure that the demodulation circuit consists of a pulse

generator, SR flip-flop, reference pulse generator and a PWM demodulator.
The PPM signal transmitted from the modulation circuit gets distorted
by the noise during transmission. This distorted PPM signal reaches the
demodulator circuit. The pulse generator employed in the circuit generates a
pulsed waveform. This waveform is of fixed duration which is fed to the reset
pin (R) of the SR flip-flop.
The reference pulse generator generates, reference pulse of a fixed
period when transmitted PPM signal is applied to it. This reference pulse is
used to set the flip-flop.
These set and reset signals generate a PWM signal at the output of
the flip-flop. This PWM signal is then further processed in order to provide
the original message signal.
Advantages of Pulse Position Modulation:
1. Similar to PWM, PPM also shows better noise immunity as compared
to PAM. This is so because information content is present in the
position of the pulses rather than amplitude.
2. As the amplitude and width of the pulses remain constant. Thus the
transmission power also remains constant and does not show
variation.
3. Recovering a PPM signal from distorted PPM is quite easy.
4. Interference due to noise in more minimal than PAM and PWM.
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 Disadvantages of Pulse Position Modulation
1. In order to have proper detection of the signal at the receiver,
transmitter and receiver must be in synchronization.
2. The bandwidth requirement is large.

Applications of Pulse Position Modulation

The technique is used in an optical communication system, in radio control
and in military applications.
Comparison between PAM, PWM, and PPM:

The following table presents the comparison between three modulation

techniques.
PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends on Bandwidth depends on the Bandwidth depends on the

the width of the pulse rise time of the pulse rise time of the pulse

Instantaneous Instantaneous transmitter

Instantaneous transmitter
transmitter power varies power varies with the
power remains constant with
with the amplitude of the amplitude and the width
the width of the pulses
pulses of the pulses

System complexity is
System complexity is low System complexity is low
high

Noise interference is high Noise interference is low Noise interference is low

It is similar to amplitude It is similar to frequency It is similar to phase

modulation modulation modulation

Output Waveform of PAM Output Waveform of PWM

Output Waveform of PPM

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Multiplexing – Definition – Types of Multiplexing
Generally, a communication channel such as an optical fiber or
coaxial cable can carry only one signal at any moment in time. This results
in wastage of bandwidth. However, we can overcome this drawback by using
a technique called multiplexing. By using the multiplexing technique, we
can easily send multiple signals simultaneously over a communication
channel (medium).
Multiplexing Definition
Multiplexing is a technique which combines multiple signals into one signal,
suitable for transmission over a communication channel such as coaxial
cable or optical fiber. Multiplexing is also sometimes referred to as muxing.
The multiplexing technique divides the communication channel into
several logical sub-channels. Each logical sub-channel is dedicated to an
individual signal.
Thus, the multiple signals are sent simultaneously over a shared
communication channel (medium). Multiplexing has-been used for many
years in long-distance telephony.
Multiplexing is done by using a device called Multiplexer or MUX. The
multiplexer combines n input lines to generate one output line.
Without Multiplexing vs With Multiplexing
The below figure shows the communication system without multiplexing.

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 The communication system without multiplexing carries only one
signal at any moment in time. Thus, it uses three communication channels
to carry three signals. In this technique, a large amount of bandwidth is
wasted.
The below figure shows the communication system with multiplexing. It
carries three signals simultaneously. Thus, it uses only one communication
channel to carry 3 signals (multiple signals). In this technique, the
bandwidth is effectively used.

Advantages of multiplexing
1. Multiple signals can be sent simultaneously over a single communication
channel.
2. Effective use of channel bandwidth
3. Multiplexing reduces cost
4. Multiplexing reduces circuit complexity
Applications of Multiplexing
1. Communication system
2. Computer memory
3. Telephone systems
4. TV broadcasting
5. Telemetry
6. Satellites

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 Types of Multiplexing
Multiplexing is mainly classified into two types:
o Analog multiplexing
o Digital multiplexing

Analog multiplexing is again classified into two types:

o Frequency Division Multiplexing
o Wavelength Division Multiplexing
In digital multiplexing, the Time Division Multiplexing is the most popular
technique. The time division multiplexing is again classified into two types:
o Synchronous TDM
o Asynchronous TDM
Analog Multiplexing
The process of combining multiple analog signals into one signal is called
analog multiplexing. It multiplexes the analog signals according to their
frequency or wavelength.
Multiplexing requires that the multiple signals be kept apart so that they do
not overlap with each other and thus can be separated at the receiving end.
This can be achieved by separating the signal in frequency.
There are two types of analog multiplexing:
o Frequency division multiplexing
o Wavelength division multiplexing
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Frequency Division Multiplexing (FDM):
Frequency division multiplexing is an analog technique. It is the most
popular multiplexing technique. We use this technique extensively in TV and
radio transmission. This technique combines multiple signals into one
signal and transmitted over the communication channel. Frequency division
multiplexing is also known as FDM.
In this technique, the bandwidth of the communication channel
should be greater than the combined bandwidth of individual signals.

The frequency division multiplexing divides the bandwidth of a

channel into several logical sub-channels (individual signal frequencies (or)
set of frequency bands).
Each logical sub-channel is allotted for a different signal frequency.
The individual signals are filtered and then modulated (frequency is shifted),
in order to fit exactly into logical sub-channels.
In this technique, each logical sub-channel (individual signal
frequency) is allotted to each user. In other words, each user owns a sub-
channel.

