MODULE 3

PULSE MODULATION

INTRODUCTION
• Many Signals in Modern Communication Systems are digital . Also, analog signals are
transmitted digitally.
• Reduced distortion and improvement in signal to noise ratios.
• PAM, PWM , PPM , PCM and DM.
• Data transmission, digital transmission, or digital communications is the physical transfer of
data (a digital bit stream or a digitized analogue signal) over a point-to-point or point-to-
multipoint communication channel.

Ex: optical fibers, wireless channels, computer buses....

➢ ELEMENTS OF DIGITAL COMMUNICATION SYSTEMS

Fig.Block diagram of Digital Communication

1. Discrete Information Source: It generates message to be transmitted. Examples are the data
from computers, text data or tele type data.

2. Source Encoder: It assigns codes to the symbols (samples) generated from discrete
information source. The code word having n number of bits. Each distinct sample having

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 distinct(unique) code word. If code word length is 8 bit(n), we can have 256 distinct
symbols(ie.,2^n).

3. Channel Encoder: We know that channel is the major source of notice due to that there are
more chance of getting errors while propagating through channel. To avoid that channel
encoding is required. In that extra bits are added to the binary sequence generated by the
source encoder. These extra bits are called as redundant bits. These bits are defined with
proper logic. The redundant will be helpful to detect the errors at the receiver bit sequence.

4. Digital Modulator: In digital modulator the message signal is digital data and carrier is
analog one, in most cases we use sinusoidal waves. Some examples are ASK,FSK,PSK.MRI
techniques.

5. Channel: It provides the link between transmitter and rceiver. Channel may be wired or
wireless channel.

❖ Problems associated with channel:

1. Addictive Noise: This noise is occur due to internal solid state devices or resistors used in
channel.

2. Ampltude and Phase Distortion: This noise is occurred due to non-linear characteristics of the
channel.

3. Attenuation: This is due to internal resistance of the channel.

6. Demodulator: This device is used to detect the digital message signal from the
modulated signal.

7. Channel Decoder: This is used to detect and correct the errors that occur in the digital
message signal.

8. Source Decoder: This produces the sampling signal from the given digital message signal.

9. Destination: The sampled signal is converted into audio signal or video signal or any text signal
depending on the signal.

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 Fig. Basic block diagram of an A/D converter

Advantages of digital communication systems

1. Easy way of transmission of signals

2. Connection of more calls through one channel i.e., Multiplexing is possible using Digital
Communication.
3. Source Encoding and Channel Encoding can be used to detect errors at the received
signal.

4. Using repeaters between source and destination, we can reproduce the original signal
with less distortions.
5. Security is the major advantage of digital communication compared to Analog
Communication.
6. Transmitting analogue signals digitally allows for greater signal processing capability.
7. Digital communication can be done over large distances through internet and other
things.
8. The messages can be stored in the device for longer times, without being damaged.
9. Advancement in communication is achieved through Digital Communication.

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 Disadvantages of digital communication systems

1. Sampling Error
2. Digital communications require greater bandwidth than analogue to transmit the same
information.
3. The detection of digital signals requires the communications system to be synchronized,
whereas generally speaking this is not the case with analogue systems.
4. Digital signals are often the approximation of voice signals, ie, we don‟t get the exact
analogue signal.

➢ TYPES OF MODULATION – TREE DIAGRAM

In Continuous Wave modulation schemes some parameter of modulated wave varies

continuously with message.
In Analog pulse modulation some parameter of each pulse is modulated by a particular sample
value of the message.

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 Pulse modulation of two types
1. Analog Pulse Modulation
• Pulse Amplitude Modulation (PAM)
• Pulse width Modulation (PWM)
• Pulse Position Modulation (PPM)
2. Digital Pulse Modulation
• Pulse code Modulation (PCM)
• Delta Modulation (DM)

1. Analog Pulse Modulation

Analog pulse modulation results when some attribute of a pulse varies continuously in one-to-one
correspondence with a sample value. In analog pulse modulation systems, the amplitude, width, or
position of a pulse can vary over a continuous range in accordance with the message amplitude at
the sampling instant, as shown in Figure 6.2. These lead to the following

Three types of pulse modulation:

1. Pulse Amplitude Modulation (PAM)
2. Pulse Width Modulation (PWM)
3. Pulse Position Modulation (PPM)

PAM: In this scheme high frequency carrier (pulse) is varied in accordance with sampled value
of message signal.

PWM: In this width of carrier pulses are varied in accordance with sampled values of message
signal. Example: Speed control of DC Motors.

PPM: In this scheme position of high frequency carrier pulse is changed in accordance with the
sampled values of message signal.

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 Fig. Representation of Various Analog Pulse Modulations

2. Digital Pulse Modulation

In systems utilizing digital pulse modulation, the transmitted samples take on only discrete
values. Two important types of digital pulse modulation are:
1. Delta Modulation (DM)

2. Pulse Code Modulation (PCM)

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5.5 Aliasing:

8. What is meant by aliasing effect, how it could be rectified?(Dec 2015)(or)

Write a detailed note on Aliasing and Signal Restoration. (06m) (April/May 2018) (or)
Explain in detail about aliasing.(or)
Explain in detail about the effects of undersampling.(Nov 2015)(or)
Write short notes on signal reconstruction (3m) (Nov 2017)
[Apr - 2019]
Aliasing Phenomenon

 Derivation of the sampling theorem, is based on the assumption that the signal g (t ) is strictly
band-limited.
 In practice, the information-bearing signalfrom the source is not a strictly band-limited signal.
 So, it resultsin some degree of undersampling.
 As a result, aliasing is produced by the sampling process.

Figure 8. (a) Spectrum of a signal. (b) Spectrum of an undersampled version of the signal,
exhibiting the aliasing phenomenon.

 Aliasing effect:
 Aliasing refers to the phenomenon of a high-frequency component in the spectrum of
the signal interferes and appears as lower frequency in the spectrum of its sampled
version, (as illustrated in Fig.)
 The aliased spectrum shown by the solid curve in Fig. 8(b) is related to an “undersampled”
version of the message signal represented by the spectrumof Fig. (a).
 To reduce the effects of aliasing in practice, thereare two corrective measures:
1. Before sampling, a low-pass anti-alias filter is used to attenuate those high-
frequencycomponents of the message signal that are not essential to the information
being conveyedby the signal.
2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

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  The use of a sampling rate higher than the Nyquist rate eases the design of the synthesis filter
which is used to recover the original signal from its sampledversion.
 Consider the example of a message signal that has been anti-alias (low-pass) filtered,
resulting in the spectrum shown in Fig. 9(a).
 The spectrum of theinstantaneously sampled version of the signal is shown in Fig. 9(b),
assuming a samplingrate higher than the Nyquist rate.
 Fromfig. 9(b), the design of a physically realizable reconstruction filter to recoverthe original
signal from its uniformly sampled version may be achieved as follows (seeFig. 9(c)):
 The reconstruction filter is of a low-pass kind with a passband extending from  W to
W , which is itself determined by the anti-alias filter.
 The filter has a non-zero transition band extending (for positive frequencies) from W
to f s  W , where f s is the sampling rate.

Fig 9 (a) Anti-alias filtered spectrum of an information-bearing signal. (b) Spectrumof

instantaneously sampled version of the signal, assuming the use of a sampling rate greaterthan the
Nyquist rate. (c)Idealized amplitude response of the reconstruction filter.

 The non-zero transition band of the filter assures physical realizability, it is shown as dashed
linesto emphasize the arbitrary way of actually realizing it.
****

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5.6 Signal Reconstruction

Reconstruction of a message process from its samples

9. Explain in detail about the reconstruction message process from its samples. (or)
Derive the mean square value of error in reconstruction process. (Dec 2015)
 This process completes the sampling process.
 Consider a wide-sense stationary message process X (t ) with autocorrelation function RX ( )
and power spectral density S X ( f ) .
 We assume that

S x ( f )  0 for f  W (01)

 Consider an infinite sequence of samples taken at a uniform rate equal to 2W , that is, twice
the highest frequency component of the process.
 Using X ' (t ) to denote the reconstructed process, based on this infinite sequence of samples,
we may write

 n  (02)
X ' (t )   X  2W  sinc (2Wt  n)
n  

where X (n / 2W ) is the random variable obtained by sampling or observing the message process

X (t ) at time t  n / 2W .
 The mean-square value of the error between the original message process X (t ) and the
reconstructed message process X ' (t ) equals

ξ= E[( X (t )  X ' (t )) ] 2

= E[( X (t )]  2E[ X (t ) X ' (t )]  E[( X ' (t )) ] (03)

2 2

 The first expectation term on the right side of Eq. (03) as the mean-square value of X (t ) ,
which equals RX (0) ; thus

E[( X 2 (t )] = RX (0) (04)

 For second expectation term, use Eq. (02) and so write

 
 n  
E[ X (t ) X ' (t )] = E  X (t )  X  sinc (2Wt  n)
 n  2W  

 Interchanging the order of summation and expectation:

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 
  n 
E[ X (t ) X ' (t )] =  E  X (t ) X  2W sinc (2Wt  n)
n


 n 
= R
n
X t 
 2W 
 sinc (2Wt  n) (05)

 For a stationary process, the expectation E[ X (t ) X ' (t )] is independent of time t.

