UNIT - II Digital Transmission

Introduction, Pulse modulation, PCM sampling, sampling rate, signal to quantization noise
rate, companding analog and digital percentage error, delta modulation, adaptive delta
modulation, differential pulse code modulation, pulse transmission types-Intersymbol
interference, eye patterns.
1 Digital Transmission

Introduction Definition

Digital Transmission refers to the transmission of information in binary format (0s

and 1s) over a communication channel. This process involves converting analog or digital
information into digital signals suitable for communication over wired or wireless media.

Key Components of Digital Transmission System

1. Source – Generates data (text, audio, video, etc.)

2. Source Encoder – Converts data to binary stream
3. Channel Encoder – Adds redundancy for error detection and correction
4. Modulator – Converts binary data into signal form suitable for transmission
5. Transmission Medium – The channel (optical fiber, coaxial cable, air, etc.)
6. Demodulator – Extracts binary data from received signal
7. Channel Decoder – Removes redundancy, corrects errors
8. Source Decoder – Reconstructs the original message

Analog vs Digital Transmission

Feature Analog Digital

Signal type Continuous Discrete

Noise immunity Poor High

Equipment cost Low High (initially)

Transmission Degrades over Maintains

quality time integrity

Bandwidth efficiency Depends on modulation Improved with coding

Advantages of Digital Transmission

 High noise immunity

 Easier encryption and compression
 Efficient use of bandwidth
 Easy integration with computer systems
 Long-distance transmission using repeaters
 Scalability and compatibility with modern digital networks
Applications

 Mobile communications (4G/5G)

 Satellite communications
 Digital television
 Secure military communications

Simple Digital Transmission System

Source Channe Transmissio

Source Modulator n
Encoder Encoder ]
] Medium]
]

Destination Source Channel Demodulator

] ]
Decoder Decoder
] ]

(2)Pulse Modulation:

Pulse Modulation is a technique in which analog information is converted into pulses for
transmission. These pulses represent the information either in their amplitude, duration, position, or
presence.

Classification of Pulse Modulation

Pulse modulation techniques are mainly divided into:

1. Analog Pulse Modulation

 Used when the modulating signal is analog.

 Types:
o Pulse Amplitude Modulation (PAM)
o Pulse Width Modulation (PWM) / Pulse Duration Modulation (PDM)
o Pulse Position Modulation (PPM)

2. Digital Pulse Modulation

 The modulating signal is sampled and quantized.

 Types:
o Pulse Code Modulation (PCM)
o Delta Modulation (DM)
 Analog Pulse Modulation:

 Pulse Amplitude Modulation (PAM):

The amplitude of the pulses varies in accordance with the instantaneous amplitude of the
modulating signal.

 Pulse Width Modulation (PWM):

The width of the pulses is varied in proportion to the amplitude of the modulating signal.

 Pulse Position Modulation (PPM):

The position of each pulse, relative to a reference pulse, is varied based on the modulating
signal's amplitude.Pulse Code Modulation (PCM)

PCM is the most commonly used digital pulse modulation technique. It converts an analog signal into
a digital signal through:

Steps in PCM:

1. Sampling – Convert analog signal into discrete-time samples.

2. Quantization – Convert the amplitude into discrete levels.
3. Encoding – Convert quantized values into binary code.

Sampling

Pulse modulation techniques deal with discrete signals. To obtain such signals, a continuous-time
signal must be converted into a discrete-time signal, a process known as sampling. Sampling is the
process of converting a continuous-time signal into its equivalent discrete-time signal by taking values
of the signal at specific intervals of time. In this process, certain instants of data from the continuous
signal are repeatedly sampled.
 The figure below illustrates a continuous-time signal and its sampled version xs(t)x_s(t)xs​(t).
When the continuous-time signal is multiplied by a periodic impulse train, the sampled signal
xs(t)x_s(t)xs​(t) is obtained

A sampling signal is a periodic train of pulses, having unit amplitude, sampled at

equal intervals of time Ts, which is called as the Sampling time. This data is transmitted at the time
instants Ts and the carrier signal is transmitted at the remaining time.Sampling Rate

To discretize the signals, the gap between the samples should be fixed. That gap
can be termed as the sampling period Ts.

