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EEE2329: DIGITAL COMMUNICATION CLASSNOTES

Analog-to-digital converters (ADCs) and digital-to-analog converters (DACs) are two types of data conv
erters that are used to convert between analog and digital signals.
An ADC converts an analog signal, such as a sound wave or a video signal, into a digital signal, which
can be stored, processed, and transmitted by digital devices. The ADC does this by sampling the
analog signal at regular intervals and then converting each sample into a digital number. The number
of bits used to represent each sample determines the resolution of the ADC. A higher resolution ADC
will be able to represent the analog signal more accurately.
A DAC converts a digital signal back into an analog signal. This is useful for applications such as audio
output, video display, and control systems. The DAC does this by taking the digital signal and
converting it into a voltage or current that is proportional to the digital value.
The basic principles of ADCs and DACs are as follows:
 ADC: The ADC first samples the analog signal at a regular interval. The sampling rate is the
number of samples taken per second. The Nyquist theorem states that the sampling rate
must be at least twice the highest frequency component of the analog signal in order to
avoid aliasing.
 Quantization: The ADC then converts each sample into a digital number. This is done by
dividing the range of the analog signal into a number of discrete levels. The number of levels
determines the resolution of the ADC.
 Encoding: The digital numbers are then encoded into a binary code. This is done by using a
comparator to compare each sample to the levels of the analog signal.
The basic principles of DACs are as follows:
 Decoding: The DAC first decodes the binary code into a set of digital numbers.
 Ramp generation: The DAC then generates a ramp voltage or current that is proportional to
the digital numbers.
 Interpolation: The DAC then interpolates between the ramp levels to produce an analog
signal that is proportional to the digital numbers.
The performance of an ADC or DAC is determined by its sampling rate, resolution, and accuracy. The
sampling rate must be high enough to avoid aliasing. The resolution determines the number of
discrete levels that the analog signal can be represented by. The accuracy determines how close the
digital representation of the analog signal is to the actual analog signal.
ADCs and DACs are used in a wide variety of applications, including:
 Digital audio
 Digital video
 Data acquisition
 Instrumentation
 Telecommunications
 Control systems
Digital signals are signals that are represented by discrete values. This means that the signal is not
continuous, but instead is made up of a series of discrete points. Quantization is the process of
converting an analog signal into a digital signal. This is done by dividing the range of the analog signal
into a finite number of levels, and then assigning a discrete value to each level.
Digitization is the process of converting a digital signal into a sequence of binary digits (bits). This is
done by representing each level of the quantized signal with a unique binary number.
The transmission of digital signals can be done in a variety of ways. One common way is to use a
digital modulation scheme. A digital modulation scheme is a method of encoding digital data onto a
carrier signal. The most common digital modulation schemes are binary amplitude modulation (PAM),
binary frequency modulation (FM), and binary phase modulation (PM).
The recognition of digital signals can be done using a variety of techniques. One common technique is
to use a digital filter. A digital filter is a circuit that can be used to extract specific features from a
digital signal. Another common technique is to use a pattern recognition algorithm. A pattern
recognition algorithm is a computer program that can be used to identify patterns in data.
Here are some of the techniques used in digital signal quantization, digitization, transmission, and
recognition:
 Quantization:

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o Uniform quantization: This is the simplest type of quantization, and it divides the
range of the analog signal into a equal number of levels.
o Non-uniform quantization: This type of quantization allows for different levels to
have different widths. This can be used to improve the accuracy of the quantization
at the expense of increased complexity.
 Digitization:
o Binary quantization: This is the most common type of digitization, and it uses binary
numbers to represent the levels of the quantized signal.
o Multi-level quantization: This type of digitization uses more than two levels to
represent the quantized signal. This can be used to improve the accuracy of the
digitization at the expense of increased complexity.
 Transmission:
o Digital modulation: This is the process of encoding digital data onto a carrier signal.
The most common digital modulation schemes are binary amplitude modulation
(PAM), binary frequency modulation (FM), and binary phase modulation (PM).
o Error correction coding: This is used to improve the reliability of digital transmission
by detecting and correcting errors that occur during transmission.
 Recognition:
o Digital filtering: This is used to extract specific features from a digital signal.
o Pattern recognition algorithms: These are used to identify patterns in data.
These are just a few of the techniques used in digital signal quantization, digitization, transmission,
and recognition. The specific techniques that are used will depend on the application.
here are many techniques for detecting digital signals from noise. Some of the most common
techniques include:
 Filtering: This involves passing the signal through a filter that removes the noise frequencies.
There are many different types of filters, such as low-pass filters, high-pass filters, and
bandpass filters.
 Thresholding: This involves setting a threshold value and declaring any signal values above
the threshold to be the signal and any values below the threshold to be noise.
 Matched filtering: This is a technique that is specifically designed for detecting signals that
are known in advance. The matched filter is designed to maximize the signal-to-noise ratio
(SNR) of the detected signal.
 Adaptive filtering: This is a technique that automatically adjusts the filter parameters to
optimize the SNR of the detected signal.
 Statistical methods: This involves using statistical methods to distinguish between signal and
noise. Some common statistical methods include the mean, variance, and autocorrelation.
The best technique to use for detecting a digital signal from noise will depend on the specific
application. The type of signal, the amount of noise, and the desired SNR are all important factors to
consider.
Here are some additional details about each of the techniques mentioned above:
 Filtering: Filtering is a very common technique for removing noise from signals. There are
many different types of filters, each with its own advantages and disadvantages. Low-pass
filters remove high-frequency noise, while high-pass filters remove low-frequency noise.
Bandpass filters remove frequencies outside of a specified band.
 Thresholding: Thresholding is a simple but effective technique for detecting signals from
noise. The threshold value is typically chosen based on the expected SNR of the signal.
Values above the threshold are declared to be the signal, while values below the threshold
are declared to be noise.
 Matched filtering: Matched filtering is a technique that is specifically designed for detecting
signals that are known in advance. The matched filter is designed to maximize the SNR of the
detected signal by taking into account the known properties of the signal.
 Adaptive filtering: Adaptive filtering is a technique that automatically adjusts the filter
parameters to optimize the SNR of the detected signal. This can be useful in situations where
the noise characteristics are changing over time.
 Statistical methods: Statistical methods can be used to distinguish between signal and noise
by exploiting the different statistical properties of the two. For example, the mean and