 Hence, putting t=0in the right side of Eq. (05) and recognizing that
 n   n 
RX    = RX  
 2w   2W 
 It may be written as

 n 
E[ X (t ) X ' (t )] = R
n  
X   sinc (n)
 2W 
(06)

 n 
 The term RX   represents sample of the autocorrelation function RX ( ) taken at   n / 2W .
 2W 
 Now, since the power spectral density S X ( f ) or equivalently the F.T. of RX ( ) is zero for

f  W , we may represent RX ( ) in terms of its samples taken at   n / 2W as follows


 n 
RX ( ) = R
n
X  sinc (2W  n)
 2W 
(07)


 n 
If   0  RX (0) = R
n
X  sinc (n)
 2W 

Accordingly, we deduce from Eqs. (06) and (07) that

(06)  E[ X (t ) X ' (t )] = RX (0) (08)

 For third and final expectation term on the right side of Eq. (03), we again use Eq. (02) and so
write
   n  
 k  
2
E[( X ' (t )) ] = E  X   sinc ( 2Wt  n )  X  sinc (2Wt  k )
n  2W  k   2W  

  
 n   k  
= E   sinc (2Wt  n)  X  X   sinc (2Wt  k )
n k   2W   2W  
 Interchanging the order of expectation and inner summation:

 
  n   k 
E[( X ' (t ))2 ] = 
n
sinc ( 2Wt  n )  EX  X   sinc (2Wt  k )
k    2W   2W 

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  
n  k 
=  sinc (2Wt  n)  R
n k 
X  2W  sinc (2Wt  k ) (09)

 However, in view of Eq. (07), the inner summation on the right side ofEq. (09)equals
 n 
RX  t  .
 2W 
 Hence, we may simplify Eq. (09) as follows

E[( X ' (t ))2 ]  n  (10)
= R
n
X  2W  sinc (2Wt  n)
t 

= RX (0)
 Finally, substituting Eqs. (04) , (08), into (10), we get the result
ξ =0
as should be expected.

 We may therefore state the sampling theorem for message processes as follows.
 If a stationary message process contains no frequencies higher than W hertz, it may be
reconstructed from its samples at a sequence of points spaced 1/2Wseconds apart with
zero mean squared error (i.e., Zero error power).

5.7 Quantization
10. Explain in detail about the quantization process. [Apr 2010, Apr 2011]
(or)
Illustrate and describe the types of quantizer? Describe the midtread and midrise type
characteristics of uniform quantizer with a suitable diagram. [Dec 2016]

 A continuous signal (i.e., voice) has a continuous range of amplitudes and therefore its
samples also have a continuous amplitude range.
 In other words, within the finite amplituderange of the signal, there are infinite number of
amplitude levels.
 It is not necessary in fact to transmit the exact amplitudes of the samples.

 Any human sense (the ear or the eye), can detect only finite intensity differences.
 So, the original continuous signal will be approximated by a signal constructed of discrete
amplitudes.
 The existence of a finite number of discrete amplitude levels is a basic condition of pulse-code
modulation.

 Amplitude quantization is defined as the process of transforming the sample amplitude

m(nTs) of a message signal m(t) at time t = nTs into a discrete amplitude v(nTs) taken from a
finite set of possible amplitudes.

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 Fig: 10. Description of a memoryless quantizer

 Assume that the quantization process ismemoryless and instantaneous.

 It means, the transformation at time t = nTs is not affected by earlier or later samples of the
message signal.
 This simple form of scalar quantization is commonly used in practice.

 When dealing with a memoryless quantizer, we may simplify the notation by dropping the
time index.
 The symbolm in place of m(nTs)as indicated in the block diagram of a quantizer shown in
Figure 10a.
 Then, as shown in Figure. 10b, the signal amplitude m is specified by the index k if it lies
inside the partition cell

where L is the total number of amplitude levels used in the quantizer.

 The discrete amplitudesmk,k = 1, 2, ... , L, at the quantizer input are called decision levels or
decision thresholds.

 At the quantizer output, the index k is transformed into an amplitude vk that represents all
amplitudes of the cell .
 The discrete amplitudes vk ,k = 1, 2, ... , L, are called representation levels or reconstruction
levels,
 The spacing between two adjacent representation levels is called a quantum size or step-size.
 Thus, the quantizer output v equals vk if the input signal sample m belongs to the interval .
 The mapping,

is the quantizer characteristic, which is a staircase function by definition.

 Types of quantizers:
 Quantizers can be of a uniform or nonuniform type.
 In a uniform quantizer, therepresentation levels are uniformly spaced; otherwise, the
quantizer is nonuniform.

5.7.1 Uniform & non-uniform quantization:

11. Illustrate and describe the types of quantizer? Describe the midtread and midrise type
characteristics of uniform quantizer with a suitable diagram. [Dec 2016]
 In a uniform quantizer, the representation levels are uniformly spaced; otherwise, the
quantizer is nonuniform.

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 5.7.1.1 Uniform Quantization
 The quantizer characteristic can also be of a midtread or midrise type.
 Midtread:
 Figure 11(a) shows the input–output characteristic of a uniform quantizer of the
midtread type
 It is so called because the origin lies in the middle of a tread of the staircaselike
graph.

Figure 11. Two types of quantization: (a) midtread and (b) midrise.
 Midrise:
 Figure 11(b) shows the corresponding input–output characteristic of a uniform
quantizer of themidrise type.
 It is so called becausethe origin lies in the middle of a rising part of the staircaselike
graph.
 Note that both the midtread and midrise types of uniformquantizers aresymmetric about the
origin.

5.7.1.2 Nonuniform Quantization

12. Explain non-uniform quantization. (Apr 2010, Apr 2011, May 2014)

 The sampled version of the message signal will be quantized.

 Quantization provides a newrepresentation of the signal that is discrete in both time and
amplitude.
 In some applications,it is preferred to use a variable separation between therepresentation
levels.
 For example, the range of voltages covered by voice signals,from the peaks of loud talk to the
weak passages of weak talk, is on the order of1000 to 1.

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  By using a nonuniformquantizer with the feature that the step size increasesas the separation
from the origin of the input–output amplitude characteristic isincreased
 The large end-step of the quantizer can take care of possible excursions ofthe voice signal
into the large amplitude ranges that occur in rare.

 The use of a nonuniformquantizer is equivalent to passing the message signalthrough a

compressor and then applying the compressed signal to a uniform quantizer.

  law

 A particular form of compression law that is used in practice is the so called   law defined
by

(01)
where the logarithm is the natural logarithm; m andv are respectively the normalized input and
output voltages, and  is a positive constant.

Figure 12. Compression laws. (a) m-law. (b) A-law.

 For convenience of presentation, the input to the quantizer and its output are both normalized
to occupy a dimensionless range of values from zero to one, as shown in Figure12(a); here
  law is plotted for varying  .
 Practical values of  tend to be approximately 255. The case of uniform quantization
corresponds to   0 .

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 Chapter-1
Introduction
Communication has been one of the greatest needs of the human race. It is essential to form social
unions, to educate the young, and to express a myriad of emotions and needs. Good communication
is central to a civilized society.
1.1 Digital communication system
Digital communication systems are communication systems where the information propagates
through the system in the form of symbols that are discrete or digital. It uses digital sequence as an
interface between the source and the channel input (and likewise between the channel output and
final destination).