SamplingFrequency=1Ts=fs

Where,

Ts = the sampling time

fs = the sampling frequency or sampling rate

Sampling Theorem

While considering the sampling rate, it is important to decide how high the rate should be. The
sampling rate must be chosen such that no data in the message signal is lost, and at the same time, the
signal is not unnecessarily oversampled. The Sampling Theorem states that a signal can be exactly
reproduced if it is sampled at a rate greater than or equal to twice the maximum frequency component
WWW present in the signal.

In simpler terms, for the accurate reproduction of the original signal, the sampling frequency
fsf_sfs​should satisfy the condition:

fs≥2Wf_s \geq 2Wfs​≥2W

 where fsf_sfs​is the sampling frequency and WWW is the highest frequency present in the signal.
This minimum required rate of sampling is called the Nyquist Rate.

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.

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

If the signal is sampled above the Nyquist rate, the original signal can be recovered. The
following figure explains a signal, if sampled at a higher rate than 2w in the frequency domain.

If the same signal is sampled at a rate less than 2w, then the sampled signal would 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.

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

signal, taking on the identity of a lower-frequency component in the spectrum of its sampled
version.

Hence, the sampling of the signal is chosen to be at the Nyquist rate, as was stated in the
sampling theorem. If the sampling rate is equal to twice the highest frequency (2W).

That means,

fs=2W
Where,

fs = 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 a good sampling rate.

Sampling Process in Digital Communication

The sampling process includes the following steps:
1. The continuous signal is taken as an input.
2. Sampling is performed to convert this signal into a digital representation.
3. In addition to sampling, quantization of a signal is performed.
4. After the above step, encoding of the signal is done.

Important Terminologies of Sampling in Digital Communication

There are few important terminologies of Sampling in Digital Communication discussed below :
 Sampling
 Sample
 Sampling Rate or Sampling Frequency
 Nyquist Rate
 Nyquist Interval
 Quantization

1. Sampling
It is the process by which, we convert CTS (continuous time signal) into DTS (discrete time
signal) by taking the signal values at some distinct points in time, meaning that this is used to take
samples of analog signals at some points in time (regular or irregular)

2. Sample

It can be defined as the numeric value of an analog signal at a specific time. It is just the signal's
measured amplitude at a particular time and converting it to a digital representation.

3. Sampling Rate or Sampling Frequency

It refers to the number of samples or data points taken per unit of time from an analog signal to
convert it into a digital format. It is also known as sampling frequency. It is measured in Hertz
(Hz).
The formula for sampling rate or sampling frequency is given by:
SamplingRate=1/Ts=fs
where,
Ts = sampling time fs = sampling

4.Nyquist Rate

It is the minimum sampling rate required to accurately capture an analog signal in digital form
without information loss. It is also known as Nyquist Frequency or Nyquist Limit.
It is defined as twice the maximum frequency component present in the analog signal.
Mathematically it can be represented as:
fs=2fmax
where,
fs = Sampling Rate or Nyquist Rate (Hz)
fmax = Maximum frequency component (Hz)

5.Nyquist Interval

The Nyquist interval, also known as the Nyquist period, is the time interval between consecutive
samples in a digital signal or digital sampling system. It is the reciprocal of the Nyquist rate,
which is the smallest sampling rate required to accurately capture an analog signal in digital form
without information loss. Mathematically it can be represented as:
T=1/NyquistRate
Where,
T = Nyquist interval (sec)
Nyquist Rate is the sampling rate (Hz)
 6.Quantization

It is the process to represent a continuous-valued signal with a limited set of discrete values. In other
words, it involves mapping a continuous signal's infinite range of potential values to a finite
collection of discrete values.

Quantization of a Signal

Methods of Sampling

1. IdealSampling
Concept:
Ideal sampling, also known as impulse or Dirac sampling, is a theoretical notion in
which samples of a continuous signal are taken at specific time intervals, often at the Dirac delta
function impulse points.
Sampling Process:
Each sample in perfect sampling is an impulse or delta function at the sampling instant.
The sampled signal can be represented mathematically as the product of the continuous signal and
the Dirac delta function.