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 Each logical sub-channel is separated by an unused bandwidth called
Guard Band to prevent overlapping of signals. In other words, there exists a
frequency gap between two adjacent signals to prevent signal overlapping. A
guard band is a narrow frequency range that separates two signal
frequencies.
How FDM system works:
The above figure shows the schematic diagram of an FDM system. The
transmitter end contains multiple transmitters and the receiver end
contains multiple receivers. The communication channel is present between
the transmitter and receiver.
At transmitter end, each transmitter sends a signal of different frequency. In
the above figure, the transmitter 1 sends a signal of 30 kHz, transmitter 2
sends a signal of 40 kHz, and transmitter 3 sends a signal of 50 kHz. These
signals of different frequencies are then multiplexed or combined by using a
device called multiplexer. It then transmits the multiplexed signals over a
communication channel.
At the receiver end, the multiplexed signals are separated by using a
device called demultiplexer. It then sends the separated signals to the
respective receivers. In the above figure, the receiver 1 receives signal of 30
kHz, receiver 2 receives signal of 40 kHz, and receiver 3 receives signal of 50
kHz.
Advantages of Frequency Division Multiplexing (FDM)
1. It transmits multiple signals simultaneously.
2. In frequency division multiplexing, the demodulation process is easy.
3. It does not need Synchronization between transmitter and receiver.
Disadvantages of Frequency Division Multiplexing (FDM)
It needs a large bandwidth communication channel.
Applications of Frequency Division Multiplexing (FDM)
1. Frequency division multiplexing is used for FM and AM radio
broadcasting.
2. It is used in first generation cellular telephone.
3. It is used in television broadcasting.

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Digital Multiplexing
The process of combining multiple digital signals into one signal is called
digital multiplexing.
Time Division Multiplexing
Time Division Multiplexing is a technique in which multiple signals are
combined and transmitted one after another on the same communication
channel.
At the receiver side, the signals are separated and received. Each signal is
received by a user at a different time.
Time Division Multiplexing is also simply referred to as TDM. It is the digital
multiplexing technique.
In frequency division multiplexing, all signals of different frequencies are
transmitted simultaneously. But in time division multiplexing, all signals
operate with the same frequency are transmitted at different times.
In frequency division multiplexing, the sharing of a channel is done on the
basis of frequency. But in time division multiplexing, the sharing of a
channel is done on the basis of time.

In time division multiplexing, each user is allotted a particular time

interval called time slot during which data is transmitted. The time interval
(time slot) allotted to each receiver (user) is so small that the receiver will not
detect that some time was used to serve another receiver (user).

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 In time division multiplexing, all signals are not transmitted
simultaneously; instead, they are transmitted one after another. For
example, as shown in the above figure, at first, we send signal A. Then after
second signal B and then after third signal C and finally, we send last signal
D. Thus, each user occupies an entire bandwidth for a short period of time.
The time division multiplexing technique is used to multiplex analog
signals or digital signals. However, the time division multiplexing is more
suitable for digital signal multiplexing.
In time division multiplexing, the bandwidth capacity of the
communication channel should be greater than the multiple input signals.
Types of TDM (Time Division Multiplexing)
Time Division Multiplexing is mainly classified into two types:
o Synchronous TDM (Time Division Multiplexing)
o Asynchronous TDM (Time Division Multiplexing)
Synchronous TDM (Time Division Multiplexing)
synchronous time division multiplexing, each device (transmitter) is
allotted with a fixed time slot, regardless of the fact that the device
(transmitter) has any data to transmit or not. The device has to transmit
data within this time slot. If the device (transmitter) does not have any data
to send then its time slot remains empty.

As shown in the figure, the various time slots are arranged into frames
and each frame consists of one or more time slots dedicated to each device
(transmitter).

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 For example, if there are 3 devices, there will be 3 slots in each frame.
Similarly, if there are 5 devices, there will be 5 slots in each frame.
The above figure shows 4 devices (transmitter A, transmitter B,
transmitter C, and transmitter D) that have 4 dedicated time slots (time slot
A, time slot B, time slot C and time slot D).
The transmitter A data is sent at time slot A, transmitter B data is sent at
time slot B, transmitter C data is sent at time slot C and transmitter D data
is sent at time slot D.
In the time frame 2, the transmitter B and C does not have any data to send
so the time slot B and C remains empty.
The main drawback of synchronous time division multiplexing is that
the channel capacity is not fully utilized. Hence, the bandwidth goes
wasted.
Asynchronous TDM (Time Division Multiplexing)
In Asynchronous time division multiplexing, the time slots are not
fixed (I.e. time slots are flexible). The asynchronous TDM is also known as
statistical time division multiplexing.

In synchronous TDM, the number of time slots is equal to the number

of devices (transmitters). But in Asynchronous TDM, the number of time
slots is not equal to the number of devices (transmitters). The time slots in
asynchronous

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 TDM are always less than the number of devices (transmitter). For
example, if we have X devices and Y time slots. Y should always be less than
X (I.e. Y < X).
In asynchronous time division multiplexing, time slots are not fixed to
a particular device; instead, they are allotted to any of the devices that have
data to send.
In the above figure, it is shown that the number of devices are 4 and
time slots are 3. The timeframe 1 (all slots) is completely filled with data
from devices A, B, and C. The timeframe 1 has only 3 time-slots. So the data
from device D is filled in the next timeframe (I.e. timeframe 2) in timeslot 1.
The data from devices A and D will be filled in timeslots 2 and 3 in
timeframe 2.
In asynchronous time division multiplexing, the multiplexer scans all
the devices (transmitters) and accepts input only from the devices that have
actual data to send and fills all the frames, and then sends it to the receiver.
If there is not enough data to fill all the slots in a frame, then the partially
filled frames are transmitted. In most of the cases, all the time slots in
frames are completely filled.

Advantages of Time Division Multiplexing (TDM)

1. Full bandwidth is utilized by a user at a particular time.
2. The time division multiplexing technique is more flexible than frequency
division multiplexing.
3. In time division multiplexing, the problem of crosstalk is very less.

