DCOM PAPER SOLUTION MUMBAI UNIVERSITY

SUBJECT CODE-ECC501

Compare ISI and ICI

ISI and ICI are abbreviations used in the field of digital communication and signal processing, and they
stand for Intersymbol Interference and Inter-Carrier Interference, respectively. While both terms refer to
types of interference that can occur in communication systems, they are associated with different aspects
of the signal.

Intersymbol Interference (ISI):

Intersymbol Interference refers to the distortion or overlap of symbols in a digital communication system.
It occurs when the received symbols spread into adjacent symbol periods, leading to interference and
difficulty in correctly detecting the symbols. ISI typically occurs in systems where the channel or
transmission medium introduces dispersion or multipath propagation.

The main causes of ISI are delays and frequency-selective fading in the channel. Delays cause different
paths of the transmitted signal to arrive at different times, while frequency-selective fading affects
different frequency components of the signal unequally. Both of these phenomena can result in
overlapping symbols and, thus, interference.

ISI can be mitigated through various techniques such as equalization, which attempts to compensate for
the distortion caused by the channel and recover the original symbols. Adaptive equalization algorithms
and techniques like maximum likelihood sequence estimation (MLSE) are commonly employed to
combat ISI.
Inter-Carrier Interference (ICI):

Inter-Carrier Interference refers to the interference that arises in multi-carrier communication systems,
such as orthogonal frequency-division multiplexing (OFDM). OFDM divides the available frequency
spectrum into multiple narrow subcarriers, each carrying a portion of the data. In such systems, ICI occurs
when the subcarriers are not perfectly synchronized, resulting in interference between adjacent
subcarriers.
The main causes of ICI are frequency offsets and Doppler shifts. Frequency offsets can occur due to
imperfections in the local oscillators of the transmitter and receiver, while Doppler shifts arise when there
is relative motion between the transmitter/receiver and the channel (e.g., in mobile communication
systems).

ICI can degrade the performance of multi-carrier systems by introducing inter-symbol interference
between the subcarriers. To mitigate ICI, techniques such as frequency and phase synchronization, cyclic
prefix insertion, and advanced equalization algorithms are employed.

In summary, ISI refers to interference caused by overlapping symbols in a single-carrier communication

system, often due to delays and frequency-selective fading. On the other hand, ICI pertains to interference
between subcarriers in multi-carrier systems like OFDM, caused by frequency offsets and Doppler shifts.

What is AWGN and Matched Filter

Matched filter diagram

Matched filter definition is: if a filter generates an output to maximize the output peak power ratio to
mean noise power within its frequency response then it is called a matched filter. In telecommunications,
it is the optimal linear filter used to increase the SNR or signal-to-noise ratio in the existence of additive
stochastic noise.

AWGN (Additive White Gaussian Noise) is a type of random noise that is often encountered in
communication systems and signal processing. It is called "white" because it has a constant power
spectral density across all frequencies, and "Gaussian" because the probability distribution of its samples
follows a Gaussian (or normal) distribution. AWGN can be introduced by various factors such as thermal
noise, atmospheric interference, or electronic components in a communication channel.

AWGN is commonly used as a model to represent the effects of noise in communication systems. When a
signal is transmitted through a noisy channel, the received signal is corrupted by the AWGN, which adds
random variations to the original signal. This noise can degrade the quality of the received signal and
make it more difficult to accurately extract the transmitted information.

Matched filtering, on the other hand, is a technique used to maximize the signal-to-noise ratio (SNR) and
improve the detection of a known signal in the presence of noise. It is commonly used in communication
systems to recover transmitted signals.

A matched filter is a linear filter whose impulse response is a time-reversed and conjugate complex
version of the transmitted signal. The received signal is passed through this filter, which essentially
correlates the received signal with the expected shape of the transmitted signal. The output of the matched
filter is then processed to make decisions about the presence or absence of the transmitted signal.

The main principle behind matched filtering is that the filter response is maximized when the received
signal matches the expected signal. This is because the correlation between the received signal and the
filter impulse response is maximized when they are aligned. As a result, the matched filter enhances the
signal power while suppressing the noise power, leading to improved detection performance.