Reconstruction:
The reconstruction of the original signal from ideal samples, we can use interpolation which
uses the functions. Ideal sampling is a simple approach to express and analyze sampling theory,
however it is not practical due to the requirement for infinite bandwidth.
 Ideal Sampling

2. Natural Sampling

Concept:
Natural sampling, also known as zero-order hold sampling, involves taking discrete
interval samples of a continuous signal, similar to uniform sampling. The difference, though, is
in how the samples are gathered.
Sampling Process:
Each sample is taken in natural sampling by retaining the value of the continuous signal
constant for the duration of the sampling period.

Sampling Circuit

Reconstruction:
The reconstruction of the original signal from natural samples, it usually involves
connecting the samples with flat line segments. This method simplifies the reconstruction
process compared to ideal sampling.

Natural Sampling
3. Flat-Top Sampling

Concept:
Flat-top sampling is a type of natural sampling in which each sample is obtained by
maintaining the value of the continuous signal constant for a set period of time, resulting in a flat-
top waveform.
Sampling Process:
Instead of retaining the value for the whole sample interval, flat-top sampling holds it only
for a portion of the interval while allowing it to change at the beginning and end.

Sampling Circuit

Reconstruction: The reconstruction of the original signal from flat-top samples, we can use
interpolation techniques. Flat-top sampling is used in applications where it is desirable to minimize
the effects of finite bandwidth and aliasing.

Flat Top Sampling

3 Signal-to-Quantization Noise

Ratio (SQNR)

Definition:

Signal-to-Quantization Noise Ratio (SQNR) is a measure of the quality of a

quantized signal, defined as the ratio of the power of the original signal to the power of the
quantization noise introduced during analog-to-digital conversion.
It is expressed in decibels (dB) and represents how much the original signal stands out compared to
the noise introduced by quantization.

Mathematical Expression

SQNR=Signal Power/Quantization Noise Power Or in decibels (dB):

SQNRdB=10 log 10(Signal Power/Quantization Noise Power)

Formula (for uniform quantization of sinusoidal signals):

If the signal is uniformly distributed and the quantization is uniform:

SQNRdB=6.02n+1.76 dB

Where:

n= number of bits used for quantization

This equation shows that increasing the number of bits by 1 increases SQNR by

approximately 6 dB. Example:

If a PCM system uses 8-bit quantization, then:

SQNR=6.02×8+1.76=49.92dB
This means the signal power is about 10^4.992/10 ≈ 98,000 times greater than the
quantization noise.

SQNR Table:

Number of Bits (n) SQNR (dB)

4 bits 25.84 dB

8 bits 49.92 dB

12 bits 74.00 dB

16 bits 98.08 dB

(4) Companding analog and digital percentage error

Companding

The word Companding is derived from the combination of the terms Compressing and
Expanding, which means it performs both operations. It is a non-linear technique used in Pulse Code
Modulation (PCM), where the signal is compressed at the transmitter and expanded at the receiver.
 By using this method, the effects of noise and crosstalk are significantly reduced, thereby
improving signal quality.

Model of Companding

The figure below illustrates the companding model, which is used to achieve non-
uniform companding.

As we can see that the companding model consists of a compressor, a uniform quantizer and an expander.

We have already discussed that companding is formed by merging the compression and
expanding. Initially at the transmitting end the signal is compressed and further at the receiving end
the compressed signal is expanded in order to have the original signal.

Initially at the transmitting end, the signal is first provided to the compressor. The
compressor unit amplifies the low value or weak signal in order to increase the signal level of the
applied input signal.
XKJ[IK3ZNK3KLLKIZ
while if the input signal is a high level signal or strong signal then compressor attenuates that
signal before providing it to the uniform quantizer present in the model.

This is done in order to have an appropriate signal level as the input to the uniform
quantizer. We know a high amplitude signal needs more bandwidth and also is more likely to
distort. Similarly, some drawbacks are associated with low amplitude signal and thus there exist
need for such a unit.