Block diagram of digital communication system

1. Information Source and Input Transducer: The source of information can be analog or digital,
e.g. analog: audio or video signal, digital: like teletype signal. In digital communication the signal
produced by this source is converted into digital signal which consists of 1′s and 0′s. For this we
need a source encoder.
2. Channel Encoder: The information sequence is passed through the channel encoder. The purpose
of the channel encoder is to introduce, in controlled manner, some redundancy in the binary
information sequence that can be used at the receiver to overcome the effects of noise and
interference encountered in the transmission on the signal through the channel. For example take k
bits of the information sequence and map that k bits to unique n bit sequence called code word.
3. Channel: The communication channel is the physical medium that is used for transmitting signals
from transmitter to receiver. In wireless system, this channel consists of atmosphere, for traditional
telephony, this channel is wired, there are optical channels, under water acoustic channels etc.We
further discriminate this channels on the basis of their property and characteristics, like AWGN
channel etc.
4. Channel Decoder: This sequence of numbers then passed through the channel decoder which
attempts to reconstruct the original information sequence from the knowledge of the code used by
the channel encoder and the redundancy contained in the received data
5. Source Encoder: In digital communication we convert the signal from source into digital signal
as mentioned above. The point to remember is we should like to use as few binary digits as possible
to represent the signal. In such a way this efficient representation of the source output results in little
or no redundancy. This sequence of binary digits is called information sequence
6. Source Decoder: At the end, if an analog signal is desired then source decoder tries to decode the
sequence from the knowledge of the encoding algorithm. And which results in the approximate
replica of the input at the transmitter end.
Advantage of digital communication
1. Digital communication can be done over large distances though internet and other things.
2. Digital communication gives facilities like video conferencing which save a lot of time,
money and effort.
3. It is easy to mix signals and data using digital techniques.
4. The digital communication is fast, easier and cheaper.
5. It can be tolerated the noise interference.
6. It can be detect and correct error easily because of channel coding.
7. Used in military application.
8. It has excellent processing techniques are available for digital signals such as data
compression, image processing, channel coding and equalization etc.

Limitation of digital communication

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).
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 sine waves but while operating on device we use square pulses.
1.2 Comparison Analog and Digital Communication

PARAMETERS ANALOG COMMUNICATION DIGITAL COMMUNICATION

Definiton Analog Communication is the Digital Communication is the

technology which uses Analog technology which uses digital signal
signal for the transmission of for the transmission of information.
information.

Noise and Get affected by Noise Immune from Noise and Distortion
Distortion

Error Probability Error Probability is high due to Error Probability is low

parallax.

Hardware Hardware is complicated and less Hardware is flexible and less

flexible than digital system. complicated than Analog system.

Cost Low Cost High Cost

Bandwidth Low bandwidth requirement High bandwidth Requirement

Requirement

Power High power is required Low Power Requirement

Requirement

Portability Less Portable as the components are More portable due to compact
heavy equipments.

Modulation Used Amplitude and Angle Modulation Pulse coded Modulation or PCM,
DPCM etc.

Representation of Analog signal can be represented by Digital signal is represented by

Signal sine wave. square wave.

Signal Values Consists of continuous values Consists of discrete values

Example of Signal Analog signal comprises of voice, Digital signals are used in
sound etc. computers
1.3 Principles of Digital Communication
When we enter data into the computer via keyboard, each keyed element is encoded by the
electronics within the keyboard into an equivalent binary coded pattern, using one of the standard
coding schemes that are used for the interchange of information. To represent all characters of the
keyboard, a unique pattern of 7 or 8 bits in size is used. The use of 7 bits means that 128 different
elements can be represented, while 8 bits can represent 256 elements. A similar procedure is
followed at the receiver that decodes every received binary pattern into the corresponding character.
Data transmission refers to the movement of data in form of bits between two or more digital
devices.This transfer of data takes place via some form of transmission media (for example, coaxial
cable, fiber optics etc.)

Types of transmission

Parallel transmission
Within a computing or communication device, the distances between different subunits are too short.
Thus, it is normal practice to transfer data between subunits using a separate wire to carry each bit of
data. There are multiple wires connecting each sub-unit and data is exchanged using a parallel
transfer mode. This mode of operation results in minimal delays in transferring each word.
• In parallel transmission, all the bits of data are transmitted simultaneously on separate
communication lines.
• In order to transmit n bits, n wires or lines are used. Thus each bit has its own line.
• All n bits of one group are transmitted with each clock pulse from one device to another i.e.
multiple bits are sent with each clock pulse.
• Parallel transmission is used for short distance communication.
• As shown in the fig, eight separate wires are used to transmit 8 bit data from sender to receiver.
Advantage of parallel transmission
It is speedy way of transmitting data as multiple bits are transmitted simultaneously with a single
clock pulse.
Disadvantage of parallel transmission
It is costly method of data transmission as it requires n lines to transmit n bits at the same time.

Serial Transmission
When transferring data between two physically separate devices, especially if the separation is more
than a few kilometers, for reasons of cost, it is more economical to use a single pair of lines. Data is
transmitted as a single bit at a time using a fixed time interval for each bit. This mode of
transmission is known as bit-serial transmission.
• In serial transmission, the various bits of data are transmitted serially one after the other.
• It requires only one communication line rather than n lines to transmit data from sender to receiver.
• Thus all the bits of data are transmitted on single line in serial fashion.
• In serial transmission, only single bit is sent with each clock pulse.
• As shown in fig., suppose an 8-bit data 11001010 is to be sent from source to destination. Then
least significant bit (LSB) i,e. 0 will be transmitted first followed by other bits. The most significant
bit (MSB) i.e. 1 will be transmitted in the end via single communication line.
• The internal circuitry of computer transmits data in parallel fashion. So in order to change this
parallel data into serial data, conversion devices are used.
• These conversion devices convert the parallel data into serial data at the sender side so that it can
be transmitted over single line.
• On receiver side, serial data received is again converted to parallel form so that the interval
circuitry of computer can accept it
• Serial transmission is used for long distance communication.
Advantage of Serial transmission
Use of single communication line reduces the transmission line cost by the factor of n as compared
to parallel transmission.
Disadvantages of Serial transmission
1. Use of conversion devices at source and destination end may lead to increase in overall
transmission cost.
2. This method is slower as compared to parallel transmission as bits are transmitted serially one
after the other.
Types of Serial Transmission
There are two types of serial transmission-synchronous and asynchronous both these transmissions
use 'Bit synchronization'
Bit Synchronization is a function that is required to determine when the beginning and end of the
data transmission occurs.
Bit synchronization helps the receiving computer to know when data begin and end during a
transmission. Therefore bit synchronization provides timing control.
A) Asynchronous Transmission
• Asynchronous transmission sends only one character at a time where a character is either a letter of
the alphabet or number or control character i.e. it sends one byte of data at a time.
• Bit synchronization between two devices is made possible using start bit and stop bit.
• Start bit indicates the beginning of data i.e. alerts the receiver to the arrival of new group of bits. A
start bit usually 0 is added to the beginning of each byte.
• Stop bit indicates the end of data i.e. to let the receiver know that byte is finished, one or more
additional bits are appended to the end of the byte. These bits, usually 1s are called stop bits.
• Addition of start and stop increase the number of data bits. Hence more bandwidth is consumed in
asynchronous transmission.
• There is idle time between the transmissions of different data bytes. This idle time is also known as
Gap
• The gap or idle time can be of varying intervals. This mechanism is called Asynchronous, because
at byte level sender and receiver need not to be synchronized. But within each byte, receiver must be
synchronized with the incoming bit stream.
Application of Asynchronous Transmission
1. Asynchronous transmission is well suited for keyboard type-terminals and paper tape
devices. The advantage of this method is that it does not require any local storage at the
terminal or the computer as transmission takes place character by character.

2. Asynchronous transmission is best suited to Internet traffic in which information is transmitted

in short bursts. This type of transmission is used by modems.
Advantages of Asynchronous transmission
1. This method of data transmission is cheaper in cost as compared to synchronous e.g. If lines are
short, asynchronous transmission is better, because line cost would be low and idle time will not be
expensive.
2. In this approach each individual character is complete in itself, therefore if character is corrupted
during transmission, its successor and predecessor character will not be affected.
3. It is possible to transmit signals from sources having different bit rates.
4. The transmission can start as soon as data byte to be transmitted becomes available.
5. Moreover, this mode of data transmission in easy to implement.
Disadvantages of asynchronous transmission
1. This method is less efficient and slower than synchronous transmission due to the overhead of
extra bits and insertion of gaps into bit stream.
2. Successful transmission inevitably depends on the recognition of the start bits. These bits can be
missed or corrupted.
B) Synchronous Transmission
• Synchronous transmission does not use start and stop bits.
• In this method bit stream is combined into longer frames that may contain multiple bytes.
• There is no gap between the various bytes in the data stream.

• In the absence of start & stop bits, bit synchronization is established between sender & receiver by
'timing' the transmission of each bit.
• Since the various bytes are placed on the link without any gap, it is the responsibility of receiver to
separate the bit stream into bytes so as to reconstruct the original information.
• In order to receive the data error free, the receiver and sender operates at the same clock frequency.
Application of Synchronous transmission
• Synchronous transmission is used for high speed communication between computers.
Advantage of Synchronous transmission
1. This method is faster as compared to asynchronous as there are no extra bits (start bit & stop bit)
and also there is no gap between the individual data bytes.
Disadvantages of Synchronous transmission
1. It is costly as compared to asynchronous method. It requires local buffer storage at the two ends of
line to assemble blocks and it also requires accurately synchronized clocks at both ends. This leads
to increase in the cost.
2. The sender and receiver have to operate at the same clock frequency. This requires proper
synchronization which makes the system complicated.
Comparison between Serial and Parallel transmission

Comparison between Asynchronous and Synchronous.