Disadvantages of Time Division Multiplexing (TDM)

In time division multiplexing, synchronization is required.

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Comparison of FDM and TDM
Sl.no FDM TDM
1 FDM stands for Frequency TDM stands for Time Division
Division Multiplexing. Multiplexing.
2 FDM is an analog technique TDM is a Digital technique.
3 The communication channel is The communication channel is
divided by frequency divided by time.
4 All signals of different frequencies All signals operate with the same
are transmitted simultaneously. frequency are transmitted at
different times.
5 Synchronization is not required Synchronization is required.
6 The bandwidth of the The bandwidth capacity of the
communication channel should be communication channel should
greater than the combined be greater than the multiple
bandwidth of individual signals. input signals.
7 FDM requires complex circuitry at TDM does not require complex
the transmitter and receiver. circuitry.
8 In FDM, the problem of crosstalk In TDM, the problem of crosstalk
is severe. is not severe.
9 The channel bandwidth is The channel bandwidth is
effectively used. wasted.
10 FDM requires Guard bands for its TDM requires sync pulse for its
operation. operation.
11 FDM is used in TV and RADIO TDM is used in Pulse code
broadcasting modulation

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 UNIT IV PULSE MODULATION PART 2

Pulse Code Modulation: PCM Generation and Reconstruction,

Quantization Noise, Non-Uniform Quantization and Companding,
DPCM, Adaptive DPCM, DM and Adaptive DM, Noise in PCM and
DM.

Pulse Modulation Technique

Lifecycle from Sound to Digital to Sound:

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 Introduction Digital Communication:

• Digital communication is a mode of communication where the

information or the thought is encoded digitally as discrete signals and
electronically transferred to the recipients.
• digital communication information flows in a digital form and the
source is generally the keyboard of the computer. A single individual is
capable of digital communication and thus it also saves wastage of
manpower and is one of the cheapest modes of communication.
• Digital communication is also a really quick way to communicate. The
information can reach the recipient within a fraction of a second. An
individual no longer has to wait to personally meet the other individual and
share his information.
Advantages of Digital Communications

1. The digital communication has mostly common structure of encoding a

signal so devices used are mostly similar.
2. The Digital Communication's main advantage is that it provides us added
security to our information signal.
3. The digital Communication system has more immunity to noise and
external interference.
4. Digital information can be saved and retrieved when necessary while it is
not possible in analog.
5. Digital Communication is cheaper than Analog Communication.
6. The configuring process of digital communication system is simple as
compared to analog communication system. Although, they are complex.
7. In Digital Communication System, the error correction and
detection techniques can be implemented easily.
Disadvantages of Digital Communications:
1). Generally, more bandwidth is required than that for analog systems.
2). Synchronization is required.
3). High power consumption (Due to various stages of conversion).

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4). Complex circuit, more sophisticated device making is also drawbacks of
digital system.
5). Introduce sampling error
6). As square wave is more affected by noise, That’s why while
communicating through channel we send sin waves but while operating on
device we use square pulses.
Pulse Code Modulation (PCM)
Definition: A technique by which analog signal gets converted into digital
form in order to have signal transmission through a digital network is
known as Pulse Code Modulation. It is abbreviated as PCM.
PCM systems are basically signal coders also known as waveform
coders. PCM allows the representation of the continuous time message
signal as a sequence of binary coded pulses. The binary form permits only 2
probable states i.e., 0 and 1.
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.
The major steps involved in PCM is sampling, quantizing and encoding .

31
Basics of PCM
In pulse code modulation, the analog message signal is first sampled, and
then the amplitude of the sample is approximated to the nearest set of
quantization level. This allows the representation of time and amplitude in a
discrete manner. Thereby, generating a discrete signal.
This discrete signal is then converted into its binary form for the
transmission of the signal.
It is to be noted here that, in PCM technique the signal gets transmitted in
the coded format and must be decoded at the receiver in order to have the
original message signal.
Block diagram of Pulse Code Modulation
The figure below shows the block diagram representing a PCM system

It is basically composed of a transmitter, a transmission path and a receiver.

The transmitter performs the sampling, quantizing and encoding of the
signal. The transmission path includes regenerative receivers that recover
the signal from the undesired noise effects.
Lastly, the receiver section that performs decoding of the coded signal after
regeneration of the signal at the receiver.

32
 PCM Transmitter
The detailed diagram of a practical PCM transmitter is shown below

 LPF: Here, the message signal which is in the continuous time form,
is allowed to pass through a low pass filter (LPF). This LPF whose cutoff
frequency is fm eliminates the high-frequency components of the signal and
passes only the frequency components that lie below fm.
 Sampler: The output of the LPF is then fed to a sampler where the
analog input signal is sampled at regular intervals. The sampling of the
signal is done at the rate of fs. This sampling frequency is so selected that it
must follow the sampling theorem that is expressed as:
fs ≥ 2fm
The output of the sampler is a signal that is discrete time continuous
amplitude signal denoted as nTs which is nothing but a PAM signal.
 Quantizer: A quantizer is a unit that rounds off each sample to the
nearest discrete level. The sampler provides a continuous range signal and
hence still an analog one. The quantizer performs the approximation of each
sample thus assigning it a particular discrete level.