The operation performed by this block is known as compression thus the unit is called compressor.

The output of the compressor is provided to uniform quantizer where the quantization of the applied

signal is performed. At the receiver end, the output of the uniform quantizer is fed to the expander.

It performs the reverse of the process executed by the compressor. This unit when receives a low
value signal then it attenuates it. While if a strong signal is achieved then the expander amplifies
it.
This is done in order to achieve the originally transmitted signal at the output.

Characteristic of Compander

As we know companding is composed of compression and expanding. So, here in this session we
will separately discuss the compressor and expander characteristic.

Compressor characteristic:
The figure below shows the graphical representation of characteristic of the compressor:

The graph clearly represents that the compressor provides high gain to weak signal and low gain to high
input signal.

Expander characteristic:
Here the figure shows the characteristic of expander:

As we have already discussed that expander performs reverse operation of the

compander. So, it is clear from the above figure that artificially boosted signals is attenuated
to have the originally transmitted signal.

The figure below represents the companding curve or PCM system:

 The compressor and expander performs inverse operations thus in the above figure the dotted line
represents the linear characteristic of the compander indicating that the originally transmitted signal is
recovered at the receiver.

Types of Companding techniques

There are two types of Companding techniques. They are

Analog Companding:
Concept:Analog companding involves compressing the signal's dynamic range before
analog-to-digital conversion (ADC) and expanding it back after digital-to-analog conversion
(DAC).

Benefit:This approach is particularly beneficial for signals with a wide dynamic range,
like audio signals. By compressing the signal, smaller amplitude variations are amplified,
making them less susceptible to quantization noise.

Percentage Error Reduction:Companding minimizes the percentage error introduced

during the quantization process by ensuring that weaker signals are less affected by the limited
number of quantization levels available in the
ADC. This is because the quantization intervals are smaller for weaker signals after compression.

Digital Companding:

Concept:Digital companding, such as A-law and µ-law, uses specific mathematical functions to
non-linearly map input signal values to digital codes.
Benefit:Similar to analog companding, digital companding provides better resolution for
weaker signals, reducing the percentage error associated with their quantization. It achieves this
by using smaller quantization steps for smaller input values.
 Example:In T-carrier systems, A-law and µ-law companding are used to reduce the
quantization error in digitized voice signals.

Percentage Error Calculation:

The percentage error is a measure of the difference between the true value and the measured
(or estimated) value, expressed as a percentage of the true value. Formula: Percentage Error =
(|True Value, Measured Value| / True Value), and 100.

A-law Companding Technique

Uniform quantization is achieved at A = 1, where the characteristic curve is linear

and no compression is done. A-law has mid-rise at the origin. Hence, it contains a non-zero
value.
A-law companding is used for PCM telephone systems.

µ-law Companding Technique

Uniform quantization is achieved at = 0, where the characteristic curve is linear

and no compression is done. µ-law has mid-tread at the origin. Hence, it contains a zero
value.
µ-law companding is used for speech and music signals.

(5)Delta Modulation(DM)?
Delta modulation is an analog to digital and digital to analog signal conversion
technique. Delta modulation is employed to realize high signal to noise ratio. It uses one bit PCM
code to realize digital transmission of analog signal. With delta modulation, instead of transmit a
coded illustration of a sample solely one bit is transmitted, that merely indicates whether or not the
sample is larger or smaller than the previous sample. it's the best type or simplest type of Differential
Pulse Code Modulation. Delta modulation signal is smaller than Pulse Code Modulation system.
If signal is large, the next bit in digital data is 1 otherwise 0.
Operating principle of Delta Modulation

The operating principle of DM is such that, a comparison between present and previously
sampled value is performed, the difference of which decides the increment or decrement in the
transmitted values.

Simply put, when the two sample values are compared, either we get difference having a
positive polarity or negative polarity.

If the difference polarity is positive, then the step of the signal denoted by Δ is increased by
1. As against in case when difference polarity is negative then step of the signal is decreased i.e.,
reduction in Δ.