 MODULE – IV
Multiplexing- Space Division Multiplexing-Frequency Division Multiplexing: Wave length Division
Multiplexing - Time Division multiplexing: Characteristics, Digital Carrier system, SONET/SDH-
Statistical time division multiplexing: Cable Modem - Code Division Multiplexing. Multiple Access–
CDMA.

1. Multiplexing
 Multiplexing to refer to the combination of information streams from multiple sources for
transmission over a shared medium.
 The aim is to share a scarce resource. For example, in telecommunications, several telephone
calls may be carried using one wire.

 each sender communicates with a single receiver

 all pairs share a single transmission medium
 multiplexor combines information from the senders for transmission in such a
way that the demultiplexer can separate the information for receivers

 There are four basic approaches to multiplexing that each have a set of variations and
implementations
1. Frequency Division Multiplexing (FDM)
2. Wavelength Division Multiplexing (WDM)
3. Time Division Multiplexing (TDM)
4. Code Division Multiplexing (CDM)

 TDM and FDM are widely used

 WDM is a form of FDM used for optical fiber
 CDM is a mathematical approach used in cell phone mechanisms

1.1 Frequency Division Multiplexing

 Frequency Division Multiplexing (FDM) Frequency-division multiplexing is a form of signal

multiplexing which involves assigning non-overlapping frequency ranges to different signals
or to each "user of a medium.
 FDM achieves the combining of several signals into one medium by sending signals in several
distinct frequency ranges over a single medium.
 Frequency division multiplexing involves translation of the speech signal from the frequency
band 300-3400 Hz to a higher frequency band. Each channel is translated to a different hand
and then all the channels are combined to form a frequency division multiplexed signal.

Lecture Note – Mr. Sarath V Sankaran, JCET

  In FDM, the speech channels are stacked at intervals of 4 kHz to provide a guard band
between adjacent channels.

 FDM can be applied when the bandwidth of a link (in hertz) is greater than the combined
bandwidths of the signals to be transmitted.

 A demultiplexer applies a set of filters that each extract a small range of frequencies near one
of the carrier frequencies

Advantage of FDM:

1. The senders can send signals continuously.

2. FDM support full duplex information flow
3. Works for analog signals too
4. Noise problem for analog communication has lesser effect
5. AM and FM radio broadcasting and Television broadcasting

Disadvantage of FDM:
1. Separate frequency for each possible communication
2. Inflexible, one channel idle and other one busy
3. The initial cost is high
4. A problem for one user can sometimes affect others
5. Each user requires a precise carrier frequency.

1.2 Wavelength Division Multiplexing (WDM)

 WDM is an analog multiplexing technique to combine optical signals.

 Wavelength-division multiplexing (WDM) is designed to use the high-data-rate capability of
fiber-optic cable.
 Also called Dense WDM (DWDM) to emphasize that many wavelengths of light can be
employed.
 The inputs and outputs of such multiplexing are wavelengths of light denoted by the Greek
letter λ, and informally called colors.

Lecture Note – Mr. Sarath V Sankaran, JCET

  Prisms form the basis of optical multiplexing and demultiplexing
o A multiplexer accepts beams of light of various wavelengths and uses a prism to
combine them into a single beam
o A demultiplexer uses a prism to separate the wavelengths.

Advantages of WDM

1) Speed: Works with low speed equipment

2) Transparency: WDM is transparent_ It does not depend on the protocol that has to be
transmitted.
3) Scalable: It is scalable. Instead of switching to a new technology, a new channel can easily be
added to existing channels.
4) Capacity Increment: It is easy for network providers to add additional capacity in a few days if
customers need it.

Disadvantages of WDM

1) Complexity: Complex transmitters and receivers.

2) Reliability and Cost: They must be wide-band, which means they are more expensive and possibly
less reliable.

1.3 Time Division Multiplexing

 Time-division multiplexing (TDM) is a digital process that allows several connections to

share the high bandwidth of a line. Instead of sharing a portion of the bandwidth as in FDM,
time is shared.
 Time Division Multiplexing is the process of dividing up one communication time slot into
smaller time slots.

Lecture Note – Mr. Sarath V Sankaran, JCET

  Time Division Multiplexing (TDM) system, a single path and carrier frequency is used.
 TDM is a digital technology.
 Each user is assigned a unique time slot for their operation.
 A central switch or multiplexer goes from one user to the next in a specific predictable
sequence and time.
 TDM system can be applied when the data rate capacity of the transmission medium is
greater than the data rate required by the sending and receiving devices.
 TDM is more efficient than FDM, in that it does not require guard bands and it operates
directly in digital form.
 In TDM, the transmission between the multiplexers is provided by a single high speed digital
transmission line.

Types of Time Division Multiplexing It can be categories into two types:

1) Synchronous TDM
2) Asynchronous TDM

1.3.1 Synchronous TDM

 Accepts input in a round-robin fashion

 Synchronous TDM works by the multiplexer giving exactly the-same amount of time to each
device connected to it.

 This time slice is allocated even if a device has nothing to transmit.

 The use of Synchronous TDM does not guarantee maximum line usage and efficiency.
 T-1 and ISDN telephone lines are common examples of synchronous time division
multiplexing.
 It is used for multiplexing digitized voice stream.

1.3.2 Asynchronous TDM

 Asynchronous Time-Division Multiplexing is a method of sending information that resembles

normal TDM, except that time slots are allocated as needed dynamically rather than pre-
assigned to specific transmitters.
 Asynchronous TDM is more intelligent and has better bandwidth efficiency than TDM.
 Good for low bandwidth lines
 Examples: used for LANs

Lecture Note – Mr. Sarath V Sankaran, JCET

 Parameter Synchronous TDM Asynchronous TDM / Statistical TDM
In Synchronous TDM data flow of In Statistical TDM slots are allotted
each input connection is divided into dynamically. i.e. input line is given slots
Working
units and each input occupies one in output frame if and only if it has data
output time slot. to send.
In Synchronous TDM no. of slots in
In Statistical TDM, No. of slots in each
No. of Slots each frame are equal to no. of input
frame are less than the no. of input lines.
lines.
Buffering is not done, frame is sent
Buffering is done and only those inputs
after a particular interval of time
Buffers are given slots in output frame whose
whether someone has data to send or
buffer contains data to send.
not.
Slots in Synchronous TDM carry data
only and there is no need of
Slots in Statistical TDM contain both
Addressing addressing. Synchronization and pre
data and address of the destination.
assigned relationships between input
and outputs that serve as an address.
Synchronization bits are used at the
Synchronization No synchronization bits are used
beginning of each frame.
The capacity of link is normally is less
Max. Bandwidth utilization if all
Capacity than the sum of the capacity of each
inputs have data to send.
channel.
In Synchronous TDM de-multiplexer
In Statistical TDM de-multiplexer at
at receiving end decomposes each
receiving end decomposes each frame
frame, discards framing bits and
Data Separation by checking local address of each data
extracts data unit in turn. This
unit. This extracted data unit from frame
extracted data unit from frame is
is then passed to destination device.
then passed to destination device.

Advantages of TDM

1) The user gets full bandwidth of the channel in a particular time slot.
2) For bursty signals such as voice or speech TDMA gives maximum utilization of the channel.
3) Most suitable technique for digital transmission.
4) It does not require precise carrier matching at both end of the links.
5) Can expand the number of users on a system at a low cost.

Lecture Note – Mr. Sarath V Sankaran, JCET

Disadvantages of TDM

1) It is not much suitable for continues signals.

2) Initial cost is high.
3) The noise problem for analog communication has greater effect.
4) Extra guard times are necessary.
5) Synchronization is necessary.

Difference between TDM and FDM

Time Division Multiplexing (TDM) Frequency Division Multiplexing(FDM)

Total available time is divided into several users
Total frequency bands are divided into several
users.
Transmission of two or more signals on the same A multiplex system for transmitting two or more
path, but at different times. signals over a common path by using a different
frequency band for each signal.
TDM imply partitioning the bandwidth of the
channel connecting two nodes into finite set of The signals multiplexed come from different
time slots sources/transmitters.
TDM is implemented using digital devices and FDM is implemented using analog devices and
circuits. circuits.
TDM system has relatively small inter-channel In FDM the non-linearites present in transmitter
crosstalk which can be arrived by ensuring and receiver circuits produce intermodulation
completely isolated and non-overlapping pulses and harmonic distortion.
indifferent time slots.