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variance of the signal and noise can be used to determine a threshold value for declaring a
signal.
The detection techniques of digital signal from noise can be classified into two main categories:
baseband analysis and filtering and pulse-shaping.
Baseband analysis techniques are used to detect the signal in the baseband, which is the frequency
band where the signal is originally transmitted. These techniques typically involve demodulating the
signal to recover the original baseband signal, and then filtering the signal to remove the noise.
One common baseband analysis technique is the matched filter. The matched filter is a filter that is
designed to maximize the signal-to-noise ratio (SNR) of the detected signal. The matched filter is
matched to the expected shape of the signal, and it amplifies the signal and attenuates the noise.
Filtering and pulse-shaping techniques are used to improve the SNR of the signal by filtering out the
noise. These techniques typically involve passing the signal through a filter that has a narrow
bandwidth, which helps to reject the noise.
One common filtering technique is the low-pass filter. The low-pass filter passes signals with
frequencies below a certain cutoff frequency, and it attenuates signals with frequencies above the
cutoff frequency. This helps to remove the noise, which typically has a wider bandwidth than the
signal.
Inter-symbol interference (ISI) is a phenomenon that occurs when the pulses of a digital signal overlap
in time. This can be caused by the limited bandwidth of the transmission channel, or by the use of a
non-ideal filter. ISI can degrade the SNR of the signal, and it can make it difficult to detect the signal.
There are a number of techniques that can be used to mitigate ISI. One common technique is to use a
pulse-shaping filter that has a sharp rise and fall time. This helps to minimize the amount of overlap
between the pulses. Another technique is to use a coding scheme that is designed to be resistant to
ISI.
The choice of detection technique depends on the specific application. In general, baseband analysis
techniques are more effective than filtering and pulse-shaping techniques, but they are also more
complex. The use of ISI mitigation techniques is also application-specific.
Here are some additional details about each of the techniques mentioned above:
 Matched filter: The matched filter is a filter that is designed to maximize the SNR of the
detected signal. The matched filter is matched to the expected shape of the signal, and it
amplifies the signal and attenuates the noise. The matched filter is the optimal detector for a
signal in white noise.
 Low-pass filter: A low-pass filter is a filter that passes signals with frequencies below a certain
cutoff frequency, and it attenuates signals with frequencies above the cutoff frequency. Low-
pass filters are commonly used to remove noise from signals.
 Inter-symbol interference (ISI): ISI is a phenomenon that occurs when the pulses of a digital
signal overlap in time. This can be caused by the limited bandwidth of the transmission
channel, or by the use of a non-ideal filter. ISI can degrade the SNR of the signal, and it can
make it difficult to detect the signal.
The performance of various digital modulation schemes is typically measured in terms of the bit error
rate (BER), which is the probability of a bit being received incorrectly. The BER depends on a number
of factors, including the signal-to-noise ratio (SNR), the modulation scheme, and the channel
characteristics.
In general, the more complex the modulation scheme, the lower the BER. This is because a more
complex modulation scheme can transmit more information per symbol, which makes it more
resistant to noise. However, a more complex modulation scheme is also more difficult to implement
and requires more bandwidth.
Here is a brief overview of some of the most common digital modulation schemes:
 Amplitude shift keying (ASK) is the simplest digital modulation scheme. It transmits data by
changing the amplitude of the carrier signal. ASK has a low BER, but it also has a low
bandwidth efficiency.
 Phase shift keying (PSK) is a more complex modulation scheme than ASK. It transmits data by
changing the phase of the carrier signal. PSK has a higher BER than ASK, but it also has a
higher bandwidth efficiency.
 Frequency shift keying (FSK) is another more complex modulation scheme than ASK. It
transmits data by changing the frequency of the carrier signal. FSK has a higher BER than ASK
and PSK, but it also has a higher bandwidth efficiency.

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 Quadrature amplitude modulation (QAM) is the most complex digital modulation scheme. It
transmits data by changing both the amplitude and phase of the carrier signal. QAM has the
lowest BER of all the modulation schemes, but it also requires the most bandwidth.
The best modulation scheme for a particular application will depend on a number of factors, including
the required BER, the available bandwidth, and the cost of implementation.
Here is a table summarizing the performance of various digital modulation schemes in terms of BER
and bandwidth efficiency:
Modulation Scheme BER Bandwidth Efficiency

ASK High Low

PSK Medium Medium

FSK Medium High

QAM Low High

In general, the modulation scheme with the lowest BER will be the best choice for applications where
a high degree of accuracy is required. However, the modulation scheme with the highest bandwidth
efficiency will be the best choice for applications where the available bandwidth is limited.
Ultimately, the best way to choose a modulation scheme is to consider the specific requirements of
the application.
Sure, here are some solved examples on digitization technique:
 Digitizing an image: To digitize an image, you can use a scanner or a digital camera. The
scanner will convert the image into a digital file, which can then be stored on a computer or
other device. The digital camera will also convert the image into a digital file, but it can also
store the image on a memory card or other storage device.
 Digitizing audio: To digitize audio, you can use a microphone and a computer. The
microphone will convert the audio signal into a digital signal, which can then be stored on
the computer. There are also many software programs that can be used to digitize audio,
such as Audacity and GarageBand.
 Digitizing video: To digitize video, you can use a video camera and a computer. The video
camera will convert the video signal into a digital signal, which can then be stored on the
computer. There are also many software programs that can be used to digitize video, such as
Adobe Premiere Pro and Final Cut Pro.
 Digitizing text: To digitize text, you can use a scanner or a OCR (optical character recognition)
software program. The scanner will convert the text into a digital image, which can then be
processed by the OCR software program to convert it into text that can be edited and stored
on a computer.
 Digitizing data: To digitize data, you can use a variety of methods, such as sensors,
transducers, and data acquisition software. The sensors will convert the physical quantity
into an electrical signal, which can then be processed by the transducers and data acquisition
software to convert it into a digital signal that can be stored on a computer.
These are just a few examples of digitization techniques. There are many other techniques that can be
used to digitize different types of data.
The geometric representation of signals is a way of representing signals as points in a vector space.
This can be done by using a set of basis functions, which are waveforms that are orthogonal to each
other. Any signal can then be represented as a linear combination of these basis functions.