When +Δ is noticed i.e., increase in step size, then 1 is transmitted. However, in the case of
–Δ i.e., decrease in step
size, 0 is transmitted.

Hence, allowing only a single binary bit to get transmitted for each sample.

Generation of delta modulated signal

The block diagram for the generation delta modulated signal is shown below:

As we can see the above figure consists of an LPF, a comparator, a product modulator along
with pulse generator and quantizer. Here, a feedback path is also provided to the circuit, where the
output of modulator acts as input to the comparator.

The message signal that is to be transmitted is fed to a low pass filter that passes the
low-frequency component and eliminates the high-frequency component. It is also referred to
as aliasing filter.

The output of LPF is then given to a comparator unit, which compares the message signal
m(t) with an arbitrary signal m'(t) for the first time. The comparator after comparing 2 signals
generates the difference between the two.
 The difference can be of either positive polarity or negative polarity. This depends on
message and arbitrary signals that are getting subtracted.

This difference signal now acts as input to the product modulator. Another input to the
modulator is a pulse signal generated by the pulse generator. These two signals are multiplied
in the modulator.

The output of the modulator is a pulsed signal whose pulses will be of equal magnitude having
polarity either positive and negative.

The polarity totally depends on the output of the comparator. The output of the
modulator is given to quantizer. The quantizer generates the output in the form of steps.

If positive magnitude pulse is provided to the quantizer as its input then quantizer performs
increment by 1 step size, Δ. It is very easy to understand that positive pulse at the output of the
modulator shows that message signal is greater than the arbitrary signal. Thus quantizer increases Δ
by 1.

Similarly, in the case of negative magnitude pulse, the step size gets decreased by 1. This is so
because m'(t) exceeds m(t), thereby generating a pulse of negative polarity.
Thus, quantizer decreases Δ by 1.

The output of the modulator at the same time, through a feedback path, is provided to the accumulator.

An accumulator is nothing but a device that stores the signal for further operation. The output of the
accumulator now behaves like the second input of the comparator. Thus, we say that the present sample
value is compared with the previous one for further operation.

Hence the process repeats in such a manner.

In the end, depending on the staircase signal if the step size is +Δ then binary 1 is transmitted and if it is
–Δ then binary 0
is transmitted.

Waveform Representation of Delta Modulation

The figure below shows the delta modulation waveform:

Here, the analog input signal is m(t) and the quantized signal is denoted by u(t). The binary sequence
according to the step size that is actually transmitted is shown at the bottom of the figure shown
above.

Detection of delta modulated signal

The detection of a delta modulated signal is not a complex process and is somewhat
reverse of generation of a delta modulated signal.

The figure below shows the block diagram for the representation of detection of delta modulated signal.

The detection circuitry basically consists of an accumulator and an LPF. The binary signal
transmitted is provided to the accumulator section.

The accumulator consists of a summation unit and a delay unit. The transmitted signal along
with the delayed signal is added at the summation unit.

If here the input is binary 1 then after a delay the output of the accumulator shown increased
step size +Δ. However, in the case of binary 0 as input, a decrease in step size is noticed. This
generates the staircase signal equivalent to the message signal.

The output of the accumulator is provided to the LPF that smoothens the staircase signal in order
to regenerate the original message signal.

Advantages of delta modulation

 Due to transmission of 1 bit per sample, it permits low channel bandwidth as well as signaling rate.
 ADC is not required. Thus permits easy generation and detection.
Disadvantages of delta modulation

 Delta modulation leads to drawbacks such as slope overload distortion and granular noise.
Applications of delta modulation

It is widely used in radio communication devices and digital voice storage and voice information
transmission where signal quality is less important.
(6) What is Adaptive Delta Modulation

Definition: This is a type of delta modulation technique. In digital modulation, there is a

problem in determining the step size, which may have an influence on the quality of the wave, which
is produced as output. We require a larger step size in the steep slope of the modulating signal, and a
small step size is required when the message has a small slope. This kind of detail is ignored in
Delta Modulation techniques. So, we need to have control over the step size according to our
requirements so that we are able to obtain the sample output in the required fashion. This is the main
concept behind adaptive Delta Modulation.