1.4 Code Division Multiplexing

 In code division multiplexing, every user can transmit over the entire frequency spectrum all the
time.
 CDM used in parts of the cellular telephone system and for some satellite communication
 The specific version of CDM used in cell phones is known as Code Division Multi-Access (CDMA)
 CDM is achieved through application of spread-spectrum techniques.
 Spread spectrum techniques can be classified into two categories:
1) Direct-Sequence
2) Frequency hopping techniques.
 Generally, CDM is a frequency hopping spread spectrum techniques which allows utilization of
full channel bandwidth for each user.
 CDM does not rely on physical properties such as frequency or time
 CDM relies on an interesting mathematical idea - “values from orthogonal vector spaces can be
combined and separated without interference”.

 Each sender is assigned a unique binary code Ci that is known as a chip sequence

Lecture Note – Mr. Sarath V Sankaran, JCET

 Chip sequences are selected to be orthogonal vectors (i.e., the dot product of any two chip
sequences is zero).
 At any point in time, each sender has a value to transmit, Vi
 The senders each multiply Ci x Vi and transmit the results
 The senders transmit at the same time
 and the values are added together
 To extract value Vi, a receiver multiplies the sum by Ci

Consider an example
 To keep the example easy to understand, use a chip sequence that is only two bits long and data
values that are four bits long.

 The first step consists of converting the binary values into vectors that use -1 to represent 0:

 If we think of the resulting values as a sequence of signal strengths to be transmitted at the same
time, the resulting signal will be the sum of the two signals

 A receiver treats the sequence as a vector

 Receiver 1 computes:

 Interpreting the result as a sequence produces: (2 -2 2 -2)

 which becomes the binary value: (1 0 1 0) note that 1010 is the correct value of V1
 Receiver 2 will extract V2 from the same transmission

3. Digital Carrier System

 A carrier system is a telecommunications system that transmits information, such as the voice
signals of a telephone call and the video signals of television etc.
 Carrier systems typically transmit multiple channels of communication simultaneously over the
shared medium.

Lecture Note – Mr. Sarath V Sankaran, JCET

 A digital carrier system is a communication system that uses digital pulse rather than analog
signals to encode information.
 Following are the digital carrier standards:
1. T-carrier: T-carrier system is entirely digital, using pulse code modulation (PCM) and
time-division multiplexing (TDM).
The system uses four wires and provides duplex capability (two wires for receiving and
two for sending at the same time).
T1 lines should be used for critical, high bandwidth applications. T1 lines are best when
the sites being connected are close together (otherwise the cost is prohibitive).
2. E-carrier: E-carrier (European carrier) A hierarchy of standards for digital transmission,
E-carrier is based on the original North America T-carrier digital carrier system, although
the specifics are quite different with respect to signalling rates, framing convention, line
coding technique, and PCM companding technique (A-law rather than μ -law). In many
respects E-carrier is a considerable improvement over T-carrier.
For example, E-1 supports 30 DS-0 payload channels, compared with T1 at 24 channels,
and the higher E-carrier levels build on that difference.
E-carrier also supports non-intrusive signalling and control through two channels
reserved for such purposes.
As a result, E-carrier supports clear channel communication of a full 64Kbps per DS-0,
compared to 56kbps data with T-carrier. The DS-0 (Digital Signal level Zero) is the
fundamental building block of E-carrier, as it is with T-carrier and J-carrier, the Japanese
version.
Through Time Division Multiplexing (TDM), E-carrier interleaves DS-0 channels at
various signalling rates to create the services that comprise the European digital
hierarchy.
3. SONET/SDH: The synchronous optical network (SONET) is a TDM standard for
transmission over optical fibers in the terrestrial United States.
 Synchronous optical network (SONET) is a standard for optical telecommunications
transport formulated by the Exchange Carriers Standards Association (ECSA) for the
American National Standards Institute (ANSI), which sets industry standards in the
U.S. for telecommunications and other industries.
 A similar standard, Synchronous Digital Hierarchy (SDH), is used in Europe by the
International Telecommunication Union Telecommunication Standardization Sector
(ITU-T). SONET equipment is generally used in North America, and SDH equipment is
generally accepted everywhere else in the world.
 Both SONET and SDH are standards for a synchronous, fiber optic transport system.
SONET, or synchronous optical network, is the North American standard.
 SDH is the similar standard used in Europe and rest of the world.
 SONET works at layer 1 (Physical).
 SONET/SDH's strength is in transporting delay-sensitive voice and video, but is also
used for high speed data transport.
 SONET incorporates a continuous series of frames.
 SONET is used for high-speed data transmission. Telephone companies have
traditionally used a lot of SONET but this may be giving way to other high-speed
transmission services.

Lecture Note – Mr. Sarath V Sankaran, JCET

  SONET uses a basic transmission rate of synchronous transport signal–level 1 (STS–1)
that is equivalent to 51.84 Mbps.

 The frame format of the STS–1 signal is shown in below Figure.

 In general, the frame can be divided into two main areas: transport overhead and the
synchronous payload envelope (SPE).
 Transport overhead is composed of section overhead and line overhead.
 The main function of the section layer is to properly format the SONET frames, and to
convert the electrical signals to optical signals.
 Line overhead originates or terminates one or more sections of a line signal. The Line-
Terminating Equipment (LTE) does the synchronization and multiplexing of
information on SONET frames.
 The synchronous payload envelope can also be divided into two parts: the STS path
overhead (POH) and the payload.
 Path-Terminating Equipment (PTE) interfaces non-SONET equipment to the SONET
network. At this layer, the payload is mapped and demapped into the SONET frame.
 STS–1 is a specific sequence of 810 bytes (6,480 bits), which includes various
overhead bytes and an envelope capacity for transporting payloads. It can be depicted
as a 90-column by 9-row structure.
 The order of transmission of bytes is row-by-row from top to bottom and from left to
right (most significant bit first).
 With a frame length of 125 μs (8,000 frames per second), STS–1 has a bit rate of
51.840 Mbps.
(9) x (90 bytes/frame) x (8 bits/byte) x (8,000 frames/s) = 51,840,000 bps = 51.840 Mbps

Lecture Note – Mr. Sarath V Sankaran, JCET

 4. Cable Modem
 A cable modem is a peripheral device used to connect to the Internet.
 It operates over coax cable TV lines and provides high-speed Internet access.
 Since cable modems offer an always-on connection and fast data transfer rates, they are
considered broadband devices.

 To support data transfer to and from a cable modem, a cable TV provider dedicates two
channels, one for transmission in each direction.
 Each channel is shared by a number of subscribers, and so some scheme is needed for
allocating capacity on each channel for transmission.
 Typically, a form of statistical TDM is used, as illustrated in figure

 In the downstream direction a cable scheduler delivers data in the form of small packets.
Because the channel is shared by a number of subscribers, if more than one subscriber is
active, each subscriber gets only a fraction of the downstream capacity.
 An individual cable modem subscriber may experience access speeds from 500 kbps to 1.5
Mbps or more, depending on the network architecture and traffic load.
 When a subscriber has data to transmit, it must first request time slots on the shared
upstream channel.
 Each subscriber is given dedicated time slots for this request purpose.
 The headend scheduler responds to a request packet by sending back an assignment of future
time slots to be used by this subscriber.
 Thus a number of subscribers can share the same upstream channel without conflict.

Lecture Note – Mr. Sarath V Sankaran, JCET

1. What is CDMA? Explain.
2. Explain Space Division Multiplexing.
3. Differentiate between Synchronous TDM and Statistical TDM. Why is a statistical time division
multiplexer more efficient than a synchronous time division multiplexer?
4. Explain frequency division multiplexing. How is interference avoided by using FDM?
5. Discuss Synchronous Optical NETwork (SONET).
6. What are the advantages of using multiplexing in data communication? How does a synchronised
time division multiplexer stay synchronized with de-multiplexer on receiving end?
7. What type of multiplexing is preferred in optical fibre communication? Justify your answer
8. Explain the modulation technique used in Asymmetric Digital Subscriber Line (ADSL) and cable
modems
9. With suitable example explain the working principle of Code division multiplexing for CDMA
technology.
10. Explain the frame format of Synchronous Optical Network(SONET) for the version SDH.
11. How Time division Multiplexing (TDM) handle disparity in the input data rate, if data rate of all
input lines are not same?
12. Which of the multiplexing technique is suitable for fiber-optics links? Explain with reasoning.
13. How upstream and downstream data transfer is done in cable modem.
14. Explain the process of statistical time division multiplexing.
15. Explain the necessity of pulse stuffing in synchronous time division multiplexing.
16. Explain how Statistical TDM utilizes channel bandwidth better than Synchronous TDM.
17. How interference is avoided in frequency division multiplexing? Explain with suitable figures.
18. Explain SONET/SDH frame format.
19. Explain Frequency Division Multiplexing process.
20. Discuss Digital Carrier Systems.
21. Discuss wave length division multiplexing

Lecture Note – Mr. Sarath V Sankaran, JCET

 DIGITAL PULSE MODULATION

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

1. PULSE CODE MODULATION(PCM)

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

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

11
Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists of Sampling,

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

➢ Low Pass Filter

This filter eliminates the high frequency components present in the input analog signal which
is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.
➢ Sampler
This is the technique which helps to collect the sample data at instantaneous values of message
signal, so as to reconstruct the original signal. The sampling rate must be greater than twice the
highest frequency component W of the message signal, in accordance with the sampling
theorem.
➢ Quantizer
Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer reduces the redundant bits and compresses the value.