The geometric representation of signals has several advantages. First, it allows us to visualize signals
and their relationships to each other. Second, it can be used to simplify the analysis of signals. Third, it
can be used to develop efficient algorithms for processing signals.

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To understand the geometric representation of signals, let's consider a simple example. Suppose we
have a signal that is a sine wave. We can represent this signal as a linear combination of two basis
functions: the sine function and the cosine function. The coefficients of these basis functions are the
amplitude and phase of the sine wave.

In general, any signal can be represented as a linear combination of a set of basis functions. The
number of basis functions needed to represent a signal depends on the complexity of the signal. For
example, a simple signal may only need a few basis functions to represent it, while a complex signal
may need many basis functions.

The geometric representation of signals is a powerful tool that can be used to analyze and process
signals. It is used in many different areas, including telecommunications, signal processing, and image
processing.

Here are some of the applications of the geometric representation of signals:

Signal analysis: The geometric representation of signals can be used to analyze the properties of
signals, such as their frequency content, energy, and power.
Signal processing: The geometric representation of signals can be used to process signals, such as
filtering, modulation, and demodulation.
Image processing: The geometric representation of signals can be used to process images, such as
edge detection, noise removal, and image compression.
Communication systems: The geometric representation of signals is used in communication systems
to transmit and receive signals.
Radar and sonar: The geometric representation of signals is used in radar and sonar systems to detect
and track objects.

 Block codes work on fixed-size blocks of bits or symbols of predetermined size. The encoder
takes a block of data bits and produces a codeword of the same size, with some of the bits
being parity bits that are added to help detect or correct errors. Convolutional codes, on the
other hand, work on bit or symbol streams of arbitrary length. The encoder takes a sequence
of data bits and produces a sequence of code bits, where the output at each time step
depends on the input bits at a few previous time steps.
 Block codes are typically easier to implement and decode than convolutional codes. This is
because block codes can be decoded using a simple algorithm that checks for the presence of
errors in the codeword. Convolutional codes, on the other hand, require more complex
decoding algorithms, such as the Viterbi algorithm.
 Block codes typically have lower error correction capability than convolutional codes. This is
because block codes only add parity bits to the data bits, while convolutional codes can add
redundancy to the data bits in a more sophisticated way.
 Block codes typically have higher latency than convolutional codes. This is because block
codes need to wait until the entire block of data bits has been received before they can start
decoding. Convolutional codes, on the other hand, can start decoding as soon as the first few
data bits have been received.
In general, block codes are a good choice for applications where simplicity and low latency are
important, such as in digital audio and video transmission. Convolutional codes are a good choice for
applications where high error correction capability is important, such as in satellite communications.
Here is a table summarizing the key differences between block codes and convolutional codes:
Feature Block Codes Convolutional Codes

Data block size Fixed Variable

Encoding complexity Low High

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Decoding complexity Low High

Error correction capability Low High

Latency High Low

The characteristic parameters of discrete information sources are the statistical properties that
describe the distribution of the symbols that the source generates. These parameters are used in
information theory to quantify the amount of information that can be transmitted by the source and
to design efficient communication systems.
The most important characteristic parameter of a discrete information source is the entropy. The
entropy of a source is a measure of the average uncertainty associated with the next symbol that the
source will generate. A higher entropy indicates that the source is more unpredictable and therefore
contains more information.
Other important characteristic parameters of discrete information sources include:
 The probability distribution of the symbols. This is a list of the probabilities that each symbol
will be generated by the source.
 The autocorrelation function. This is a measure of the statistical dependence between the
current symbol and previous symbols generated by the source.
 The power spectrum. This is a measure of the frequency content of the symbols generated
by the source.
These parameters can be estimated from a sample of the symbols generated by the source. The
estimation process is typically based on maximum likelihood estimation or Bayesian estimation.
The characteristic parameters of discrete information sources are used in a variety of applications,
including:
 Data compression. The entropy of a source can be used to design efficient data compression
algorithms.
 Error correction coding. The autocorrelation function of a source can be used to design error
correction codes that are more robust to errors.
 Channel capacity. The capacity of a communication channel is the maximum rate at which
information can be transmitted over the channel without errors. The capacity of the channel
can be expressed in terms of the entropy of the source and the characteristics of the
channel.

There are many detection techniques for digital signals from noise in baseband analyses. Some of the
most common techniques include:
 Matched filtering: This is a technique that uses a filter that is matched to the transmitted
signal to maximize the signal-to-noise ratio (SNR).
 Decision feedback equalization: This is a technique that uses a feedback loop to correct for
the distortion caused by the channel.
 Maximum likelihood sequence estimation: This is a technique that uses the maximum
likelihood principle to determine the most likely sequence of transmitted symbols.
 Bit-error rate (BER) estimation: This is a technique that estimates the probability of a bit
error occurring.
 Constellation diagram: This is a graphical representation of the possible transmitted symbols.
The choice of detection technique depends on the specific application and the characteristics of the
noise. For example, matched filtering is a good choice for applications where the noise is white and
Gaussian. Decision feedback equalization is a good choice for applications where the noise is colored
or non-Gaussian. Maximum likelihood sequence estimation is a good choice for applications where
the noise is bursty or non-stationary.
Here is a brief overview of each of these techniques:
 Matched filtering: Matched filtering is a technique that uses a filter that is matched to the
transmitted signal to maximize the SNR. The filter is designed to have a response that is