Block Diagram of ADM

The block diagram of adaptive delta modulation is shown below which includes an ADM
transmitter & receiver. The transmitter has a summer, a quantizer, a delay circuit, and a logic circuit.
Here, the step size is kept fixed between some predefined maximum and minimum values.

Adaptive Delta Modulation Transmitter

The upper limit is used to control the slope overload distortion and the lower limit is used to
control the granular noise. The step size increase or decreases based on a certain set of rules.
Adaptive Delta Modulation Receiver

The above figure represents an ADM receiver. The ADM receiver has two parts. The first
part is used to produce the step size from the incoming bits. The bits are then applied to the second
part of the receiver which contains an accumulator. The function of the accumulator is to build up
the staircase waveform. The signal is then passed through a low pass filter which is used to
smoothen the staircase waveform and reconstruct the original signal.

Adaptive Delta Modulation Theory

An Adaptive Delta Modulator is basically used to quantize the difference between the current
signal value and the predicted value of the following signal. It uses variable step height in order to
predict the consequent values. In ADM, the voltage gain which is controlled by the amplifier is
adjusted by the output signal which is produced from the sampler.

The step size is determined by the amplifier gain. This is because the step size is proportional
to the amplifier gain. So, we can say that ADM is basically a type of delta modulation. It is mainly
used to overcome slope overload distortion and granular noises which occurs in delta modulation.
Here, the step size of the quantifier varies according to the time.

Differences between Delta Modulation & Adaptive Delta Modulation

The differences between these two include the following.

 In delta modulation, the step size cannot be varied. It remains fixed for the entire signal.
However, in the case of ADM, the step size can be varied according to the variation of the
signal.
 In delta modulation, slope overload distortion and granular noise can be present but in ADM,
quantization noise is present. The other types of errors are mainly avoided.
 This modulation utilizes the bandwidth a lot more efficiently as compared to that of delta
modulation.
 It has a wider dynamic range than delta modulation.

Advantages

The advantages of adaptive delta modulation include the following.

 This modulation can be used to reduce the slope error which is present in delta modulation.
 This can be used to remove the granular noises from the signal.
 It has an improved signal to noise ratio as compared to that of delta modulation.
 ADM has a low pass filter that can be used to remove the quantization noise.
 It has a higher dynamic range as compared to delta modulation.
There is no need for error detection in the case of adaptive delta modulation.
Applications

The applications of adaptive delta modulation include the following.

 It is effectively used in audio communications.

 It can be used in systems that require improved voice quality.
 It is used in voice coding.
 It is used in television signal transmission.
 The SECURENET line of Motorola uses 12kbit/sec adaptive delta modulation.
 It is used by NASA for different communications between spacecraft and mission control.
 The US Army also uses adaptive delta modulation in order to conserve bandwidth over various
tactical links.

(7)Differential pulse code modulation:

Differential Pulse Code Modulation (DPCM) is a digital signal processing technique that
improves upon standard Pulse Code Modulation (PCM) by encoding the difference between
successive signal samples, rather than encoding the absolute sample values.

Working of DPCM:

Prediction:
Predict the current sample value using previous samples (usually using a linear predictor).

Differencing:
Subtract the predicted value from the actual sample value to get the prediction error.

Quantization:
Quantize the prediction error to a limited number of levels.

Encoding:
Encode the quantized error for transmission.

Decoding:
At the receiver, the quantized error is added to the predicted sample to reconstruct the
original signal.

Mathematical Expression:

Let:

x(n) = original sampled signal x^(n) = predicted value of x(n)

e(n)=x(n)−x^(n) = prediction error q(e(n))= quantized error

Transmitted signal: q(e(n))

Reconstructed signal at receiver:

x^(n)+q(e(n))

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 −

x(nTs) is the sampled input xˆ(nTs) is the predicted sample

e(nTs) is the difference of sampled input and predicted output, often called as
prediction error v(nTs) is the quantized output

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

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.

Advantages of DPCM:

 Requires fewer bits than PCM (more efficient).