33
 As it basically rounds off the value to a certain level this shows some
variation by the actual amount. Thus we can say, quantizing a signal
introduces some distortion or noise into it. This is known as quantization
error.
This noise factor is somewhat better than the channel noise as it is
controllable.
A quantizer can be of two types, uniform and non-uniform quantizer. In
uniform quantizer, there exists a uniform spacing in between the level. As
against, in non-uniform quantizer, the spacing in between the levels is not
uniform. Here, we have employed a uniform quantizer.
For a low signal level, the quantization error is high i.e., bad SNR. But,
for a high signal level, the quantization error is low providing good SNR.
The figure below shows the sampling of analog signal and further
quantization of the samples

 Encoder: An encoder performs the conversion of the quantized signal

into binary codes. This unit generates a digitally encoded signal which is a
sequence of binary pulses that acts as the modulated output.
As it is a binary encoder thus generates a binary coded sequence. That is
transmitted through the transmission path.
34
Transmission path in a PCM system
A PCM system has a better control over signal distortion introduced during
transmission through the channel than other systems. PCM achieves low
signal distortion by employing regenerative receivers along the transmission
path.
The channel introduces distortion in the signal during transmission. This
distortion is eliminated by the regenerator in order to provide a
distortionless PCM signal. Resultantly, enhancing the transmission ability of
the system.
The figure below shows the block diagram of a regenerative repeater. It
basically performs equalization and timing and then executes decision
making.

The PCM signal when provided to the regenerative repeater, the equalizer
circuit at the beginning performs the reshaping of the distorted signal. At
the same time, the timing circuit generates a pulse train that is a derivative
of input PCM pulses.
This pulse train is then utilized by the decision-making device in order
to sample the PCM pulses. This sampling is done at the instant where
maximum SNR can be achieved. In this way, the decision-making device
generates the distortionless PCM wave.
The decision device decides its output based on whether the
amplitude of the quantized pulse and the noise, exceeds a pre-determined
value or not.These are few of the techniques used in digital communications

35
 PCM Receiver
The figure below shows the functional block diagram of a PCM receiver

 Regenerator: A regenerative repeater is placed at the receiving end

also so as to have an exact PCM transmitted signal. Here, also the
regenerator works in a similar manner as that when employed in the
transmission path. It eliminates the channel induced noise and reshapes
the pulse.
 DAC and Sampler: Digital to analog converter performs the
conversion of digital signal again into its analog form by making use of the
sampler. As the actual message signal was analog thus at the receiver end
there is a necessity to again convert it into its original form.
 LPF: The sampler generates analog signal but that is not the original
message signal. Thus, the output of the sampler is fed to the LPF having
cutoff frequency fm. This is sometimes termed as the reconstruction filter
that produces the original message signal.
The process done at the transmitter is somewhat reversed at the receiver in
order to generate the original analog message signal. The figure below shows
the reconstruction of the actual analog message signal at the receiver.

36
Advantages of PCM
 Immune to channel induced noise and distortion.
 Repeaters can be employed along the transmitting channel.
 Encoders allow secured data transmission.
 It ensures uniform transmission quality.
Disadvantages of PCM
 Pulse code modulation increases the transmission bandwidth.
 A PCM system is somewhat more complex than another system.
Thus from the above discussion, we can conclude that a PCM system,
transmits data in a coded format, that ensures secured transmission. But,
this at the same time needs decoding system in order to reproduce exact
message signal that increases system complexity.
Applications:
1) PCM in Compact Disk (CD)
2) PCM in Wired Telephony
QUANTIZATION BASICS:
In PCM, conversion of analog signal to digital signal is done in two steps
Sampling and Quantization
Below figure shows sampling step:

Quantization is the process of rounding of the sample value to the nearest

quantization level. Remember that number of quantization levels is
predefined.
If n = number of bits used to represent the sample
Then, q = number of quantization levels =2n

37
Defination of Quantization:
Representing the analog sampled values by a finite set of levels is called
quantization. While sampling converts a continuous time signal to discrete
time signal.
Quantizer converts continuous amplitude to discrete amplitude
samples.
The analog signal is quantized into countable & discrete levels known
as quantization levels. Each of these levels represents a fixed input
amplitude.
During quantization, the input amplitude is round off to the nearest
quantized level. This rounding off is known as quantization error.
Quantization error can be reduced by increasing the numbers of
quantization levels.
In quantization, Approximating each sampled values to its nearest
predefined discrete amplitude level (I,e quantization level)

Process of Quantization:

i) Consider message signal m(t) ,whose range is from VL to VH

ii) Divide this range in to L equal intervals, each size Δ, Step size

Where L= No. of Quantization levels =2n

n= no.of encoded bits in PCM

iii) In the center of each of these steps, locate Quantization

levels,m0,m1,m2….. m7. ( i,e quantization error = )

38
 iv) Generate the Quantized signal mq(t).

Where so, s1, s2....... are output levels of the quantizer, and
L0,L1....... are the analog information source.

Figure: Quantization Noise

• The difference between two adjacent discrete values is called a
quantum or Step size.
• The original signal can have an infinite number of signal levels, the
quantizing process will produce errors called quantizing errors or quantizing
noise. It is defined as the difference between input and output signals is the
Quantization error(qe). or Quantization noise (OR)
• The difference between actual signal S(t) and quantized value of the
signal Sq(t) is called Quantization error(qe). Or Quantization noise q(t) = s(t) -
sq(t) S(t) - Actual value of the signal
39
• Sq(t) – Error value or Quantized value of the signal

• The minimum value of quantization error is

• Where, Step size

• If, decreases the Step size(Δ) then we can increase the no.of levels

(L), so that bandwidth also increases.

•
l

40
 Quantization error = Δ/2 = 0.25/2 = 0.125

Analog
Information Quantizat
source ion levels

11

10

01

00

Quantizing 0.4 V sample value: See that level 2 is near to 0.4 V. So, for
sample voltage 0.4 V, 01 code is transmitted.
Quantization error = 0.4 – 0.375 = 0.025 V
Quantizing 0.78 V sample: Level 4 is nearest to 0.78 V. So, digital code
11 is transmitted.