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equal to the complex conjugate of the transmitted signal. This means that the filter will pass
the transmitted signal with minimal attenuation and will reject the noise.
 Decision feedback equalization: Decision feedback equalization (DFE) is a technique that uses
a feedback loop to correct for the distortion caused by the channel. The DFE works by first
making a decision about the transmitted symbol. This decision is then used to correct the
received signal for the distortion caused by the channel. The corrected signal is then used to
make a new decision about the transmitted symbol. This process is repeated until the desired
accuracy is achieved.
 Maximum likelihood sequence estimation: Maximum likelihood sequence estimation (MLSE)
is a technique that uses the maximum likelihood principle to determine the most likely
sequence of transmitted symbols. The MLSE works by considering all possible sequences of
transmitted symbols and choosing the sequence that is most likely to have produced the
received signal.
 Bit-error rate (BER) estimation: Bit-error rate (BER) estimation is a technique that estimates
the probability of a bit error occurring. The BER is estimated by sending a known sequence of
bits and then measuring the number of bit errors that occur. The BER can then be used to
determine the performance of the detection technique.
 Constellation diagram: A constellation diagram is a graphical representation of the possible
transmitted symbols. The constellation diagram shows the amplitude and phase of each
possible symbol. The constellation diagram can be used to visualize the effect of noise on the
transmitted signal.
These are just a few of the many detection techniques that can be used for digital signals from noise
in baseband analyses. The choice of detection technique depends on the specific application and the
characteristics of the noise.
An analog-to-digital converter (ADC) is a device that converts an analog signal, such as a sound picked
up by a microphone or light entering a digital camera, into a digital signal. The operation of an ADC
can be divided into three steps:
1. Sampling: The analog signal is sampled at regular intervals. This means that the value of the
signal is captured at a specific point in time. The sampling rate is the number of times the
signal is sampled per second. The Nyquist theorem states that the sampling rate must be at
least twice the highest frequency component of the analog signal in order to avoid aliasing.
2. Quantization: The sampled signal is then quantized, which means that it is divided into a
finite number of levels. The number of levels is determined by the resolution of the ADC. A
higher resolution ADC will be able to represent the analog signal more accurately.
3. Coding: The quantized signal is then coded into a digital format. This is usually done by using
a binary code. The number of bits used to represent the signal is determined by the
resolution of the ADC. A higher resolution ADC will require more bits to represent the signal.
The overall accuracy of an ADC is determined by its sampling rate, resolution, and the noise
introduced by the ADC itself.
There are many different types of ADCs, each with its own advantages and disadvantages. Some of
the most common types of ADCs include:
 Flash ADC: This is the fastest type of ADC, but it is also the most expensive. It works by
comparing the analog signal to a series of reference voltages.
 Successive approximation ADC: This is a slower type of ADC, but it is less expensive than a
flash ADC. It works by gradually increasing or decreasing the value of a digital output until it
matches the analog input.
 Delta-sigma ADC: This is a very accurate type of ADC, but it is also the slowest. It works by
comparing the analog input to a previous value of the digital output.
 Pipelined ADC: This is a type of ADC that combines the speed of a flash ADC with the
accuracy of a delta-sigma ADC. It works by dividing the conversion process into a series of
stages, each of which is less complex than a flash ADC.
ADCs are used in a wide variety of applications, including:
 Digital audio and video
 Data acquisition
 Medical imaging
 Telecommunications

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 Industrial control
 Scientific measurement
Broadband systems use modulation techniques to reduce the effect of noise in the environment.
Broadband transmission employs multiple channel unidirectional transmission using a combination of
phase and amplitude modulation.

Baseband is a digital signal transmitted on the medium using one of the signal codes like NRZ, RZ
Manchester biphase-M code, etc. called baseband tra These are the following differences between
Broadband and Baseband transmission.

Baseband transmission:nsmission.
Digital signaling.
Frequency division multiplexing is not possible.
Baseband is the bi-directional transmission.
A short-distance signal traveling.
The entire bandwidth is for single signal transmission.
Example: Ethernet is using Basebands for LAN.
Broadband transmission:

Analog signaling.
The transmission of data is unidirectional.
Signal traveling distance is long.
Frequency division multiplexing is possible.
Simultaneous transmission of multiple signals over different frequencies.
Example: Used to transmit cable TV to premises.
S. No Basis of Comparison Baseband Transmission Broadband Transmission
1. Type of Signal In baseband transmission, the type of signaling used is digital. In
broadband transmission, the type of signaling used is analog.
2. Direction Type Baseband Transmission is bidirectional in nature. Broadband
Transmission is unidirectional in nature.
3. Signal Transmission The Signal can be sent in both directions. Sending of Signal in
one direction only.
4. Distance covered by the signal Signals can only travel over short distances. For long
distances, attenuation is required. Signals can be traveled over long distances without being
attenuated.
5. Topology It works well with bus topology. It is used with a bus as well as tree
topology.
6. Device used to increase signal strength Repeaters are used to enhance signal strength.
Amplifiers are used to enhance signal strength.
7. Type of Multiplexing used It utilizes Time Division Multiplexing. It utilizes Frequency
Division Multiplexing.
8. Encoding Techniques In baseband transmission, Manchester and Differential
Manchester encoding are used. Only PSK encoding is used.
9. Transfer medium Twisted-pair cables, coaxial cables, and wires are used as a transfer
medium for digital signals in baseband transmission. Broadband signals were sent through optical
fiber cables, coaxial cables, and radio waves.
10. Impedance Baseband transmission has a 50-ohm impedance. Broadband
transmission has a 70-ohm impedance.
11. Data Streams It can only transfer one data stream at a time in bi-directional mode. It
can send multiple signal waves at once but in one direction only.
12. Installation and Maintenance Baseband transmission is easy to install and maintain.
Broadband transmission is difficult to install and maintain.
13. Cost This transmission is cheaper to design. This transmission is expensive to
design.
14. Application Typically seen in Ethernet LAN networks. Typically found in cable and
telephone networks.