 Reduces redundancy between adjacent samples.
 Suitable for audio and image compression.

Disadvantages:

 Sensitive to transmission errors.

 Prediction errors can accumulate (error propagation).

Applications:

 Voice compression (e.g., in telephony).

 Image and video coding (e.g., JPEG and MPEG).
 Speech transmission in mobile networks.

(8) Intersymbol interference(ISI)

Inter-Symbol Interference (ISI) is a form of distortion that occurs when one symbol in a digital
signal interferes with subsequent symbols. This happens because the tail end of one symbol "spills
over" into the next, causing confusion at the receiver and degrading the performance of the
communication system.

Why ISI Happens:

Bandwidth Limitation: Transmission channels (like cables, optical fibers, or wireless

media) have limited bandwidth, which causes pulse spreading.

Multipath Propagation: In wireless systems, signals arrive via multiple paths with different

delays.

Dispersive Channels: In fiber optics or some copper channels, different frequencies travel at

different speeds.

Filtering Effects: Poor design or excessive filtering distorts signal shape.

Effects of ISI:

Impact Description

Bit Errors The receiver may misinterpret a 1 as a 0 or vice versa.

Eye Pattern Closure ISI causes the eye diagram to close horizontally and vertically.

Reduced Data Rates More spacing (guard bands or error correction) may be needed to overcome ISI.

How to Reduce ISI:

Equalization: Use equalizers at the receiver to compensate for channel distortion. Pulse Shaping: Use

pulses like raised cosine or Nyquist pulses to minimize ISI. Channel Coding: Introduce redundancy to

help detect and correct errors.

Timing Recovery: Ensure accurate sampling at optimal times (center of eye diagram).

Adaptive Filters: Dynamically adjust filtering based on channel conditions.

Correlative Coding

Correlative Coding and ISI

So far, we have discussed that Inter-Symbol Interference (ISI) is generally an unwanted

phenomenon that degrades the quality of a signal. However, if ISI is introduced in a controlled manner,
it becomes possible to achieve a bit rate of 2W2W2W bits per second in a channel of bandwidth WWW
Hertz. Such schemes are called Correlative Coding or Partial Response Signaling Schemes.

Since the amount of ISI in these schemes is known in advance, the receiver can be designed
accordingly to compensate for the interference, ensuring reliable signal detection. One of the
fundamental methods to implement correlative coding is Duo-binary Signaling.

Duo-binary Signaling

The term duo-binary refers to doubling the transmission capability of a binary system. To
understand this, let us consider a binary input sequence consisting of uncorrelated binary digits, each
having a duration of TbT_bTb​seconds. In this scheme, the binary symbol "1" is represented by a
voltage of +V+V+V, while the binary symbol "0" is represented by a voltage of −V-V−V. By
introducing controlled correlation between adjacent symbols, duo-binary signaling effectively increases
the efficiency of data transmission while keeping ISI manageable.
 The output of a duo-binary coder is expressed as the sum of the present binary digit and the
previous binary digit. Mathematically, it is given by:

ck=ak+ak−1c_k = a_k + a_{k-1}ck​=ak​+ak−1​

where aka_kak​represents the present binary digit, and ak−1a_{k-1}ak−1​represents the previous
binary digit.

This equation shows that the original input sequence of uncorrelated binary digits is transformed
into a sequence of correlated three-level pulses {ck}\{c_k\}{ck​}. The correlation between these pulses
can be understood as the deliberate introduction of Inter-Symbol Interference (ISI) into the transmitted
signal in a controlled manner, which is the essence of duo-binary signaling.

Countering ISI:

There are several techniques in telecommunications and data storage that try to work around the
problem of intersymbol interference.

 Design systems such that the impulse response is short enough that very little energy from one
symbol smears into the next symbol.
 Separate symbols in time with guard periods.
 Apply an equalizer at the receiver, that, broadly speaking, attempts to undo the effect of the
channel by applying an inverse filter.
 Apply a sequence detector at the receiver, that attempts to estimate the sequence of
transmitted symbols using the Viterbi algorithm.