12
 ➢ Encoder

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

➢ Regenerative repeater:

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

Regenerative repeaters have three functions.

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

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

Timing circuits provides periodic pulse trains.

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

➢ Decoder

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

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

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

Fig.PCM waveform

Bit rate and bandwidth requirements of PCM :

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

Important Relations
• Quantization Noise (𝑁𝑞)=Δ2/2
• Signal to Noise ratio
(𝑆𝑄𝑁𝑅)=32.22𝑛 𝑜𝑟 𝑆𝑄𝑁𝑅 𝑖𝑛 𝑑𝐵=1.76+6.02𝑛≅(1.8+6𝑛)𝑑𝐵
• 𝐵𝑖𝑡 𝑟𝑎𝑡𝑒=𝑁𝑜.𝑜𝑓 𝑏𝑖𝑡𝑠 𝑝𝑒𝑟 𝑠𝑎𝑚𝑝𝑙𝑒×𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑟𝑎𝑡𝑒=𝑛𝑓𝑠
• Bandwidth for PCM signal =n.fm
Where,
n – No. of bits in PCM code
Fm – signal bandwidth
fs – sampling rate

14
 ➢ COMPANDING IN PCM SYSTEMS

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

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

Fig Companding curves for PCM

Companding is used to maintain constant Signal to Noise Ratio throughout dynamic quantization
ratio

Fig. Non Uniform Quantization

24
 There are two types of Companding techniques. They are –

1.A-law Companding Technique

i. Uniform quantization is achieved at A = 1, where the characteristic curve is linear
and no compression is done.
ii. A-law has mid-rise at the origin. Hence, it contains a non-zero value.
iii. A-law Companding is used for PCM telephone systems.

Y= A│x│ ; where 0≤ x ≤ 1/A 1+ln(A)

= 1+ln A│x│ ; 1/A≤ x ≤ 1 1+ln(A)

practically A=87.56

if A=1 we get uniform quantization

2.µ-law Companding Technique

i. Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and
no compression is done.
ii. µ-law has mid-tread at the origin. Hence, it contains a zero value.
iii. µ-law companding is used for speech and music signals.

Y= ±ln(1+µ│x│) ;│x│≤1 ln(1+µ)

Practically µ value is 256

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

25
2. DIFFERENTIAL PCM (DPCM)

DPCM Transmitter

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

The signals at each point are named as −

i. x(nTs) is the sampled input
ii. x^(nTs) is the predicted sample
iii. e(nTs) is the difference of sampled input and predicted output, often called as
prediction error
iv. v(nTs) is the quantized output
v. u(nTs) is the predictor input which is actually the summer output of
the predictor output and the quantizer output

The predictor produces the assumed samples from the previous outputs of the transmitter
circuit. The input to this predictor is the quantized versions of the input signal x(nTs).
Quantizer Output is represented as −
v(nTs)=Q[e(nTs)]
=e(nTs)+q(nTs)
Where,
q (nTs) is the quantization error

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

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

DPCM Receiver

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

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

Advantages of DPCM

1) Bandwidth Requirement Of DPCM Is Less Compared To PCM

2) Quantization Error Is Reduced Because Of Prediction Filter.

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

33
3. DELTA MODULATION

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

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

Fig. Block diagram of delta modulator and demodulator

Features of Delta Modulation

Following are some of the features of delta modulation.
• An over-sampled input is taken to make full use of the signal correlation.
• The quantization design is simple.
• The input sequence is much higher than the Nyquist rate.
• The quality is moderate.
• The design of the modulator and the demodulator is simple.
• The stair-case approximation of output waveform.
• The step-size is very small, i.e., Δ (delta).
• The bit rate can be decided by the user.
• This involves simpler implementation.

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

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

From the above diagram, we have the notations as −

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

Using these notations, now we shall try to figure out the process of delta modulation.
ep(nTs)=x(nTs)−xˆ(nTs) ------------------(1)
=x(nTs)−u([n−1]Ts)
=x(nTs)−[xˆ[[n−1]Ts]+v[[n−1]Ts]] ---------------(2)
Further,
v(nTs)=eq(nTs)=S∑.[ep(nTs)]------------------ (3)
u(nTs)=xˆ(nTs)+eq(nTs)
Where,
xˆ(nTs) = the previous value of the delay circuit

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

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

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

Delta Demodulator
The delta demodulator comprises of a low pass filter, a summer, and a delay circuit. The
predictor circuit is eliminated here and hence no assumed input is given to the demodulator.
Following is the diagram for delta demodulator.

From the above diagram, we have the notations as −

• v^(nTs) is the input sample

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

Advantages of DM Over DPCM

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

Advantages of Delta Modulation

• In Delta modulation electronic circuit requirement for modulation at transmitter and for
demodulation at receiver is substantially simpler compare to PCM.
• In delta modulation, amplitude of speech signal does not exceed maximum sinusoidal
amplitude.
• Signaling rate and bandwidth of DPCM or delta modulation is less than PCM technique.

Disadvantages of Delta Modulation

• If changes in signal is less than the step size, then modulator no longer follow signal.
Thus produces train of alternating positive and negative pulses.
• Modulator overloads when slope of signal is too high.
• High bit rate.
• It requires predictor circuit and hence it is very complex.
• Its practical usage is limited.

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

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

Fig.1: Quantization Errors in Delta Modulation

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

2. Granular noise

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

In digital modulation, we have come across certain problem of determining the step-size, which
influences the quality of the output wave.
A larger step-size is needed in the steep slope of modulating signal and a smaller step size is
needed where the message has a small slope. The minute details get missed in the process. So, it
would be better if we can control the adjustment of step-size, according to our requirement in
order to obtain the sampling in a desired fashion. This is the concept of Adaptive Delta
Modulation.
The performance of a delta modulator can be improved significantly by making the step size
of the modulator assume a time-varying form. In particular, during a steep segment
of the input signal the step size is increased. Conversely, when the input signal is
varying slowly, the step size is reduced.
In this way, the size is adapted to the level of the input signal. The resulting
method is called adaptive delta modulation (ADM).
There are several types of ADM, depending on the type of scheme used for adjusting
the step size. In this ADM, a discrete set of values is provided for the step size.

39
 A large step size was required when sampling those parts of the input waveform of steep slope. But a
large step size worsened the granularity of the sampled signal when the waveform being sampled was
changing slowly. A small step size is preferred in regions where the message has a small slope. This
suggests the need for a controllable step size - the control being sensitive to the slope of the sampled
signal.

The Implementation of ADM Modulator

The audio signal will pass through a low-pass filter, which can remove all the unwanted signal and only
obtain the audio signal. The input signals of the comparator are the audio signal and triangle wave signal, and
40
then the output of the comparator is the square wave signal. The D type flip flop is used as sampling

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

Fig. Waveforms of ADM

41
 COMPARISON OF PCM AND DM SYSTEMS

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

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

42
 Noise in PCM and DM systems

Signal to Quantization Noise ratio in PCM:

The signal to quantization noise ratio is given as:

The number of quantization value is equal to:

Putting this value in eqn (6), we get

Substitute this value in eq, we get

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

43
COMPARISON OF PCM AND DM SYSTEMS

S.No Parameter Pulse Code Modulation Delta Modulation

1 Number of bits Very high, It can use 4,8 It uses one bit per
or 16 bits per sample sample
2 Quantization levels It depends on number of One bit quantizer is
bits q=2v used
3 Type of error Quantization error Slope overload error
and granular noise
4 Signal to Noise Ratio Very high Moderate
5 Bandwidth Highest bandwidth is Lowest bandwidth is
needed since the number enough
of bits are high
6 Complexity Complex system to Simple to
implement implement

44
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W h a t i s A n a l o g to D i g i t a l C o n ve r t e r & I t s
Wo r k i n g
Almost every environmental measurable parameter is in analog form like temperature, sound, p
light, etc. Consider a temperature monitoring system wherein acquiring, analyzing, and process
temperature data from sensors is not possible with digital computers and processors. Therefore
system needs an intermediate device to convert the analog temperature data into digital data in
communicate with digital processors like microcontrollers and microprocessors. Analog to Digita
Converter (ADC) is an electronic integrated circuit used to convert the analog signals such as v
digital or binary form consisting of 1s and 0s. Most of the ADCs take a voltage input as 0 to 10V
+5V, etc., and correspondingly produces digital output as some sort of a binary number.