41
 Quantization error = 0.875 -0.78 = 0.095 V

NOTE: Quantization process introduces a certain amount of error or

distortion. This error known as quantization noise and is minimised by
increasing the number of quantization levels. But increasing number of
quantization levels increases number of bits to represents each sample and
hence increases bit rate and cost of transmission.
Example (2):

Sampled value

Quantized value

EX:3

42
43
Types of Quantization:
There are two types of Quantization: i) Uniform Quantization and
ii) Non-uniform Quantization.
The type of quantization in which the quantization levels are uniformly
spaced is termed as a Uniform Quantization.
(or)
If the step size (Δ) is maintained constant throughout the quantization
process is called Uniform Quantization.
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. (or)
If the step size (Δ) changes in throughout the quantization process is is
called Non-uniform Quantization.
There are two types of uniform quantization. They are
a) Mid-Rise type
b) Mid-Tread type.
Uniform Quantization: Example: Uniform n =3 bit quantizer
L=8 and XQ = {1,3,5,7}

 Most ADC’s use uniform quantizers.

 The quantization levels of a uniform quantizer are equally spaced
apart.
 Uniform quantizers are optimal when the input distribution is
uniform. When all values within the Dynamic Range of the quantizer
are equally likely.
44
The following figures represent the two types of uniform quantization.
Types of Uniform Quantization

Midtread: Origin lies in the middle of a tread of the staircase like graph in
fig (a), utilized for odd levels
Midrise: Origin lies in the middle of a rising part of the staircase like graph
fig.(b), utilized for even levels

45
 
 Many signals such as speech have a nonuniform distribution.
 The amplitude is more likely to be close to zero than to be at
higher levels.
 Nonuniform quantizers have unequally spaced levels
The spacing can be chosen to optimize the SNR for a particular type of
signal

Application: Commonly used for speech

Companding
• Companding is a technique of achieving non-uniform quantization. It
is a word formed by the combination of words compression and
expanding. Companding is done in order to improve SNR of weak
signals.
• We know if the characteristics of the quantizer is non-linear then it
causes the step size to be variable despite being constant then it is
known as non-uniform quantization.
• As we know in non-uniform quantization, the step size varies
according to the signal level. If the signal level is low then step size will
be small. So, the step size will be low for weak signal. Thus the
quantization noise will also be low.
• So, in order to maintain proper signal to quantization noise ratio, the
step size must be variable according to the signal level.
• Thus in order to achieve non-uniform quantization the process of
companding is used.
• The process of compressing and expanding is called Companding.

46
 • The higher amplitude analog signals are compressed prior to
transmission and then expanded in receiver. Improving the Dynamic
Range of a communication system.

Initially at the transmitting end the signal is compressed and further at the
receiving end the compressed signal is expanded in order to have the
original signal.
Companding Functions:

47
Companding Characteristics in Non uniform quantization: :

Method of Companding:
For the compression, two laws are adopted: the -law in US and Japan and
the A-law in Europe.
 -law Vmax ln( 1   Vin Vmax )
Vout 
ln( 1   )
 A-law
 AVin Vmax  Vin 1
 Vmax 0 
 1  ln A Vmax A

1  ln( A Vmax )
Vout Vin
1 Vin
 1
 1  ln A A Vmax

Where: Vmax= Max uncompressed analog input voltage

Vin= amplitude of the input signal at a particular of instant time
Vout= compressed output amplitude
A, = parameter define the amount of compression
 The typical values used in practice are: =255 and A=87.6.
 After quantization the different quantized levels have to be represented
in a form suitable for transmission. This is done via an encoding
process.
Example: A companding system with µ = 200 is used to compand -4V to 4V
signal. Calculate the system output voltage for Vin = -4, -2, 0, 2 and 4V.
Equation: Vmax ln( 1   Vin Vmax )
Vout 
ln( 1   )

48
 Vin (V) -4 -2 0 2 4

Vout (V) -4 -3.48 0 3.48 4

Plot the compression characteristic that will handle input voltage in the
given range and draw an 8 level non-uniform quantizer characteristic that
corresponds to the given µ.

SNR Performance of Compander:

• The output SNR is a function of input signal level for uniform

quantizing.
• But it is relatively insensitive for input level for a compander.
• α = 4.77 - 20 Log ( V/xrms) for Uniform Quantizer
V is the peak signal level and xrms is the rms value
• α = 4.77 - 20 log[Ln(1 + μ)] for μ-law companding
• α = 4.77 - 20 log[1 + Ln A] for A-law companding
PCM Encoding

 The output of the quantizer is one of L possible signal levels.

If we want to use a binary transmission system, then we need to
map each quantized sample into an n bit binary word.
 Encoding is the process of representing each quantized sample by n
bit code word.
The mapping is one-to-one so there is no distortion introduced
by encoding.
49
Table: Encoding levels with belonging code words
Example:

Bit rate and Bandwidth requirements of PCM:

The transmission bandwidth of a PCM system is associated with a
number of bits per sample. If the number of bits per sample increases, the
bandwidth also increases. In order to have a good approximation, a large
number of levels must be used but that will lead to a larger bandwidth
requirement.
The bandwidth of (serial) binary PCM waveforms depends on the bit rate (rb)
and the waveform pulse shape used to represent the data.
The no. of quantization levels L=2n, where n is the no. of bits/sample.
The sampling rate fs = no. of samples/sec
The bit rate of a PCM signal can be calculated form the number of bits per
sample x the sampling rate