Consecutive raised-cosine impulses, demonstrating zero-ISI property

(9)EYE PATTERN :
An eye pattern, also known as an eye diagram, is a visual representation of a digital signal's
characteristics, used to evaluate signal quality in digital communication systems. It's essentially a
superposition of all possible bit sequences within a certain time interval, displayed on an
oscilloscope. The "eye" shape that emerges reveals information about intersymbol interference
(ISI), noise, jitter, and other distortions.
 How it works:

1. Display:

An oscilloscope is used to display the received signal. The horizontal axis represents
time, typically one or two bit periods, and the vertical axis represents signal amplitude.

2. Superposition:

The oscilloscope repeatedly samples the signal and overlaps these samples, creating a
composite image.

3. Eye Opening:
The resulting pattern looks like a series of eyes, hence the name. The "eye" opening is
the area between the upper and lower envelopes of the pattern.

4. Interpretation:

. ISI: A narrower eye opening indicates more ISI, which occurs when the tail of one pulse
interferes with the adjacent pulse.
. Noise: Noise degrades the eye pattern, making it less clear and potentially causing errors.
. Jitter: Jitter, or timing variations, widens the eye opening horizontally.
. Optimal Sampling Time: The eye opening also indicates the best time to sample the signal to
minimize errors.

Importance in Digital Communication:

Signal Quality Assessment:

Eye patterns provide a quick and intuitive way to assess the quality of a received digital signal.

Error Rate Prediction:

The size and shape of the eye opening are directly related to the bit error rate (BER).

Troubleshooting:

Eye patterns help identify the source of signal degradation, such as ISI, noise, or jitter.

System Optimization:

By analyzing the eye pattern, engineers can optimize system parameters like equalization and
filtering to improve signal quality.
Jitter is the short-term variation of the instant of digital signal, from its ideal position, which may lead to
data errors.

When the effect of ISI increases, traces from the upper portion to the lower portion of the
eye opening increases and the eye gets completely closed, if ISI is very high.
An eye pattern provides the following information about a particular system.

Actual eye patterns are used to estimate the bit error rate and the signal-to-noise ratio.

The width of the eye opening defines the time interval over which the received wave can be
sampled without error from ISI.

The instant of time when the eye opening is wide, will be the preferred time for sampling.

The rate of the closure of the eye, according to the sampling time, determines how sensitive
the system is to the timing error.

The height of the eye opening, at a specified sampling time, defines the margin over noise.

Equalization

For reliable communication to be established, we need to have a quality output. The

transmission losses of the channel and other factors affecting the quality of the signal, have to be
treated. The most occurring loss, as we have discussed, is the ISI.To make the signal free from ISI,
and to ensure a maximum signal to noise ratio, we need to implement a method called
Equalization. The following figure shows an equalizer in the receiver portion of the
communication system.

The noise and interferences which are denoted in the figure, are likely to occur, during
transmission. The regenerative repeater has an equalizer circuit, which compensates the
transmission losses by shaping the circuit. The Equalizer is feasible to get implemented.
Features of the Eye Pattern:

Feature Description

Eye opening The vertical opening indicates the noise margin. Wider opening means better
signal quality.

Eye closure Partial or full closure indicates distortion from ISI, jitter, or noise.

Timing jitter Horizontal spread at the zero-crossings shows timing uncertainties.

Rise and fall times Show the speed of the signal transitions.

Signal amplitude The vertical eye height reflects voltage levels and potential degradation.

Factors Affecting Eye Pattern:

Inter-Symbol Interference (ISI): Overlapping of adjacent bits causes blurring.

Noise: Random variations distort the shape and close the eye vertically.

Timing Jitter: Variations in the bit timing cause horizontal eye closure.

Bandwidth limitations: Slower transitions due to limited bandwidth affect the clarity of the eye.

Advantages of Eye Patterns:

 Quick visual method to assess signal quality.

 Can identify various impairments (jitter, noise, ISI). Helps in timing recovery and clock

synchronization.
Limitations:

 Qualitative more than quantitative. Requires a stable triggering clock.

 Not effective for non-periodic or bursty signals.