What is Analog to Digital Converter?

A converter that is used to change the analog signal to digital is known as an analog to digital c
ADC converter. This converter is one kind of integrated circuit or IC that converts the signal dire

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continuous form to discrete form. This converter can be expressed in A/D, ADC, A to D. The inv
function of DAC is nothing but ADC. The analog to digital converter symbol is shown below.

The process of converting an analog signal to digital can be done in several ways. There are di
types of ADC chips available in the market from different manufacturers like the ADC08xx serie
simple ADC can be designed with the help of discrete components.

The main features of ADC are sample rate and bit resolution.

• The sample rate of an ADC is nothing but how fast an ADC can convert the signal from analo
• Bit resolution is nothing but how much accuracy can an analog to digital converter can conve
signal from analog to digital.

Analog to Digital Converter

One of the major benefits of ADC converter is the high data acquisition rate even at multiple
With the invention of a wide variety of ADC integrated circuits (IC’s), data acquisition from vario

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becomes more accurate and faster. Dynamic characteristics of the high-performance ADCs a
measurement repeatability, low power consumption, precise throughput, high linearity, excellen
Noise Ratio (SNR), and so on.

A variety of applications of the ADCs are measurement and control systems, industrial instr
communication systems, and all other sensory-based systems. Classification of ADCs based
like performance, bit rates, power, cost, etc.

ADC Block Diagram

The block diagram of ADC is shown below which includes sample, hold, quantize, and encoder
process of ADC can be done like the following.

First, the analog signal is applied to the first block namely a sample wherever it can be sampled
exact sampling frequency. The amplitude value of the sample like an analog value can be main
well as held within the second block like Hold. The hold sample can be quantized into discrete v
through the third block like quantize. Finally, the last block like encoder changes the discrete am
a binary number.

In ADC, the conversion of the signal from analog to digital can be explained through the above
diagram.

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Sample

In the sample block, the analog signal can be sampled at an exact interval of time. The samples
in continuous amplitude and hold real value however they are discrete with respect to time. Wh
converting the signal, the sampling frequency plays an essential role. So it can be maintained a
rate. Based on the system requirement, the sampling rate can be fixed.

Hold

In ADC, HOLD is the second block and it doesn’t have any function because it simply holds the
amplitude till the next sample is taken. So the value of hold doesn’t change until the next sampl

Quantize

In ADC, this is the third block which is mainly used for quantization. The main function of this is
the amplitude from continuous (analog) into discrete. The value of continuous amplitude within
moves throughout quantize block to turn into discrete in amplitude. Now, the signal will be in dig
because it includes discrete amplitude as well as time.

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Encoder

The final block in ADC is an encoder that converts the signal from digital form to binary. We kno
digital device works by using binary signals. So it is required to change the signal from digital to
the help of an encoder. So this is the entire method to change an analog signal to digital using a
The time taken for the entire conversion can be done within a microsecond.

Analog to Digital Conversion Process

There are many methods to convert analog signals to digital signals. These converters
applications as an intermediate device to convert the signals from analog to digital form, displa
LCD through a microcontroller. The objective of an A/D converter is to determine the output
corresponding to an analog signal. Now we are going to see an ADC of 0804. It is an 8-bit conv
5V power supply. It can take only one analog signal as input.

Analog to Digital Converter for Signal

The digital output varies from 0-255. ADC needs a clock to operate. The time taken to conver

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to digital value depends on the clock source. An external clock can be given to CLK IN pin no.4
RC circuit is connected between the clock IN and clock R pins to use the internal clock. Pin2
pin – High to low pulse brings the data from the internal register to the output pins after convers
a Write – Low to high pulse is given to the external clock. Pin11 to 18 are data pins from MSB to

Analog to Digital Converter samples the analog signal on each falling or rising edge of the sam
each cycle, the ADC gets the analog signal, measures it, and converts it into a digital value
converts the output data into a series of digital values by approximates the signal with fixed pre

In ADCs, two factors determine the accuracy of the digital value that captures the original an
These are quantization level or bit rate and sampling rate. The below figure depicts how anal
conversion takes place. Bit rate decides the resolution of digitized output and you can observe
figure where 3-bit ADC is used for converting the analog signal.

Analog to Digital Conversion Process

Assume that one-volt signal has to be converted from digital by using 3-bit ADC as sh
Therefore, a total of 2^3=8 divisions are available for producing 1V output. This results 1/8
called as minimum change or quantization level represented for each division as 000 for 0V, 00
and likewise upto 111 for 1V. If we increase the bit rates like 6, 8, 12, 14, 16, etc. we will
precision of the signal. Thus, bit rate or quantization gives the smallest output change in the a
value that results from a change in the digital representation.

Suppose if the signal is about 0-5V and we have used 8-bit ADC then the binary output of 5V

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for 3V it is 133 as shown below.

There is an absolute chance of misrepresenting the input signal on the output side if it is sa
different frequency than the desired one. Therefore, another important consideration of the
sampling rate. The Nyquist theorem states that the acquired signal reconstruction introduce
unless it is sampled at (minimum) twice the rate of the largest frequency content of the signal
observe in the diagram. But this rate is 5-10 times the maximum frequency of the signal in prac

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Sampling Rate of Analog to Digital Converter

Factors
The ADC performance can be evaluated through its performance based on different factors. Fro
following two main factors are explained below.

SNR (Signal-to-Noise Ratio)

The SNR reflects the average number of bits without noise in any particular sample.

Bandwidth

The bandwidth of an ADC can be determined by estimating the sampling rate. The analog sour
sampled per second to produce discrete values.

Types of Analog to Digital Converters

ADC is available in different types and some of the types of analog to digital converters include

• Dual Slope A/D Converter

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• Flash A/D Converter

• Successive Approximation A/D Converter
• Semi-flash ADC
• Sigma-Delta ADC
• Pipelined ADC

Dual Slope A/D Converter

In this type of ADC converter, comparison voltage is generated by using an integrator circ
formed by a resistor, capacitor, and operational amplifier combination. By the set value o
integrator generates a sawtooth waveform on its output from zero to the value Vref. When th
waveform is started correspondingly counter starts counting from 0 to 2^n-1 where n is the nu
of ADC.

Dual Slope Analog to Digital Converter

When the input voltage Vin equal to the voltage of the waveform, then the control circuit capture
counter value which is the digital value of the corresponding analog input value. This Dual slope
relatively medium cost and slow speed device.

Flash A/D Converter

This ADC converter IC is also called parallel ADC, which is the most widely used efficient ADC
its speed. This flash analog to digital converter circuit consists of a series of comparators whe
compares the input signal with a unique reference voltage. At each comparator, the output w
state when the analog input voltage exceeds the reference voltage. This output is further g
priority encoder for generating binary code based on higher-order input activity by ignoring
inputs. This flash type is a high-cost and high-speed device.

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Flash A/D Converter

Successive Approximation A/D Converter

The SAR ADC a most modern ADC IC and much faster than dual slope and flash ADCs sin
digital logic that converges the analog input voltage to the closest value. This circuit co
comparator, output latches, successive approximation register (SAR), and D/A converter.

Successive Approximation A/D Converter

At the start, SAR is reset and as the LOW to HIGH transition is introduced, the MSB of the S
Then this output is given to the D/A converter that produces an analog equivalent of the MSB
compared with the analog input Vin. If comparator output is LOW, then MSB will be cleared b
otherwise, the MSB will be set to the next position. This process continues till all the bits are tri
Q0, the SAR makes the parallel output lines to contain valid data.

Semi-flash ADC

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These types of analog to digital converts mainly works approximately their limitation size throug
separate flash converters, where each converter resolution is half of the bits for the semi-flush d
capacity of a single flash converter is, it handles the MSBs (most significant bits) whereas the o
handles the LSB (least significant bits).

Sigma-Delta ADC

Sigma Delta ADC (ΣΔ) is fairly a recent design. These are extremely slow as compared to othe
designs however they offer the maximum resolution for all kinds of ADC. Thus, they are extrem
compatible with high-fidelity based audio applications, however, they are normally not utilizable
high BW (bandwidth) is required.

Pipelined ADC

Pipelined ADCs are also known as sub ranging quantizers which are related in concept to succ
approximations, even though more sophisticated. While successive approximations grow throug
step by going to the next MSB, this ADC uses the following process.

• It is used for a coarse conversion. After that, it evaluates that change toward the input signal.
• This converter acts as a better conversion by allowing for a temporary conversion with a rang
• Usually, pipelined designs offer a center ground among SARs as well as flash analog to digit
converters by balancing its size, speed & high resolution.