50
 Bit rate (rb) = nb x fs
= nb x 1/Ts
The bandwidth required to transmit this signal depends on the type of line
encoding used. Practically, The transmission bandwidth to be a little higher
than the minimum bandwidth required.
• The min. bandwidth of a PCM System is half of the bit rate(rb) hence
B.W of PCM = rb/2
= nb x fs / 2
BW ≥ nfm
Noise considerations in PCM
The performance of a PCM system is influenced by two major sources of
noise:
1. Channel noise, which is introduced anywhere between the transmitter
output and the receiver input. Channel noise is always present, once the
equipment is switched on.
2. Quantization noise, which is introduced in the transmitter and is
carried all the way along to the receiver output. Unlike channel noise,
quantization noise is signal dependent in the sense that it disappears when
the message signal is switched off.
The main effect of channel noise is to introduce bit errors into the
received signal. In the case of a binary PCM system, the presence of a bit
error causes symbol 1 to be mistaken for symbol 0, or vice versa. Clearly,
the more frequently bit errors occur, the more dissimilar the receiver output
becomes compared to the original message signal.
The fidelity of information transmission by PCM in the presence of
channel noise may be measured in terms of the average probability of
symbol error, which is defined as the probability that the reconstructed
symbol at the receiver output differs from the transmitted binary symbol, on
the average. The average probability of symbol error, also referred to as the
bit error rate (BER), assumes that all the bits in the original binary wave are
of equal importance.

51
 To optimize system performance in the presence of channel noise, we
need to minimize the average probability of symbol error. For this
evaluation, it is customary to model the channel noise as additive, white,
and Gaussian. The effect of channel noise can be made practically negligible
by ensuring the use of an adequate signal energy-to-noise density ratio
through the provision of short-enough spacing between the regenerative
repeaters in the PCM system.
Quantization noise is essentially under the designer's control. It can
be made negligibly small through the use of an adequate number of
representation levels in the quantizer and the selection of a companding
strategy matched to the characteristics of the type of message signal being
transmitted.
(Extra information reated noise in PCM system)
Noise in PCM Systems related to Probability of bit error and Signal to
Noise ratio (SNR):
 Two main effects produce the noise or distortion in the PCM output:
– Quantizing noise that is caused by the M-step quantizer at the
PCM transmitter.
– Bit errors in the recovered PCM signal, caused by channel noise
and improper filtering.
• If the input analog signal is band limited and sampled fast enough so
that the aliasing noise on the recovered signal is negligible, the ratio of
the recovered analog peak signal power to the total average noise
power is:

•
• The ratio of the average signal power to the average noise power is

•
– M is the number of quantized levels used in the PCM system.

52
 – Pe is the probability of bit error in the recovered binary PCM
signal at the receiver DAC before it is converted back into an
analog signal.

• If Pe is negligible, there are no bit errors resulting from channel noise

and no ISI, the Peak SNR resulting from only quantizing error is:

•
• The Average SNR due to quantizing errors is:

•
• Above equations can be expresses in decibels as,

•
• Where, M = 2n
• α = 4.77 for peak SNR α = 0 for average SNR

53
Problem (1):
A message signal m(t)= 5cos 4Π103t is transmitted through the channel
using 5- bit PCM. The sampling rate is twice the Nyquist rate. Calculate all
parameters of PCM.

Solution: +5v

-5v

n=5

5cos 4Π103t = 5cos 2×2Π103t = fm = 2khz

(iii) Sampling rate is twice the Nyquist rate

54
Problem (2):
A message signal m(t)= 8cos 4Π103t is transmitted through the channel
using 3-bit PCM. The sampling rate is 50% higher then the Nyquist rate.
a)Calculate all parameters of PCM.
b) If the sampled values are (7.8, 5, 2.2, -1.8, -3.7, -5.2, -7.2, -7.9) calculate
the quantized output, code number, encoder output and find the
quantization error in each sample.
Solution: +8v

-8v
n=3

8cos 4Π103t = 8cos 2×2Π103t = fm = 2khz

Sampling rate is 50% higher then the Nyquist rate

55
b) Sampled values are (7.8, 5, 2.2, -1.8, -3.7, -5.2, -7.2, -7.9)

Sampled Quantized Code Encoder Quantization

value output Number output error
7.8 7 7 111 0.8
5 5 6 110 0
2.2 3 5 101 -0.8
-1.8 -1 3 011 -0.8
-3.7 -3 2 010 -0.7
-5.2 -5 1 001 -0.2
-7.2 -7 0 000 -0.2
-7.9 -7 0 000 -0.9

56
Differential Pulse Code Modulation:
Differential Pulse Code Modulation (DPCM) is a procedure of
converting an analog into a digital signal in which an analog signal is
sampled and then the difference between the actual sample value and its
predicted value (predicted value is based on previous sample or samples) is
quantized and then encoded forming a digital value.
Which means, its value from present sample to next sample does not
vary by a large amount.The adjacent samples of the signal carry the same
information with a little difference.
DPCM code words represent differences between samples unlike PCM
where code words represented a sample value.
Redundant Information in PCM
Fig.1 shows a continuous time signal x(t) by dotted line. This signal is
sampled by flat top sampling at intervals Ts , 2Ts , 3Ts ….. nTs .

Fig.1: Illustration of redundant information in PCM

The sampling frequency is selected to be higher than nyquist rate. The
samples are encoded by using 3 bit (7 levels) PCM. The sample is quantized
to the nearest digital level as shown by small circles in fig.1.
The encoded binary value of each sample is written on the top of the
samples. We can observe from fig.1 that the samples taken at 4Ts , 5Ts and
6Ts are encoded to same value of (110). This information can be carried only
by one sample. But three smaples are carrying the same information means
that it is redundant.
57
 We consider another example of samples taken at 9Ts and 10Ts. The
difference between these samples only due to last bit and first two bits are
redundant, as they do not change. If this redundancy is reduced, then
overall bit rate will decrease and number of bits required to transmit one
sample will also be reduced. This type of digital pulse modulation technique
is called as Differential Code Modulation (DPCM).
Working Principle

The differential pulse code modulation works on the principle of

prediction. The value of the present sample is predicted from the past
samples.The prediction may not be exact but it is very close to the actual
sample value.
Fig.2 shows the transmitter of DPCM system.