In the presence of AWGN, the matched filter maximizes the SNR by exploiting the known characteristics
of the transmitted signal. By maximizing the SNR, the matched filter improves the reliability and
accuracy of signal detection, enabling efficient communication and signal processing in noisy
environments.

find the bandwidth to transmitting 120bps by QPSK,4 -ary FSK ,8-ary PSK and 16 -ary QASK
find the bandwidth to transmitting 120bps by QPSK,FSK To find the bandwidth required for transmitting
data at a rate of 120 bps using different modulation schemes (QPSK, 4-ary FSK, 8-ary PSK, and 16-ary
QAM), we need to consider the symbol rates and the bandwidth formulas for each modulation scheme.

QPSK (Quadrature Phase Shift Keying):

In QPSK, each symbol represents 2 bits of information. Therefore, the symbol rate required to achieve a
data rate of 120 bps is 60 symbols per second (120 bps / 2 bits per symbol = 60 symbols/s).
The bandwidth required for QPSK can be calculated using the formula:
Bandwidth_QPSK = Symbol Rate × (1 + Roll-off Factor)

Assuming a roll-off factor of 0.35 (commonly used value), we can calculate the bandwidth:

Bandwidth_QPSK = 60 symbols/s × (1 + 0.35)

4-ary FSK (Frequency Shift Keying):

In 4-ary FSK, each symbol represents 2 bits of information. Therefore, the symbol rate required to
achieve a data rate of 120 bps is also 60 symbols per second (120 bps / 2 bits per symbol = 60 symbols/s).
The bandwidth required for 4-ary FSK can be calculated using the formula:

Bandwidth_4FSK = Symbol Rate × (1 + Roll-off Factor)

Assuming the same roll-off factor of 0.35, we can calculate the bandwidth:

Bandwidth_4FSK = 60 symbols/s × (1 + 0.35)

8-ary PSK (Phase Shift Keying):

In 8-ary PSK, each symbol represents 3 bits of information. Therefore, the symbol rate required to
achieve a data rate of 120 bps is 40 symbols per second (120 bps / 3 bits per symbol = 40 symbols/s).
The bandwidth required for 8-ary PSK can be calculated using the formula:

Bandwidth_8PSK = Symbol Rate × (1 + Roll-off Factor)

Assuming the same roll-off factor of 0.35, we can calculate the bandwidth:

Bandwidth_8PSK = 40 symbols/s × (1 + 0.35)

16-ary QAM (Quadrature Amplitude Modulation):

In 16-ary QAM, each symbol represents 4 bits of information. Therefore, the symbol rate required to
achieve a data rate of 120 bps is 30 symbols per second (120 bps / 4 bits per symbol = 30 symbols/s).
The bandwidth required for 16-ary QAM can be calculated using the formula:

Bandwidth_16QAM = Symbol Rate × (1 + Roll-off Factor)

Assuming the same roll-off factor of 0.35, we can calculate the bandwidth:

Bandwidth_16QAM = 30 symbols/s × (1 + 0.35)

By substituting the values into the formulas and performing the calculations, you can determine the
bandwidth required for transmitting at 120 bps using QPSK, 4-ary FSK, 8-ary PSK, and 16-ary QAM.
Question:-Calculate maximum capacity of gaussian channel with BW is 3khz and SNR 30dB,if
bandwidth is double the calculate new channel capacity
Solution:-
To calculate the maximum capacity of a Gaussian channel with a given bandwidth of 3 kHz and a signal-
to-noise ratio (SNR) of 30 dB, we can use Shannon's Channel Capacity formula:

C = B * log2(1 + SNR)

Where:
C is the channel capacity in bits per second (bps).
B is the bandwidth in hertz (Hz).
SNR is the signal-to-noise ratio (unitless).

Using the provided values, we can calculate the initial channel capacity:

C_initial = 3,000 Hz * log2(1 + 10^(30/10))

Let's perform the calculation:

C_initial = 3,000 Hz * log2(1 + 10^(3))

C_initial = 3,000 Hz * log2(1 + 1,000)

C_initial = 3,000 Hz * log2(1,001)

C_initial ≈ 3,000 Hz * 9.9658

C_initial ≈ 29,897.4 bps

Therefore, the maximum capacity of the Gaussian channel with a bandwidth of 3 kHz and an SNR of 30
dB is approximately 29,897.4 bps.

