MODULE 3

Data or information can be stored in two ways, analog and digital. For a computer to use the
data, it must be in discrete digital form. Similar to data, signals can also be in analog and digital
form. To transmit data digitally, it needs to be first converted to digital form.
Digital-to-Digital Conversion
This section explains how to convert digital data into digital signals. It can be done in two ways,
line coding and block coding. For all communications, line coding is necessary whereas block
coding is optional.
Line Coding
The process for converting digital data into digital signal is said to be Line Coding. Digital data
is found in binary format. It is represented (stored) internally as series of 1s and 0s.

Digital signal is denoted by discreet signal, which represents digital data. There are three types of
line coding schemes available:

Uni-polar Encoding
Unipolar encoding schemes use single voltage level to represent data. In this case, to represent
binary 1, high voltage is transmitted and to represent 0, no voltage is transmitted. It is also called
Unipolar-Non-return-to-zero, because there is no rest condition i.e. it either represents 1 or 0.
Polar Encoding
Polar encoding scheme uses multiple voltage levels to represent binary values. Polar encodings is
available in four types:
Polar Non-Return to Zero (Polar NRZ)
It uses two different voltage levels to represent binary values. Generally, positive voltage
represents 1 and negative value represents 0. It is also NRZ because there is no rest condition.
NRZ scheme has two variants: NRZ-L ( bit 0 = positive & bit 1 = negative ) and NRZ-I ( bit 0 =
& bit 1 = negative ).

NRZ-L changes voltage level at when a different bit is encountered whereas NRZ-I changes
voltage when a 1 is encountered.
Return to Zero (RZ)
Problem with NRZ is that the receiver cannot conclude when a bit ended and when the next bit is
started, in case when sender and receiver’s clock are not synchronized.
RZ uses three voltage levels, positive voltage to represent 1, negative voltage to represent 0 and
zero voltage for none. Signals change during bits not between bits.
Manchester
This encoding scheme is a combination of RZ and NRZ-L. Bit time is divided into two halves. It
transits in the middle of the bit and changes phase when a different bit is encountered.
Differential Manchester
This encoding scheme is a combination of RZ and NRZ-I. It also transit at the middle of the bit
but changes phase only when 1 is encountered.
Bipolar Encoding
Bipolar encoding uses three voltage levels, positive, negative and zero. Zero voltage represents
binary 0 and bit 1 is represented by altering positive and negative voltages.

Block Coding
To ensure accuracy of the received data frame redundant bits are used. For example, in even-
parity, one parity bit is added to make the count of 1s in the frame even. This way the original
number of bits is increased. It is called Block Coding.
Block coding is represented by slash notation, mB/nB.Means, m-bit block is substituted with n-
bit block where n > m. Block coding involves three steps:
Division,
Substitution
Combination.
After block coding is done, it is line coded for transmission.
Scrambling
Scrambling is a technique that does not increase the number of bits and does provide
synchronization. The problem with techniques like Bipolar AMI(Alternate Mark Inversion) is
that continuous sequence of zero’s create synchronization problems one solution to this is
Scrambling.
There are two common scrambling techniques:
B8ZS(Bipolar with 8-zero substitution)
HDB3(High-density bipolar3-zero)
B8ZS(Bipolar with 8-zero substitution): This technique is similar to Bipolar AMI except when
eight consecutive zero-level voltages are encountered they are replaced by the sequence,
“000VB0VB”.
Note:
V(Violation), is a non-zero voltage which means the signal has the same polarity as the previous
non-zero voltage. Thus it is a violation of the general AMI technique.
B(Bipolar), is also a non-zero voltage level that is in accordance with the AMI rule (i.e., opposite
polarity from the previous non-zero voltage).
Example: Data = 100000000