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15. Frequency In this, capacity of frequency is less than 100 kHz. In this, capacity of
frequency is higher than 100 kHz.
16. Suitable for It is best for wired networks. It is best for non-wired networks.
17. Structure The structure is very simple, and no special hardware is required. The
structure is complex as it needs unique hardware.

The average signal to quantizer noise power ratio (SQNR) at the input of a Q-step uniform quantizer is
given by the following formula:

SQNR = 6.02 * Q dB
where Q is the number of bits per sample.

The derivation of this formula is as follows.

The average power of the signal is given by:

Ps = a^2 / 2
where a is the half-width of the quantization interval.

The average power of the quantization noise is given by:

Pqn = q^2 / 12
where q is the quantization step size.

The SQNR is then given by:

SQNR = Ps / Pqn = (a^2 / 2) / (q^2 / 12) = 6.02 * Q

Therefore, the average SQNR at the input of a Q-step uniform quantizer is 6.02 dB per bit.

In your case, the input to the quantizer has a uniform probability density function over the interval [-
a, a]. This means that the average signal power is given by:

Ps = a^2 / 2
The quantization step size is given by:

q=a/Q
Therefore, the average SQNR at the input of the quantizer is given by:

SQNR = 6.02 * Q = 6.02 * (a / Q)

This can be written as:

SQNR = 6.02a / Q
In decibels, this is equal to:

SQNRdB = 6.02 * 10 * log10(a / Q)

Filtering and pulse-shaping are two different techniques used to detect digital signals.

Filtering is a technique that uses a filter to remove unwanted noise from the signal. The filter is
designed to pass the desired signal frequencies and reject the noise frequencies.

Pulse-shaping is a technique that changes the shape of the signal to improve its performance. The
signal is shaped to have a narrower bandwidth and a sharper rise and fall time. This helps to reduce
the intersymbol interference (ISI) that can occur in digital communication systems.

Here are some examples of each technique:

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Filtering: A low-pass filter can be used to remove high-frequency noise from a digital signal. This is a
common technique used in audio and video systems.
Pulse-shaping: A raised-cosine filter can be used to shape the pulses in a digital communication
system. This helps to reduce the ISI and improve the signal-to-noise ratio.
The best technique to use depends on the specific application. For example, filtering is often used in
applications where the noise is a major problem, such as audio and video systems. Pulse-shaping is
often used in applications where the bandwidth is limited, such as digital communication systems.

Here is a table summarizing the key differences between filtering and pulse-shaping:

Feature Filtering Pulse-shaping

Purpose Removes noise Improves signal performance
Technique Uses a filter Changes the signal shape
Applications Audio, video, other applications with noise Digital communication systems
Binary Phase Shift Keying (BPSK) is a digital modulation technique that conveys data by changing the
phase of a constant frequency carrier wave. The modulation is accomplished by varying the sine and
cosine inputs at a precise time. It is widely used for wireless LANs, RFID and Bluetooth
communication.
In BPSK, the data signal is represented by two phases of the carrier wave, typically 0 degrees and 180
degrees. A binary 0 is represented by a phase shift of 180 degrees, while a binary 1 is represented by
no phase shift.
The constellation diagram for BPSK is a circle with two points, one at 0 degrees and one at 180
degrees. The distance between the two points is called the symbol spacing. The symbol spacing is
determined by the bit rate of the data signal.
The signal-to-noise ratio (SNR) of BPSK is determined by the symbol spacing and the noise power. The
higher the symbol spacing, the better the SNR.
BPSK is a simple and effective modulation technique that is relatively immune to noise. It is a good
choice for applications where the bandwidth is limited and the noise is not too high.
Here are some of the advantages of BPSK:
 Simple and easy to implement
 Relatively immune to noise
 Low bandwidth requirements
Here are some of the disadvantages of BPSK:
 Can only transmit two bits per symbol
 Not as efficient as other modulation techniques such as quadrature phase shift keying (QPSK)
Overall, BPSK is a good choice for applications where simplicity and robustness are more important
than efficiency. It is widely used in wireless LANs, RFID and Bluetooth communication
Problem: A BPSK signal is transmitted with a bit rate of 1 Mbps. The symbol spacing is 100 kHz. What
is the carrier frequency?
Solution: The carrier frequency is given by:

f_c = symbol spacing * bit rate

Plugging in the values, we get:

f_c = 100 kHz * 1 Mbps = 1 MHz

Therefore, the carrier frequency is 1 MHz.

Problem: A BPSK signal is transmitted with a symbol-to-noise ratio (SNR) of 10 dB. What is the
probability of bit error?
Solution: The probability of bit error for BPSK is given by:

P_e = 1 / (1 + SNR)
Plugging in the value of SNR, we get:

P_e = 1 / (1 + 10 dB) = 1 / 11
Therefore, the probability of bit error is 1/11.

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Problem: A BPSK signal is transmitted over a noisy channel. The received signal is:
r(t) = s(t) + n(t)
where s(t) is the transmitted signal, n(t) is the noise signal, and r(t) is the received signal.

The signal-to-noise ratio (SNR) is given by:

SNR = P_s / P_n

where P_s is the power of the signal and P_n is the power of the noise.

The probability of bit error is given by:

P_e = Q(sqrt(2 * P_n / P_s))

where Q() is the Q-function.

The Q-function is a mathematical function that gives the probability that a standard normal variable
will be less than a certain value.

Problem: A BPSK signal is transmitted over a channel with additive white Gaussian noise (AWGN). The
noise power is -10 dBm. The bit rate is 1 Mbps. What is the minimum carrier-to-noise ratio (CNR)
required to achieve a bit error rate of 10^-5?
Solution: The noise power in watts is given by:

P_n = -10 dBm * 10^-3 watts

The carrier power is given by:

P_c = bit rate * bandwidth

The bandwidth of the BPSK signal is equal to the symbol rate, which is 1 MHz. So, the carrier power is:

P_c = 1 Mbps * 1 MHz = 1 mW

The minimum CNR required to achieve a bit error rate of 10^-5 is given by:

CNR = P_c / P_n = 1 mW / (-10 dBm * 10^-3 watts) = 10^5 dB

Therefore, the minimum CNR required is 10^5 dB.