If the bandwidth is doubled, the new channel capacity can be calculated using the same formula but with
the doubled bandwidth:

C_new = 6,000 Hz * log2(1 + 10^(30/10))

Let's calculate the new channel capacity:

C_new = 6,000 Hz * log2(1 + 10^(3))

C_new = 6,000 Hz * log2(1 + 1,000)

C_new = 6,000 Hz * log2(1,001)

C_new ≈ 6,000 Hz * 9.9658

C_new ≈ 59,785.2 bps

Therefore, if the bandwidth is doubled to 6 kHz, the new channel capacity would be approximately
59,785.2 bps.

Implement Cyclic encoder and decoder for (7,4) Using generator polynomial g(x)=x3+x2+1 /
D3+D2+1
Solution:
Let consider message is 1001
find 3 bit HRC and 3 bit checksum for data 101011001111 and show that these code can
detect 3 continuous bit errors.

Solution:To find a 3-bit Hamming Redundancy Check (HRC) code and a 3-bit checksum for the given
data, we can apply the following steps:
Step 1: Generate the HRC (Hamming Redundancy Check) code:

Start with the original data: 101011001111.

Determine the number of parity bits required. In this case, we need to find
the smallest number, x, that satisfies the equation 2^x ≥ m + x + 1, where m
is the number of data bits. Here, m = 12, so we need to find the smallest x
such that 2^x ≥ 12 + x + 1.
Checking the values of x, we find that x = 4 satisfies the equation (2^4 = 16 ≥
12 + 4 + 1).
Insert x number of 0s in the data to make space for the parity bits. The resulting data becomes: 1010 1100
1111 0000.
Determine the positions of the parity bits. In a 3-bit HRC, the positions are 1, 2, 4, 8.
Calculate the values of the parity bits. For each parity bit position, calculate the parity of the
corresponding bits. Place the calculated parity bits in their respective positions.
Parity bit 1: Calculate the parity for all bits whose position has a 1 in the least significant bit (LSB) of
their binary representation (1, 3, 5, 7, 9, 11). The parity bit 1 is placed at position 1.
Parity bit 1 = 1 0 1 1 1 1 = 1.
Parity bit 2: Calculate the parity for all bits whose position has a 1 in the second bit from the right (2, 3, 6,
7, 10, 11). The parity bit 2 is placed at position 2.
Parity bit 2 = 0 0 1 1 1 1 = 0.
Parity bit 4: Calculate the parity for all bits whose position has a 1 in the third bit from the right (4, 5, 6,
7, 12). The parity bit 4 is placed at position 4.
Parity bit 4 = 0 1 1 1 0 = 0.
Parity bit 8: Calculate the parity for all bits whose position has a 1 in the fourth bit from the right (8, 9,
10, 11, 12). The parity bit 8 is placed at position 8.
Parity bit 8 = 1 1 1 1 0 = 0.
Insert the calculated parity bits into their respective positions in the data.
The HRC code for the data 101011001111 is 1000101001110000.
Step 2: Generate the 3-bit checksum:

Split the data into chunks of 3 bits each: 101 011 001 111.
Calculate the sum of each chunk and take the modulo 2 of the sum.
Sum of 101 = 1 + 0 + 1 = 2.
Sum of 011 = 0 + 1 + 1 = 2.
Sum of 001 = 0 + 0 + 1 = 1.
Sum of 111 = 1 + 1 + 1 = 3.
Take the modulo 2 of each sum:
2 mod 2 = 0.
2 mod 2 = 0.
1 mod 2 = 1.
3 mod 2 = 1.
Concatenate the results to form the checksum: 0011.
To show that these codes can detect 3 consecutive bit errors, let's introduce 3 errors in consecutive
positions:
Original data: 101011001111
Modified data: 101000001111 (Errors introduced in positions 4, 5, and 6)

For the HRC code:

Original HRC: 1000 1010 0111 0000
Modified HRC: 1000 0000 0111 0000

The HRC code detects the error since the modified HRC is different from the original HRC.

For the checksum:

Original checksum: 0011
Modified checksum: 0001

The checksum detects the error since the modified checksum is different from the original checksum.

In both cases, the codes can detect the introduced 3 consecutive bit errors.