Note: Both figures (left and right one) are correct, depending upon the last non-zero voltage
signal of the previous data sequence (i.e., sequence before current data sequence “100000000”).
HDB3(High-density bipolar3-zero): In this technique, four consecutive zero-level voltages are
replaced with a sequence “000V” or “B00V”. Rules for using these sequences:
If the number of nonzero pulses after the last substitution is odd, the substitution pattern will be
“000V”, this helps in maintaining a total number of nonzero pulses even.
If the number of nonzero pulses after the last substitution is even, the substitution pattern will be
“B00V”. Hence even the number of nonzero pulses is maintained again.
Example: Data = 1100001000000000
 Output explanation: After representing the first two 1’s of data we encounter four consecutive
zeros. Since our last substitutions were two 1’s(thus the number of non-zero pulses is even). So,
we substitute four zeros with “B00V”.
Scrambling is a technique used in digital electronics to provide a known sequence of bits to
allow for synchronization and detect errors. The code to implement scrambling will depend on
the specific requirements of the system.
Analog-to-Digital Conversion
Microphones create analog voice and camera creates analog videos, which are treated is analog
data. To transmit this analog data over digital signals, we need analog to digital conversion.
Analog data is a continuous stream of data in the wave form whereas digital data is discrete. To
convert analog wave into digital data, we use Pulse Code Modulation (PCM).
PCM is one of the most commonly used method to convert analog data into digital form.
It involves three steps:
 Sampling
 Quantization
 Encoding.
Sampling

The analog signal is sampled every T interval. Most important factor in sampling is the rate at
which analog signal is sampled. According to Nyquist Theorem, the sampling rate must be at
least two times of the highest frequency of the signal.
Quantization

Sampling yields discrete form of continuous analog signal. Every discrete pattern shows the
amplitude of the analog signal at that instance. The quantization is done between the maximum
amplitude value and the minimum amplitude value. Quantization is approximation of the
instantaneous analog value.
Encoding

In encoding, each approximated value is then converted into binary format.

Transmission Modes
The transmission mode decides how data is transmitted between two computers.The binary data
in the form of 1s and 0s can be sent in two different modes: Parallel and Serial.
Parallel Transmission

The binary bits are organized in-to groups of fixed length. Both sender and receiver are
connected in parallel with the equal number of data lines. Both computers distinguish between
high order and low order data lines. The sender sends all the bits at once on all lines.Because the
data lines are equal to the number of bits in a group or data frame, a complete group of bits (data
frame) is sent in one go. Advantage of Parallel transmission is high speed and disadvantage is the
cost of wires, as it is equal to the number of bits sent in parallel.
Serial Transmission
In serial transmission, bits are sent one after another in a queue manner. Serial transmission
requires only one communication channel.

Serial transmission can be either asynchronous or synchronous.

Asynchronous Serial Transmission
It is named so because there’is no importance of timing. Data-bits have specific pattern and they
help receiver recognize the start and end data bits.For example, a 0 is prefixed on every data byte
and one or more 1s are added at the end.
Two continuous data-frames (bytes) may have a gap between them.
Synchronous Serial Transmission
Timing in synchronous transmission has importance as there is no mechanism followed to
recognize start and end data bits.There is no pattern or prefix/suffix method. Data bits are sent in
burst mode without maintaining gap between bytes (8-bits). Single burst of data bits may contain
a number of bytes. Therefore, timing becomes very important.
It is up to the receiver to recognize and separate bits into bytes.The advantage of synchronous
transmission is high speed, and it has no overhead of extra header and footer bits as in
asynchronous transmission.
Introduction
The Pulse Amplitude Modulation is referred to as a very simple model of signal modulation
technique. These are the signals that are further transmitted and are inspected at a very particular
interval. On top of it, the modulating signal is proportional directly to the amplitude of the signal
Therefore it general multiple carrier signals like frequency, bandwidth and amplitude which are
also its core features. This particular phenomenon is also an outcome because of adding details to
the carrier signal in the course of transmission.
PAM sampling techniques