These are just a few solved problems on Binary Phase Shift Keying (BPSK). There are many other
problems that can be solved using BPS
A constellation diagram is a representation of a signal modulated by a digital modulation scheme
such as quadrature amplitude modulation or phase-shift keying. It displays the signal as a two-
dimensional xy-plane scatter diagram in the complex plane at symbol sampling instants. In a manner
similar to that of a phasor diagram, the angle of a point, measured counterclockwise from the
horizontal axis, represents the phase shift of the carrier wave from a reference phase; the distance of
a point from the origin represents a measure of the amplitude or power of the signal.

In a digital modulation system, information is transmitted as a series of samples, each occupying a

uniform time slot. During each sample, the carrier wave has a constant amplitude and phase, which is
restricted to one of a finite number of values. So each sample encodes one of a finite number of
"symbols", which in turn represent one or more binary digits (bits) of information. Each symbol is
encoded as a different combination of amplitude and phase of the carrier, so each symbol is
represented by a point on the constellation diagram, called a constellation point. The constellation
diagram shows all the possible symbols that can be transmitted by the system as a collection of
points. In a frequency or phase modulated signal, the signal am The carrier representing each symbol
can be created by adding together different amounts of a cosine wave representing the "I" or in-
phase carrier, and a sine wave, shifted by 90° from the I carrier called the "Q" or quadrature carrier.
Thus each symbol can be represented by a complex number, and the constellation diagram can be
regarded as a complex plane, with the horizontal real axis representing the I component and the
vertical imaginary axis representing the Q component. A coherent detector is able to independently
demodulate these carriers. This principle of using two independently modulated carriers is the

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foundation of quadrature modulation. In pure phase modulation, the phase of the modulating symbol
is the phase of the carrier itself and this is the best representation of the modulated signal.

A 'signal space diagram' is an ideal constellation diagram showing the correct position of the point
representing each symbol. After passing through a communication channel, due to electronic noise or
distortion added to the signal, the amplitude and phase received by the demodulator may differ from
the correct value for the symbol. When plotted on a constellation diagram the point representing that
received sample will be offset from the correct position for that symbol. An electronic test instrument
called a vector signal analyzer can display the constellation diagram of a digital signal by sampling the
signal and plotting each received symbol as a point. The result is a 'ball' or 'cloud' of points
surrounding each symbol position. Measured constellation diagrams can be used to recognize the
type of interference and distortion in a signal.

The constellation as received, with noise added

The number of constellation points in a diagram gives the size of the "alphabet" of symbols that can
be transmitted by each sample, and so determines the number of bits transmitted per sample. It is
usually a power of 2. A diagram with four points, for example, represents a modulation scheme that
can separately encode all 4 combinations of two bits: 00, 01, 10, and 11, and so can transmit two bits
per sample. Thus in general a modulation with

After passing through the communication channel the signal is decoded by a demodulator. The
function of the demodulator is to classify each sample as a symbol. The set of sample values which
the demodulator classifies as a given symbol can be represented by a region in the plane drawn
around each constellation point. If noise causes the point representing a sample to stray into the
region representing another symbol, the demodulator will misidentify that sample as the other
symbol, resulting in a symbol error. Most demodulators choose, as its estimate of what was actually
transmitted, the constellation point which is closest (in a Euclidean distance sense) to that of the
received sample; this is called maximum likelihood detection. On the constellation diagram these
detection regions can be easily represented by dividing the plane by lines equidistant from each
adjacent pair of points.

One half the distance between each pair of neighboring points is the amplitude of additive noise or
distortion required to cause one of the points to be misidentified as the other, and thus cause a
symbol error. Therefore, the further the points are separated from one another, the greater the noise
immunity of the modulation. Practical modulation systems are designed to maximize the minimum
noise needed to cause a symbol error; on the constellation diagram this means that the distance
between each pair of adjacent points is equal.

The received signal quality can be analyzed by displaying the constellation diagram of the signal at the
receiver on a vector signal analyzer. Some types of distortion show up as characteristic patterns on
the diagram:

Gaussian noise causes the samples to land in a random ball about each constellation point
Non-coherent single frequency interference shows as samples making circles about each constellation
point
Phase noise shows as constellation points spreading into arcs centered on the origin
Amplifier compression causes the corner points to move towards the center
A constellation diagram visualises phenomena similar to those an eye pattern does for one-
dimensional signals. The eye pattern can be used to see timing jitter in one dimension of modulation.
plitude is constant, so the points lie on a circle around the origin.
Quadrature phase shift keying (QPSK) is a modulation technique that conveys two bits of data per
symbol by changing the phase of the carrier wave by 45 degrees. The data signal is represented by
four phases of the carrier wave, typically 0 degrees, 90 degrees, 180 degrees, and 270 degrees. A
binary 00 is represented by a phase shift of 0 degrees, a binary 01 is represented by a phase shift of 90
degrees, a binary 10 is represented by a phase shift of 180 degrees, and a binary 11 is represented by
a phase shift of 270 degrees.

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The constellation diagram for QPSK is a square with four points, one at each of the four quadrants.
The distance between the points is called the symbol spacing. The symbol spacing is determined by
the bit rate of the data signal.

The signal-to-noise ratio (SNR) of QPSK is determined by the symbol spacing and the noise power. The
higher the symbol spacing, the better the SNR.

QPSK is a more efficient modulation technique than BPSK because it can transmit two bits per symbol.
This means that it can achieve twice the data rate of BPSK for the same bandwidth.

QPSK is also more robust to noise than BPSK because the four points in the constellation diagram are
further apart, which makes it more difficult for noise to corrupt the signal.

Here are the principles of operation of QPSK:

The data signal is divided into two streams of bits.