Derive PDF of BFSK sketch power spectrum to find the bandwidth

To derive the probability density function (PDF) of Binary Frequency Shift Keying (BFSK) and find its
bandwidth, we need to consider the following steps:

Define the BFSK Signal:

BFSK is a modulation technique where two different frequencies, denoted as f1 and f2, are used to
represent binary symbols (0 and 1).
Let's assume that f1 corresponds to binary symbol 0 and f2 corresponds to binary symbol 1.
Probability Density Function (PDF):

Assuming equal probability for both binary symbols (p0 = p1 = 0.5), the PDF of BFSK can be
represented as:
PDF = (p0 * δ(f - f1)) + (p1 * δ(f - f2))
where δ represents the Dirac delta function, and f is the frequency variable.
Power Spectrum:

The power spectrum of a signal can be obtained by taking the Fourier Transform of the autocorrelation
function.
For BFSK, the autocorrelation function is given by:
Rxx(τ) = E[s(t)s(t - τ)]
where E[.] denotes the expectation operator and s(t) represents the BFSK signal.
Since the frequencies for binary symbols are orthogonal, the autocorrelation between different
frequencies is zero.
Rxx(τ) = (A^2/2) * δ(τ) + (A^2/2) * δ(τ - T)
where A is the amplitude of the signal and T is the symbol duration (time period for one symbol).
Taking the Fourier Transform of the autocorrelation function, we obtain the power spectrum:
Sxx(f) = |F[Rxx(τ)]|^2
where F[.] denotes the Fourier Transform operator.
Substituting the autocorrelation function into the power spectrum equation:
Sxx(f) = |F[(A^2/2) * δ(τ) + (A^2/2) * δ(τ - T)]|^2
= |(A^2/2) * 1 + (A^2/2) * e^(-j2πfT)|^2
= |A^2/2 + (A^2/2) * e^(-j2πfT)|^2
= |A^2/2|^2 + |(A^2/2) * e^(-j2πfT)|^2 + 2 * Re{(A^2/2) * (A^2/2) * e^(-j2πfT)}
= (A^2/2)^2 + (A^2/2)^2 * |e^(-j2πfT)|^2 + 2 * Re{(A^2/2)^2 * e^(-j2πfT)}
= (A^2/2)^2 + (A^2/2)^2 * 1 + 2 * Re{(A^2/2)^2 * e^(-j2πfT)}
= (A^2/2)^2 + (A^2/2)^2 + 2 * Re{(A^2/2)^2 * e^(-j2πfT)}
= A^4/4 + A^4/4 + 2 * Re{A^4/4 * e^(-j2πfT)}
= A^4/2 + A^4/2 * cos(2πfT)
Bandwidth:

The bandwidth of a signal is usually defined as the frequency range where the power spectral density is
significant (non-negligible).
In the case of BFSK, the significant frequency range is determined by the cosine term in the power
spectrum equation.
The bandwidth can be approximated by considering the frequency range where the cosine term is non-
zero.
Since the cosine function oscillates between -1 and 1, the bandwidth can be determined by finding the
frequency range where the cosine term is larger than a certain threshold.
Typically, a threshold of 0.5 is used to determine the bandwidth.
So, we need to find the frequency range where |cos(2πfT)| > 0.5.
Solving this inequality, we get:
-0.5 < cos(2πfT) < 0.5
-0.5 < cos(2πfT) and cos(2πfT) < 0.5
The frequency range can be determined by solving the above inequalities.
Please note that the above derivation provides a general understanding of the PDF and bandwidth
calculation for BFSK. The specific values of amplitudes, symbol duration, and threshold will depend on
the parameters of your BFSK system.

compare NRZ UNIPOLAR,NRZ POLAR,NRZ MANCHESTER AND NRZ AMI IN

TERMS OF PARAMETERS

compare NRZ Unipolar, NRZ Polar, NRZ Manchester, and NRZ AMI in terms of different parameters:

Voltage Levels:

NRZ Unipolar: Uses a single voltage level to represent binary data (e.g., high level for "1" and zero level
for "0").
NRZ Polar: Uses two voltage levels, typically positive and negative, to represent binary data.
NRZ Manchester: Uses two voltage levels, with transitions in the middle of each bit period to represent
binary data.
NRZ AMI: Uses three voltage levels, including positive, negative, and zero levels, to represent binary
data.
Bandwidth Efficiency:

NRZ Unipolar: Low bandwidth efficiency as it does not utilize transitions for synchronization.
NRZ Polar: Moderate bandwidth efficiency due to the use of two voltage levels.
NRZ Manchester: Lower bandwidth efficiency compared to NRZ Polar due to the requirement of frequent
transitions for synchronization.
NRZ AMI: Moderate bandwidth efficiency due to the use of three voltage levels.
Synchronization:

NRZ Unipolar: Prone to synchronization errors due to the lack of frequent transitions.
NRZ Polar: Synchronization can be challenging, especially during long sequences of "0" bits.
NRZ Manchester: Provides self-clocking and better synchronization capabilities due to the frequent
transitions.
NRZ AMI: Offers good synchronization capabilities as transitions are present for each bit.
DC Component:

NRZ Unipolar: Has a significant DC component due to the use of a constant voltage level.
NRZ Polar: Eliminates the DC component by using two opposite voltage levels.
NRZ Manchester: Eliminates the DC component due to the frequent transitions.
NRZ AMI: Eliminates the DC component by alternating between positive and negative voltage levels.
Error Detection:

NRZ Unipolar: Limited error detection capability.

NRZ Polar: Limited error detection capability.
NRZ Manchester: Limited error detection capability.
NRZ AMI: Provides a reliable means for error detection due to the use of alternating voltage levels.
Bandwidth:

NRZ Unipolar: Requires the lowest bandwidth among the four schemes.
NRZ Polar: Requires a moderate bandwidth.
NRZ Manchester: Requires a higher bandwidth compared to NRZ Polar due to the presence of frequent
transitions.
NRZ AMI: Requires a moderate bandwidth similar to NRZ Polar.
Example to draw different line code

Explain advantages of offset QPSK over QPSK

Solution :-
Offset Quadrature Phase Shift Keying (OQPSK) is a modified version of Quadrature Phase Shift Keying
(QPSK) that offers several advantages over traditional QPSK modulation. Here are the advantages of
OQPSK over QPSK:

1.Reduced Peak-to-Average Power Ratio (PAPR):

OQPSK exhibits a lower PAPR compared to QPSK.
In QPSK, the constellation points can align in such a way that the peak power of the transmitted signal is
significantly higher than the average power.
OQPSK mitigates this issue by offsetting the phase of the in-phase and quadrature carriers, resulting in a
reduced PAPR.
The lower PAPR simplifies power amplifier design, improves power efficiency, and reduces the
likelihood of nonlinear distortion.

2.Improved Spectral Efficiency:

OQPSK achieves higher spectral efficiency compared to QPSK.
QPSK transmits two bits per symbol, while OQPSK transmits one bit per symbol.
This is achieved by using the phase transitions in OQPSK to represent the bit transitions, resulting in a
higher data rate within the same bandwidth.

3.Enhanced Phase Continuity:

OQPSK maintains phase continuity during bit transitions.
In QPSK, the phase can change abruptly between adjacent symbols, which can lead to spectral splatter
and increased interference with neighboring channels.
OQPSK ensures a smooth phase transition, reducing interference and improving signal quality.

4.Simplified Receiver Design:

OQPSK simplifies the receiver design compared to QPSK.
In QPSK, the receiver needs to recover both the in-phase and quadrature components separately and
synchronize them accurately.
OQPSK, on the other hand, only requires the recovery of the phase transitions, which simplifies the
demodulation process and reduces the complexity of the receiver.

5.Improved Error Performance:

OQPSK typically offers better error performance in the presence of phase noise and frequency offset
compared to QPSK.
The phase transitions in OQPSK provide more robustness against phase errors, making it more suitable
for scenarios where phase distortions are present.
Note -for offset QPSK draw waveform after Tb delay/offset

find variance of huffman coding and shannon fano code example for 0.,0.1,0.4,0.2 and 0.1

A discrete memoryless source has symbols x1, x2, x3, x4, x5 with
probabilities of 0.4, 0.2, 0.1, 0.2, 0.1 respectively. Construct a Shannon
Solution : Step 1 : Arrange the given probabilities in descending order. given
probabilities P1 = 0.4, P2 = 0.2 ,P3 = 0.1 , P4 = 0.2 , P5 = 0.1
Probabilities in descending order
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13Marks_compressed.pdf