The sampled signal is denoted by x(nTs) and predicted signal is

denoted by xˆ(nTs). The comparator finds out the difference between the
actual sample value x(nTs) and predicted sample value xˆ(nTs). This is
known as prediction error and it is denoted by e(nTs).
It can be defined as,
e(nTs) = x(nTs) – xˆ(nTs)..............................(1)

58
 The predicted value is produced by using a prediction filter. The
quantizer output signal gap eq(nTs) and previous prediction is added and
given as input to the prediction filter. This signal is called xq(nTs). This
makes the prediction more and more close to the actual sampled signal.
We can observe that the quantized error signal eq(nTs) is very small
and can be encoded by using small number of bits.
Thus number of bits per sample are reduced in DPCM. The quantizer output
can be written as ,
eq(nTs) = e(nTs) + q(nTs)………………………..(2)
Here, q(nTs) is the quantization error.
As shown in fig.2, the prediction filter input xq(nTs) is obtained by
sum xˆ(nTs) and quantizer output. i.e.,
xq(nTs) = xˆ(nTs) + eq(nTs)……………………..(3)
Substituting the value of eq(nTs) from eq.(2) in the above eq. (3) , we get,
xq(nTs) = xˆ(nTs) + e(nTs) + q(nTs) ………………….(4)
eq.(1) is written as,
e(nTs) = x(nTs) – xˆ(nTs)
∴ e(nTs) + xˆ(nTs) = x(nTs)
Therefore, substituing the value of e(nTs) + xˆ(nTs) from the above
equation into eq. (4), we get,
xq(nTs) = x(nTs) + q(nTs) …………………..(5)
The quantized signal xq(nTs) is the sum of Original sample values x(nTs)
and quantization error q(nTs)
Reception of DPCM Signal
Fig.3 shows the block diagram of DPCM receiver.

Fig.3: DPCM Receiver

59
  The decoder first reconstructs the quantized error signal from
incoming binary signal.
 The prediction filter output and quantized error signals are summed
up to give the quantized version of the original signal.
 Thus the signal at the receiver differs from actual signal by
quantization error q(nTs), which is introduced permanently in the
reconstructed signal.

Advantages of DPCM
 As the difference between x(nTs) and xˆ(nTs) is being encoded and
transmitted by the DPCM technique, a small difference voltage is to be
quantized and encoded.
 This will require less number of quantization levels and hence less
number of bits to represent them.
 Thus signaling rate and bandwidth of a DPCM system will be less
than that of PCM.
Applications of DPCM:
• The DPCM technique mainly used Speech, image and audio signal
compression.
• The DPCM conducted on signals with the correlation between
successive samples leads to good compression ratios. In images, there
is a correlation between the neighboring pixels, in video signals, the
correlation is between the same pixels in consecutive frames and
inside frames (which is the same as correlation inside the image).
• This method is suitable for real-Time applications.
• To understand the efficiency of this method of medical compression
and real-time application of medical imaging such as telemedicine and
online diagnosis.
• Therefore, it can be efficient for lossless compression and
implementation for lossless or near-lossless medical image
compression.

60
Adaptive Differential Pulse Code Modulation (ADPCM)
• Adaptive Differential Pulse Code Modulation (ADPCM)– Very popular
waveform coding technique
• Main application is Telecommunications– Speech compression for
transmission,
• Storage and reconstruction– Reduce the bit data rate while
maintaining good voice quality.
• Technique can apply to all waveforms which need high-quality audio,
image and modem data, Integrated Circuit Design Project.
Disadvantages of DPCM:
1) i. In DPCM we subtract adjacent samples and code the differences,
however this method is inefficient since they do not adapt themselves
to varying magnitudes of audio stream.
2) ii. Hence, better results are obtained using ADPCM (Adaptive
Differential Pulse Code Modulation).
3) iii. ADPCM uses the previous sample to predict the current sample.
4) iv. It then computes the difference between the current sample and its
predictions and quantizes the difference.

Figure illustrates the basic configuration of ADPCM.

• The number of quantization level L is fixed.
• When a fixed quantization step Δv is applied, either the quantization
error is too large because Δv is too big or the quantizer cannot cover
the necessary signal range when Δv is too small.

61
 • Therefore, it would be better for the quantization step Δv to be
adaptive so that Δv is large or small depending on whether the
prediction error quantizing is large or small.
• It is important to note that the quantized prediction error can be
good indicator of the prediction error size.
Applications:
• ADPCM speech coder and decoder (codec )
• ADPCM codec are used in the personal handyphone systems(PHS) or
Personal Access System ( PAS ).
• The PHS is a mobile network system similar to a cellular network. It is
a short range low power facility and operates in the 1880 to 1930 MHz
frequency band.
Delta Modulation (DM):
Definition: A modulation technique that converts or encodes message
signal into a binary bit stream is known as Delta Modulation. Here only 1
bit is used to encode 1 voltage level thus, the technique allows transmission
of only 1 bit per sample.
As PCM has the property of converting message signal directly into a
sequence of a binary coded pulse, this resultantly increases the bandwidth
requirement of the system. So, in order to remove the drawbacks of PCM,
delta modulation is used.
Working Principle
Delta modulation transmits only one bit per sample. Here, the present
sample value is compared with the previous sample value and this result
whether the amplitude is increased or decreased is transmitted.
Input signal x(t) is approximated to step signal by the delta modulator.This
step size is kept fixed.
The difference between the input signal x(t) and staircase approximated
signal is confined to two levels, i.e., +Δ and -Δ.
When +Δ is noticed i.e., increase in step size, then 1 is
transmitted. However, in the case of –Δ i.e., decrease in step size, 0 is
transmitted. Hence, DM allowing only a single binary bit to get
transmitted for each sample. (OR)
62
When the step is reduced, ‘0’ is transmitted and if the step is increased, ‘1’
is transmitted. Hence, for each sample, only one binary bit is
transmitted.
Below Fig.1. Delta Modulation Waveform shows the analog signal x(t) and
its staircase approximated signal by the delta modulator.