Each stream of bits is modulated onto a separate carrier wave.
The two carrier waves are then combined to form the QPSK signal.
The QPSK signal is transmitted over the channel.
At the receiver, the QPSK signal is demodulated to recover the original data signal.
QPSK is a widely used modulation technique in wireless communication systems, such as Wi-Fi and 4G
LTE. It is also used in some wired communication systems, such as Ethernet.

Here are some of the advantages of QPSK:

More efficient than BPSK

More robust to noise than BPSK
Widely used in wireless communication systems
Here are some of the disadvantages of QPSK:

More complex to implement than BPSK

Requires more bandwidth than BPSK
Overall, QPSK is a good choice for applications where bandwidth is not a major constraint and where
noise immunity is important.
Differential phase shift keying (DPSK) is a modulation technique that conveys data by changing the
phase of the carrier wave relative to the previous signal element. No reference signal is considered
here. The signal phase follows the high or low state of the previous element. This DPSK technique
doesn't need a reference oscillator.

In DPSK, the data signal is represented by two phases of the carrier wave, typically 0 degrees and 180
degrees. A binary 0 is represented by a phase shift of 180 degrees relative to the previous signal
element, while a binary 1 is represented by no phase shift.

The constellation diagram for DPSK is a circle with two points, one at 0 degrees and one at 180
degrees. The distance between the two points is called the symbol spacing. The symbol spacing is
determined by the bit rate of the data signal.

The signal-to-noise ratio (SNR) of DPSK is determined by the symbol spacing and the noise power. The
higher the symbol spacing, the better the SNR.

DPSK is a more robust to noise than BPSK because the two points in the constellation diagram are
further apart, which makes it more difficult for noise to corrupt the signal.

In the presence of noise, the DPSK signal will be corrupted and the receiver will not be able to recover
the original data signal perfectly. However, the DPSK signal is more robust to noise than BPSK, so the
receiver will still be able to recover the original data signal with a lower bit error rate.

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The use of ideal filters in DPSK can help to improve the performance of the system in the presence of
noise. Ideal filters are filters that perfectly remove noise from the signal. However, ideal filters are not
practical to implement in real systems.

In practice, DPSK systems use non-ideal filters to remove noise from the signal. Non-ideal filters do
not perfectly remove noise from the signal, but they can still improve the performance of the system.

The choice of filter for a DPSK system depends on the specific application. For example, a system that
requires a low bit error rate may use a more complex filter than a system that does not require a low
bit error rate.

Here are some of the advantages of DPSK with noise and ideal filters:

More robust to noise than BPSK

Can be implemented with non-ideal filters
Widely used in wireless communication systems
Here are some of the disadvantages of DPSK with noise and ideal filters:

Requires more complex receiver than BPSK

Can be sensitive to timing errors
Overall, DPSK with noise and ideal filters is a good choice for applications where robustness to noise is
important.
Differential phase shift keying (DPSK) is a modulation technique that conveys data by changing the
phase of the carrier wave relative to the previous signal element. No reference signal is considered
here. The signal phase follows the high or low state of the previous element. This DPSK technique
doesn't need a reference oscillator.

In DPSK, the data signal is represented by two phases of the carrier wave, typically 0 degrees and 180
degrees. A binary 0 is represented by a phase shift of 180 degrees relative to the previous signal
element, while a binary 1 is represented by no phase shift.
A Gaussian minimum-shift keying (GMSK) modulator is a type of digital modulation technique that
uses a Gaussian filter to shape the signal before it is modulated onto a carrier wave. The I-Q
modulator block diagram is a simplified representation of the modulator, which consists of the
following blocks:

Data input: This block is where the digital data signal is input to the modulator.
Bit synchronizer: This block ensures that the data signal is synchronized to the carrier frequency.
Gaussian filter: This block shapes the data signal to have a Gaussian distribution.
I-Q modulator: This block combines the in-phase (I) and quadrature-phase (Q) components of the
signal to form the modulated signal.
Carrier oscillator: This block generates the carrier wave.
The functions of each block are as follows:

Data input: This block is where the digital data signal is input to the modulator. The data signal is
typically a binary signal, but it can also be an analog signal that has been digitized.
Bit synchronizer: This block ensures that the data signal is synchronized to the carrier frequency. This
is important because the Gaussian filter will only work properly if the data signal is synchronized to
the carrier frequency.
Gaussian filter: This block shapes the data signal to have a Gaussian distribution. This is done to
reduce the sidebands of the signal, which improves the spectral efficiency of the modulation.
I-Q modulator: This block combines the in-phase (I) and quadrature-phase (Q) components of the
signal to form the modulated signal. The I component is the signal that is aligned with the carrier
wave, while the Q component is the signal that is perpendicular to the carrier wave.
Carrier oscillator: This block generates the carrier wave. The carrier wave is a sinusoidal signal that has
the same frequency as the data signal.
The GMSK modulator is a complex circuit, but it is a very efficient modulation technique that is used in
many wireless communication systems.

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Here are some additional details about each block:

Data input: The data input can be a binary signal or an analog signal that has been digitized. The data
signal is typically represented by a sequence of bits, where each bit can be either a 0 or a 1.
Bit synchronizer: The bit synchronizer ensures that the data signal is synchronized to the carrier
frequency. This is done by using a phase-locked loop (PLL) circuit. The PLL circuit compares the phase
of the data signal to the phase of the carrier wave and generates an error signal. This error signal is
then used to adjust the frequency of the data signal so that it becomes synchronized to the carrier
frequency.
Gaussian filter: The Gaussian filter shapes the data signal to have a Gaussian distribution. This is done
by passing the data signal through a filter that has a Gaussian impulse response. The Gaussian filter
reduces the sidebands of the signal, which improves the spectral efficiency of the modulation.
I-Q modulator: The I-Q modulator combines the in-phase (I) and quadrature-phase (Q) components of
the signal to form the modulated signal. The I component is the signal that is aligned with the carrier
wave, while the Q component is the signal that is perpendicular to the carrier wave. The I-Q
modulator does this by multiplying the I and Q components of the signal by two sinusoids that are 90
degrees out of phase with each other.
Carrier oscillator: The carrier oscillator generates the carrier wave. The carrier wave is a sinusoidal
signal that has the same frequency as the data signal. The carrier oscillator is typically a crystal
oscillator, which is a very stable oscillator that can maintain its frequency over a wide range of
temperatures.
In digital communication systems, an error occurs when the transmitted signal is not received
correctly at the receiver. There are many different types of errors that can occur in digital
communication systems, but some of the most common ones include:

Bit errors: A bit error is when a single bit is changed from a 0 to a 1 or vice versa.
Block errors: A block error is when a group of bits is changed.
Timing errors: A timing error is when the received signal is not synchronized with the transmitted
signal.
Noise: Noise is any unwanted signal that can corrupt the transmitted signal.
Interference: Interference is a signal that is emitted from another source and that can corrupt the
transmitted signal.
The sources of errors in digital communication systems can be divided into two categories: internal
and external.

Internal errors: Internal errors are caused by the components of the communication system itself. For
example, a bit error can be caused by a faulty transistor in the transmitter or receiver.
External errors: External errors are caused by factors outside of the communication system. For
example, noise can be caused by lightning or other electromagnetic interference.
The measurements of errors in digital communication systems are used to quantify the performance
of the system. Some of the most common measurements include:

Bit error rate (BER): The BER is the number of bit errors per unit time.
Block error rate (BLER): The BLER is the number of block errors per unit time.
Symbol error rate (SER): The SER is the number of symbol errors per unit time.
Carrier-to-noise ratio (CNR): The CNR is the ratio of the power of the carrier signal to the power of the
noise.
Eb/No: The Eb/No is the ratio of the energy per bit to the noise power spectral density.
The BER, BLER, SER, and CNR are all important measurements of the performance of digital
communication systems. The BER is the most basic measurement, but it can be misleading if the
errors are not evenly distributed. The BLER is a more robust measurement, but it is also more difficult
to measure. The SER is a measure of the errors in the symbols, which can be useful for systems that
use modulation techniques that transmit multiple bits per symbol. The CNR and Eb/No are measures
of the signal-to-noise ratio, which is a fundamental parameter that affects the performance of all
digital communication systems.

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Direct-sequence spread spectrum (DSSS) and frequency hopping spread spectrum (FHSS) are two
types of spread spectrum modulation techniques used in digital signal transmission over airwaves.

DSSS works by multiplying the data signal with a pseudorandom binary sequence (PRBS), also known
as a chip code. This spreads the data signal over a wider bandwidth, making it more difficult to
intercept or jam. DSSS is the most common type of spread spectrum modulation and is used in many
wireless applications, including Wi-Fi, Bluetooth, and cellular networks.

FHSS works by rapidly hopping the carrier frequency of the data signal between a number of
predefined frequencies. This makes it difficult for an adversary to intercept the signal because they
would need to know the hopping pattern in order to decode it. FHSS is less common than DSSS but is
used in some applications, such as military communications.

Here is a table comparing the two techniques:

Feature DSSS FHSS

Spreading method Multiplies the data signal with a PRBS Hops the carrier frequency
between a number of predefined frequencies
Bandwidth Wider Narrower
Security More secure Less secure
Jamming resistance More resistant Less resistant
Multipath fading resistance Less resistant More resistant
Complexity Less complex More complex
Cost Less expensive More expensive
The best spread spectrum technique to use depends on the specific application. DSSS is generally the
better choice for applications that require high security and jamming resistance, such as military
communications. FHSS is a better choice for applications that are more susceptible to multipath
fading, such as wireless networks in indoor environments.

Here are some of the advantages of using spread spectrum modulation techniques:

Security: Spread spectrum signals are more difficult to intercept or jam than narrowband signals. This
is because the spread spectrum signal occupies a wider bandwidth, making it more difficult to identify
and filter out.
Robustness: Spread spectrum signals are more robust to noise and interference than narrowband
signals. This is because the spread spectrum signal is spread out over a wider frequency range, so the
noise and interference are spread out as well.
Multiple access: Spread spectrum signals can be used to support multiple users on the same
frequency band. This is because each user's signal is spread out over a different frequency range.

OFDM (Orthogonal Frequency-Division Multiplexing) and UWB (Ultra-Wideband) are two advanced
modulation techniques that are used in wireless communication systems.
OFDM
OFDM is a technique that divides the available bandwidth into a number of smaller, overlapping
frequency bands called subcarriers. Each subcarrier is modulated using a different modulation
scheme, such as QAM (Quadrature Amplitude Modulation) or PSK (Phase-Shift Keying). The
modulated subcarriers are then summed together to form the transmitted signal.
OFDM has several advantages over other modulation techniques. It is very efficient in terms of
spectral usage, meaning that it can transmit a lot of data in a small amount of bandwidth. It is also
very robust to multipath fading, which is a common problem in wireless channels.
UWB
UWB is a technique that uses a very wide frequency band for transmission. This allows UWB systems
to achieve very high data rates, even in the presence of interference from other wireless devices.
However, UWB systems also have a shorter range than OFDM systems.
UWB is often used for applications that require high data rates over short distances, such as wireless
sensor networks and medical imaging.
Here is a table comparing the two modulation techniques:

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Feature OFDM UWB

Spectral efficiency High Low

Robustness to multipath fading High Low

Data rate High Very high

Range Long Short

Applications Wi-Fi, LTE, 5G, digital TV Wireless sensor networks, medical imaging, radar

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In general, OFDM is a better choice for applications that require high data rates and good range, such
as wireless networks. UWB is a better choice for applications that require very high data rates over
short distances, such as wireless sensor networks and medical imaging.
Here are some additional details about each modulation technique:
 OFDM is used in a wide variety of wireless communication systems, including Wi-Fi, LTE, and
5G. It is also used in digital television broadcasting.
 UWB is a relatively new modulation technique that is still under development. However, it
has the potential to revolutionize wireless communication by enabling high data rates over
short distances.

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