Analog to Digital Converter Examples

The examples of analog to digital converter are discussed below.

ADC0808

ADC0808 is a converter that has 8 analog inputs and 8 digital outputs. ADC0808 allows us to
to 8 different transducers using only a single chip. This eliminates the need for external zero a
adjustments.

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ADC0808 IC

ADC0808 is a monolithic CMOS device, offers high speed, high accuracy, minimal t
dependence, excellent long-term accuracy and repeatability and consumes minimal power. The
make this device ideally suited to applications from process and machine control to con
automotive applications. The pin diagram of ADC0808 is shown in the figure below:

Features

The main features of ADC0808 include the following.

• Easy interface to all microprocessors

• No zero or full-scale adjust required
• 8-channel multiplexer with address logic
• 0V to 5V input range with single 5V power supply
• Outputs meet TTL voltage level specifications
• Carrier chip package with 28-pin

Specifications

The specifications of ADC0808 include the following.

• Resolution: 8 Bits

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• Total Unadjusted Error: ±½ LSB and ±1 LSB

• Single Supply: 5 VDC
• Low Power: 15 mW
• Conversion Time: 100 μs

Generally, the ADC0808 input which is to be changed over to digital form can be selected by
address lines A, B, C which are pins 23, 24, and 25. The step size is chosen dependent u
reference value. Step size is the change in analog input to cause a unit change in the outp
ADC0808 needs an external clock to operate, unlike ADC0804 which has an internal clock.

The continuous 8-bit digital output corresponding to the instantaneous value of analog inpu
extreme level of the input voltage must be reduced proportionally to +5V.

The ADC 0808 IC requires a clock signal of typically 550 kHz, ADC0808 is used to convert t
digital form required for the microcontroller.

Application of ADC0808
The ADC0808 has got many applications; here we have given some application on ADC:

From the below circuit the clock, start, and EOC pins are connected to a microcontroller. G
have 8 inputs; here we are using only 4 inputs for the operation.

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ADC0808 Circuit

• The LM35 temperature sensor is using which is connected to the first 4 inputs of the analog t
converter IC. The sensor has got 3 pins i.e., VCC, GND, and output pins when the sensor he
voltage at output increases.
• The address lines A, B, C are connected to the microcontroller for the commands. In this, the
follows the low to high operation.
• When the start pin is held high no conversion begins, but when the start pin is low the conver
start within 8 clock periods.
• At the point when the conversion is completed the EOC pin goes low to indicate the finish of
and data ready to be picked up.
• The output enables (OE) is then raised high. This enables the TRI-STATE outputs, allowing t
be read.

ADC0804

We already know that analog-to-digital (ADCs) converters are the most widely used devices for
securing to translate the analog signals to digital numbers so the microcontroller can read t
There are many ADC converters like ADC0801, ADC0802, ADC0803, ADC0804, and ADC
article, we are going to discuss the ADC0804 converter.

ADC0804

ADC0804 is a very commonly used 8-bit analog to digital converter. It works with 0V to 5V a

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voltage. It has single analog input and 8-digital outputs. Conversion time is another major facto
an ADC, in ADC0804 conversion time varies depending on the clocking signals applied to CLK
IN pins, but it cannot be faster than 110 μs.

Pin Description of ADC804

Pin 1: It is a chip select pin and activates ADC, active low

Pin 2: It is an input pin; high to low pulse brings the data from internal registers to the output pin
conversion

Pin 3: It is an input pin; low to high pulse is given to start the conversion

Pin 4: It is a clock input pin, to give the external clock

Pin 5: It is an output pin, goes low when the conversion is complete

Pin 6: Analog non-inverting input

Pin 7: Analog inverting input, it’s normally ground

Pin 8: Ground (0V)

Pin 9: It is an input pin, sets the reference voltage for analog input

Pin 10: Ground (0V)

Pin 11 – Pin 18: It is an 8-bit digital output pin

Pin 19: Is used with Clock IN pin when internal clock source is used

Pin 20: Supply voltage; 5V

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Features of ADC0804
The main features of ADC0804 include the following.

• 0V to 5V analog input voltage range with single 5V supply

• Compatible with microcontrollers, access time is 135 ns
• Easy interface to all microprocessors
• Logic inputs and outputs meet both MOS and TTL voltage level specifications
• Works with 2.5V (LM336) voltage reference
• On-chip clock generator
• No zero adjust required
• 0.3[Prime] standard width 20-pin DIP package
• Operates ratio metrically or with 5 VDC, 2.5 VDC, or analog span adjusted voltage reference
• Differential analog voltage inputs

It is an 8-bit converter with a 5V power supply. It can take only one analog signal as input.
output varies from 0-255. ADC needs a clock to operate. The time taken to convert the anal
value depends on the clock source. An external clock can be given to CLK IN. Pin2 is the inpu
to low pulse brings the data from the internal register to the output pins after conversion. Pin3
Low to high pulse is given to the external clock.

Application
From the simple circuit, pin 1 of ADC is connected to GND where pin4 is connected to GND
capacitor; pin 2, 3, and 5 of ADC are connected to 13, 14, and 15 pins of the microcontroller. P
are shorted and connected to GND, 19 pins of ADC is to 4th pin through resistor 10k. Pin 11 to
are connected to 1 to 8 pins of the microcontroller which belongs to port1.

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ADC0804 Circuit

When the logic high is applied to CS and RD, input has been clocked through the 8-bit s
completing the specific absorption rate (SAR) search, on the next clock pulse; the dig
transferred to the tri-state output. The output of the interrupt is inverted to provide an INTR o
high during conversion and low when the conversion is completed. When a low is at both CS
output is applied to the DB0 through DB7 outputs and the interrupt is reset. When either the
inputs return to a high state, the DB0 through DB7 outputs are disabled (returned to the high
state). Thus depending on the logic the voltage various from 0 to 5V which is transformed to a
of 8-bit resolution, being fed as an input to the microcontroller port 1.

ADC0804 Component Used Projects

• Underground Cable Fault Distance Locator

ADC0808 Component Used Projects

• Scada (Supervisory Control & Data Acquisition) For Remote Industrial Plant

ADC Testing
The testing of analog to digital converter mainly needs an analog input source as well as hardw
transmit the control signals as well as to capture digital data o/p. Some kinds of ADCs need a p
reference signal source. The ADC can be tested by using the following key parameters

• DC Offset Error

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• Power Dissipation
• DC Gain Error
• Spurious Free Dynamic Range
• SNR (Signal to Noise Ratio)
• INL or Integral Nonlinearity
• DNL or Differential Nonlinearity
• THD or Total Harmonic Distortion

The testing of ADCs or Analog-to-digital converters is mainly done for several reasons. Apart fr
reason, the society of IEEE Instrumentation & Measurement, the waveform generation & analys
committee was developed the IEEE Standard for ADC for Terminology as well as Test Methods
different general test setups which include Sine Wave, Arbitrary Waveform, Step Waveform & F
Loop. To determine analog to digital converters’ stable performance, then different methods are
the servo based, ramp based, the ac histogram technique, the triangle histogram technique & th
technique. The one technique that is used for dynamic testing is the sine wave test.

Applications of Analog to Digital Converter

The applications of ADC include the following.

• At present, the usage of digital devices is increasing. These devices work based on the digita
analog to digital converter plays a key role in such kind of devices to convert the signal from
digital. The applications of analog to digital converters are limitless which are discussed belo
• AC (air conditioner) includes temperature sensors to maintain the temperature within the roo
conversion of temperature can be done from analog to digital with the help of ADC.
• It is also used in a digital oscilloscope to convert the signal from analog to digital to display.
• ADC is used to convert the analog voice signal to digital in mobile phones because mobile ph
digital voice signals but actually, the voice signal is in the form of analog. So ADC is used to
signal before sending the signal toward the transmitter of the cell phone.
• ADC is used in medical devices like MRI and X-Ray to convert the images from analog to dig
alteration.
• The camera in the mobile mainly used for capturing images as well as videos. These are sto

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digital device, so these are converted to digital form using ADC.

• The cassette music can also be changed into a digital like CDS & thumb drives use ADC.
• At present ADC is used in every device because almost all devices available in the market ar
version. So these devices use ADC.

Thus, this is about an overview of analog to digital converter or ADC converter & its types
understanding, only a few ADC converters are discussed in this article. We hope this furnishe
more informative to readers. Any further queries, doubts, and technical help on this topic you ca
below.

Photo Credits:

• Analog to Digital Conversion by microcontrollerboard

• Dual Slope A/D Converter by imgur
• Flash A/D Converter by bunniestudios
• Successive Approximation A/D Converter by electronics.dit

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