DM Transmitter:
Fig. 2 (a) shows the transmitter. It is also known as Delta modulator.

Fig.2 (a) Delta Modulation Transmitter

It consists of a 1-bit quantizer and a delay circuit along with two summer
circuits.
Mathematical Expressions
The error between the sampled value of x(t) and last approximated sample is

given as:
Where e( nTs) = error at present sample
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x(nTs) = sampled signal of x(t)

If we assume u(nTs) as the present sample approximation of staircase

output, then

Let us define a quantity b( nTs) in such a way that,

This means that depending on the sign of error e( nTs) , the sign of step
size Δ is decided. In other words we can write

Also if b (nTs) =+Δ then a binary ‘1’ is transmitted

and if b (nTs) =-Δ then a binary ‘0’ is transmitted
Here Ts = sampling interval.
The summer in the accumulator adds quantizer output (±Δ) with the
previous sample approximation. This gives present sample approximation.
i.e.,

The previous sample approximation u[(n-1)Ts ] is restored by delaying one

sample period Ts .
The samples input signal x(nTs ) and staircase approximated signal xˆ(nTs )
are subtracted to get error signal e(nTs ).
Thus, depending on the sign of e(nTs ), one bit quantizer generates an output
of +Δ or -Δ .
If the step size is +Δ, then binary ‘1’ is transmitted and

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 if it is -Δ, then binary ‘0’ is transmitted .

Detection of delta modulated signal:

The detection of a delta modulated signal is not a complex process
and is somewhat reverse of generation of a delta modulated signal.
The figure below shows the block diagram for the representation of detection
of delta modulated signal.

The detection circuitry basically consists of an accumulator and an

LPF. The predictor circuit is eliminated here and hence no assumed input is
given to the demodulator.
The binary signal transmitted is provided to the accumulator section.
The accumulator consists of a summation unit and a delay unit. The
transmitted signal along with the delayed signal is added at the summation
unit.
If here the input is binary 1 then after a delay the output of the
accumulator shown increased step size +Δ. However, in the case of binary 0
as input, a decrease in step size is noticed. This generates the staircase
signal equivalent to the message signal.
The output of the accumulator is provided to the LPF that smoothens
the staircase signal in order to regenerate the original message signal.

Advantages of delta modulation

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 Due to transmission of 1 bit per sample, it permits less channel
bandwidth as well as signaling rate.
 ADC is not required. Thus permits easy generation and detection.
Applications of delta modulation
It is widely used in radio communication devices and digital voice storage
and voice information transmission where signal quality is less important.
Noise in delta modulation (or)
Disadvantages of delta modulation:
The delta modulation has two major drawbacks as under:
1. Slope overload distortion or error
2. Granular or idle noise
Now, we will discuss these two drawbacks in detail.
1. Slope Overload Distortion or error
In Slope overload distortion, The amplitude of the original signal is high
when compared to the step size signal and there is no approximation in
between two signals. I,e, the amplitude of the original signal overcomes the
stepsize signal.
This distortion arises because of large dynamic range of the input
signal.

Fig.1: Quantization Errors in Delta Modulation

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
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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.
2. Granular or Idle Noise
Granular or Idle noise occurs when the step size is too large compared
to small variation in the input signal.
This means that for very small variations in the input signal, the
staircase signal is changed by large amount (Δ) because of large step size.
Above Fig.1 shows that when the input signal is almost flat , the staircase
signal u(t) keeps on oscillating by ±Δ around the signal.
The error between the input and approximated signal is called granular
noise. The solution to this problem is to make the step size small.
Adaptive Delta Modulation:
Reasons to use Adaptive Delta Modulation:
In order to overcome the quantization errors due to slope overload and
granular noise, the step size (Δ) is made adaptive to variations in the input
signal x(t).
Particularly in the steep segment of the signal x(t), the step size is
increased. And the step is decreased when the input is varying slowly. This
method is known as Adaptive Delta Modulation (ADM).
The adaptive delta modulators can take continuous changes in step
size or discrete changes in step size. or
Change step size according to the changes in the input signal.
ADM Transmitter:
Fig.1 shows the transmitter of an ADM.

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  The logic for step size control is added in the diagram.
 The step size increases or decreases according to a specified rule
depending on one bit quantizer output.
 For an example, if one bit quantizer output is high (i.e., 1), then step
size may be doubled for next sample.
 If one bit quantizer output is low, then step size may be reduced by
one step.
Fig.2 shows the staircase waveforms of adaptive delta modulator and
sequence of bits to be transmitted.

Fig.2 : Waveform for adaptive delta modulation

ADM Receiver:
Fig.3 shows the receiver of an ADM.

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 Fig.3: ADM Receiver
The receiver has two portions. The first portion produces the step size
from each incoming bit.Exactly the same process is followed as that in
transmitter.
The previous input and present input decide the step size.
It is then applied to the second portion i.e., an accumulator which builds up
staircase waveform.
The low pass filter then smoothens out the staircase waveform to
reconstruct the original signal.
Advantages of Adaptive Delta Modulation
Adaptive delta modulation has certain advantages over delta modulation as
under :
1. The signal to noise ratio of ADM is better than that of DM because of
the reduction in slope overload distortion and idle noise.
2. Because of the variable step size , the dynamic range of ADM is wider
than DM.
3. Utilization of bandwidth is better in ADM than DM.

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Comparison of PCM, DPCM, DM and ADM:

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