Flat top PAM is one of the crucial sampling techniques where the amplitude of the pulses is
proportional to modulating signals amplitude in the occurrence of pulses. In Natural PAM during
the occurrence of pulses the modulating signal amplitude is proportional to the amplitude
PAM: Advantages and disadvantages
Advantages
The circuit of PAM is very easy to make and the relative operation is quite easy as well
In PAM the transmission is quite easy and fast. The amplitude signal reception experiences zero
interference from any external factors
The Pulse Amplitude Modulation is capable of functioning with two distinct purposes, as this
modulation technique can carry various transmitted messages. PAM is also able to produce pulse
signals and transmit signals at the same time
In PAM, the modulation and demodulation happens automatically and it does not demand any
sort of manual activity for its functionality
Disadvantages
Pulse Amplitude Modulation calls for higher bandwidth for the transmission of a signal
It creates extra noises remnants that cause various disturbances.
In certain cases, it has been observed that PAM requires a huge volume of power for its
functionality
Application of Pulse Amplitude Modulation
The Pulse Amplitude Modulation is used in various sections.
It is used in Broadband interface communication for connectivity of the Ethernet.
It is used in controlling signals of the micro-controllers.
Pulse Amplitude Modulation is also used in the graphics cards and the associated high-speed
networking and it is also useful in the reduction of the noise to signal ratio.
PAM is also used in photobiology for the usage of spectrofluorimetric measurements during
photosynthesis.
It is also used in LED drivers for efficiency in the energy of the lighting.
Pulse Amplitude Modulation is also used to obtain clear pictures and proper signal clarity in
digital televisions.
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.
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.
Encoder
The digitization of analog signal is done by the encoder. It designates each quantized level by a
binary code. The sampling done here is the sample-and-hold process. These three sections (LPF,
Sampler, and Quantizer) will act as an analog to digital converter. Encoding minimizes the
bandwidth used.
Regenerative Repeater
This section increases the signal strength. The output of the channel also has one regenerative
repeater circuit, to compensate the signal loss and reconstruct the signal, and also to increase its
strength.
Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original signal. This
circuit acts as the demodulator.
Reconstruction Filter
After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a low-
pass filter is employed, called as the reconstruction filter to get back the original signal.
 Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and samples
it, and then transmits it in an analog form. This whole process is repeated in a reverse pattern to
obtain the original signal.
Sampling is defined as, “The process of measuring the instantaneous values of continuous-time
signal in a discrete form.”
Sample is a piece of data taken from the whole data which is continuous in the time domain.
When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e.,
High or Low, the signal has to be discretized in time. This discretization of analog signal is
called as Sampling.
The following figure indicates a continuous-time signal x (t) and a sampled signal xs (t). When x
(t) is multiplied by a periodic impulse train, the sampled signal xs (t) is obtained.

Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed
as a sampling period Ts.
Sampling Frequency fs = 1/Ts
Where,
 Ts is the sampling time
 fs is the sampling frequency or the sampling rate
Sampling frequency is the reciprocal of the sampling period. This sampling frequency, can be
simply called as Sampling rate. The sampling rate denotes the number of samples taken per
second, or for a finite set of values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be
highly considered. The rate of sampling should be such that the data in the message signal should
neither be lost nor it should get over-lapped. Hence, a rate was fixed for this, called as Nyquist
rate.
Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient
sample rate in terms of bandwidth for the class of functions that are bandlimited.
The sampling theorem states that, “a signal can be exactly reproduced if it is sampled at the
rate fs which is greater than twice the maximum frequency W.”
To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal whose
value is non-zero between some –W and W Hertz.
Such a signal is represented as x(f)=0 for| f|>W
For the continuous-time signal x (t), the band-limited signal in frequency domain, can be
represented as shown in the following figure.

We need a sampling frequency, a frequency at which there should be no loss of information,

even after sampling. For this, we have the Nyquist rate that the sampling frequency should be
two times the maximum frequency. It is the critical rate of sampling.
If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered, and if it
is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency
domain.

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

Where Ts = Sampling Period and

 Let us see what happens if the sampling rate is equal to twice the highest frequency (2W)
That means, fs=2W
Where,
 fs is the sampling frequency
 W is the highest frequency

 The result will be as shown in the above figure. The information is replaced without any
loss. Hence, this is also a good sampling rate.
 Now, let us look at the condition,
 fs<2W
 The resultant pattern will look like the following figure.


 We can observe from the above pattern that the over-lapping of information is done,
which leads to mixing up and loss of information. This unwanted phenomenon of over-
lapping is called as Aliasing.
Delta Modulation
Delta modulation (DM) is a signal conversion method used for speech communication when
quality is not a concern. In DM, the most basic kind of differential pulse-code modulation, the
difference between two sequences is stored in n-bit data streams (DPCM).
Delta modulation reduces the delivered data to a 1-bit data stream. It has the following
characteristics:
To mimic the analog signal, a series of segments is employed.
The next bits are determined by comparing the previous bits to each section of the expected
signal.
The change in information, i.e. an increase or decrease in signal amplitude from the previous
sample, is sent.
Whereas the no-change condition retains the modulated signal in the same 0 or 1 state as the
sample provided.
Delta modulation necessitates the use of resampling techniques to get a high signal-to-noise
ratio, which implies that the analog signal is recorded at a rate many times faster than the
Nyquist rate.
Principle of Delta Modulation
Delta modulation quantized the gap between the observed and previous steps instead of the value
of the input analog waveform.
The process of delta modulation compares the current sample value to the prior sample value.
The amplitude of the step signal will be raised or lowered based on the difference. When the
amplitude is raised, the step size is increased by one step, resulting in bit 1. If the amplitude is
lowered, the step size is reduced by one step, resulting in bit 0.
Block Diagram of Delta Modulation
Delta modulation employs oversampling to produce a high signal-to-noise ratio. The transmitter
circuit of a delta modulation system is made up of a Modulator, Channel, and Demodulator.

The integrator circuit has a Ts delay. The integrator’s output is a Ts-delayed staircase
approximation. The error signal is the difference between this staircase approximation and the
current sampling input signal.
This erroneous signal is sent into the quantizer circuit, which is made up of a hard limiter and an
input-output relationship. The mistake is quantized here into two values, namely. The quantizer
output is then coded to generate the appropriate Delta modulated wave.
Demodulation is performed in the receiver circuit using an integrator and a low pass filter. The
modulated wave is decoded first with a decoder, then the staircase approximation is recreated by
feeding the positive and negative pulses produced by the decoder to the integrator.
The high-frequency staircase waveform’s out-of-band quantizing noise is eliminated by running
the signal through a low-pass filter with a bandwidth equal to the original signal bandwidth.
Waveform Representation of Delta Modulation
The delta modulation waveform is represented by the below figure.

Here, the analog input signal is m(t) and the quantized signal is denoted by u(t).
Detection of a Delta Modulated Signal :- To detect a delta modulated signal, the following
steps can be performed:
Detect the start of frame. This can be done by looking for a specific bit pattern in the data stream
that indicates the start of a frame.
Determine the sampling rate and step size. These will be known parameters for decoding the
signal.
Initialize the integrator to the initial sample value. The integrator acts as a digital-to-analog
converter.
For each bit in the frame:
If the bit is 1, increase the integrator by the step size
If the bit is 0, decrease the integrator by the step size
The output of the integrator at each step will approximate the analog input signal.
Apply low pass filtering to the output to smooth out the steps and improve the approximation.
Advantages of Delta Modulation :- Some of its main advantages are as follows:
Delta modulation is a high-performance approach.
This modulation technique is utilized when extreme circuit simplicity is critical and the usage of
a high bit rate is permissible.
This technology gets rid of the need for correcting circuits in radio design and error
identification.
The dynamic range is large because the different step size covers such a variation.
Delta modulation is effective with narrower channel bandwidths.
There is no indication of granular or slope overload.
Slope error is reduced to a lesser degree.
Disadvantages of Delta Modulation :- Some drawbacks include:
Signals vary at a faster rate.
Granular noise can be observed.
Applications of Delta Modulation :- Some uses of delta modulation are listed below –
This modulation method is widely used in voice transmission systems such as telephones and
radio communications.
Delta modulation is particularly beneficial in systems where data quality is less critical than rapid
data delivery to the receiver.
For database reduction and real-time signal processing, this modulation is used for ECG
waveforms.
This modulation technique is used for analog-to-PCM encoding.
Delta modulation is a technique used in television systems.