This paper proposes a new method for designing a two-channel quadrature mirror filter bank using polyphase components of the low-pass prototype filter. The method formulates an objective function that is minimized using a power optimization approach. The objective function is a linear combination of passband error and stopband residual energy. Simulation results show the proposed method requires less computational time and fewer iterations compared to existing methods. Design examples demonstrate the method produces improved stopband attenuation and reconstruction error performance.
This document discusses filter banks, which are arrays of bandpass filters that separate an input signal into multiple sub-band components. It covers types of filter banks like analysis and synthesis banks, as well as uniform and non-uniform filter banks. Two-channel and polyphase two-channel filter banks are explained in more detail. Applications like signal compression and graphic equalizers are also mentioned. Lifting approaches for implementing filters efficiently are briefly outlined.
we demonstrate that it can be
advantageous from a computational point of view
to use a two-stage realization instead of a single stage
realization for sample rate conversions with
prime numbers. One of the stages performs a
conversion by a factor of two using linear-phase, or
approximately linear-phase, half-band filters. The
other stage changes the sample rate by the
rational factor N/2 using a linear-phase FIR filter.
The actual filtering can be performed at the lowest
of the two sample frequencies involved. It is also
possible to exploit the coefficient symmetry of the
linear-phase FIR filter in the stage that changes
the rate by a rational factor. The overall workload
of the two-stage realization can therefore be
reduced compared with the corresponding single stage
realization.
An efficient multi resolution filter bank based on da based multiplicationVLSICS Design
Multi-resolution filter bank (MRFB)-based on the fast filter bank design can be used for multiple resolution
spectrum sensing. MRFB overcomes the constraint of fixed sensing resolution in spectrum sensors based
on conventional discrete Fourier transform filter banks (DFTFB) without hardware re-implementation.
Multipliers have a greater impact on complexity and performance of the design because a large number of
constant multiplications are required in the multiplication of filter coefficients with the filter input.
Modified multiplier architecture Distributed Arithmetic(DA) is used to improve the efficiency
This document provides an overview of multirate digital signal processing using filter banks and subband processing. It describes:
1) How signals are split into multiple frequency bands or subbands using an analysis filter bank and then downsampled.
2) Individual subband processing such as coding/compression can then be applied.
3) The subbands are then reconstructed using upsampling and a synthesis filter bank to combine the signals back into the original sampling rate.
4) Perfect reconstruction filter banks can be designed to remove aliasing introduced during downsampling to accurately reconstruct the original signal up to a delay.
This document discusses decimation and interpolation using polyphase filters. It begins by defining decimation and interpolation and describing how decimators use anti-aliasing filters followed by downsamplers, while interpolators use upsamplers followed by anti-imaging filters. It then presents the principles of polyphase decimation and interpolation, showing how they can be represented in both the time and z-transform domains. This allows implementing decimators and interpolators more efficiently using fewer filter operations and less memory than traditional implementations.
The document provides an overview of adaptive filters. It discusses that adaptive filters are digital filters that have self-adjusting characteristics to changes in input signals. They have two main components: a digital filter with adjustable coefficients and an adaptive algorithm. Common adaptive algorithms are LMS and RLS. Adaptive filters are used for applications like noise cancellation, system identification, channel equalization, and signal prediction. The key aspects of adaptive filter theory and algorithms like LMS, RLS, Wiener filters are also covered.
This document discusses baseband shaping for data transmission. It introduces several digital modulation formats used for discrete PAM signals like unipolar, polar, bipolar and Manchester encoding. It then discusses factors that affect transmission like DC component, bandwidth, bit synchronization and error detection. It describes intersymbol interference and its causes like multipath propagation and bandlimited channels. It presents the baseband binary data transmission system model. Nyquist's criterion for zero ISI is defined. Practical solutions like raised cosine filters are discussed. The transmission bandwidth requirement is derived based on the rolloff factor. Finally, it describes what an eye diagram reveals about the system performance.
This document summarizes a presentation on multirate digital signal processing. Multirate systems involve processing signals at different sampling rates, using operations like decimation to lower the sampling rate and interpolation to increase it. Decimation involves downsampling by discarding samples, while interpolation involves upsampling by inserting zeros. These operations are used for applications like sampling rate conversion, audio/video encoding, and communications systems. Key aspects of multirate signal processing discussed include anti-alias filtering, sampling rate conversion using cascaded decimation and interpolation, and choosing optimal filter designs.
This document discusses filter banks, which are arrays of bandpass filters that separate an input signal into multiple sub-band components. It covers types of filter banks like analysis and synthesis banks, as well as uniform and non-uniform filter banks. Two-channel and polyphase two-channel filter banks are explained in more detail. Applications like signal compression and graphic equalizers are also mentioned. Lifting approaches for implementing filters efficiently are briefly outlined.
we demonstrate that it can be
advantageous from a computational point of view
to use a two-stage realization instead of a single stage
realization for sample rate conversions with
prime numbers. One of the stages performs a
conversion by a factor of two using linear-phase, or
approximately linear-phase, half-band filters. The
other stage changes the sample rate by the
rational factor N/2 using a linear-phase FIR filter.
The actual filtering can be performed at the lowest
of the two sample frequencies involved. It is also
possible to exploit the coefficient symmetry of the
linear-phase FIR filter in the stage that changes
the rate by a rational factor. The overall workload
of the two-stage realization can therefore be
reduced compared with the corresponding single stage
realization.
An efficient multi resolution filter bank based on da based multiplicationVLSICS Design
Multi-resolution filter bank (MRFB)-based on the fast filter bank design can be used for multiple resolution
spectrum sensing. MRFB overcomes the constraint of fixed sensing resolution in spectrum sensors based
on conventional discrete Fourier transform filter banks (DFTFB) without hardware re-implementation.
Multipliers have a greater impact on complexity and performance of the design because a large number of
constant multiplications are required in the multiplication of filter coefficients with the filter input.
Modified multiplier architecture Distributed Arithmetic(DA) is used to improve the efficiency
This document provides an overview of multirate digital signal processing using filter banks and subband processing. It describes:
1) How signals are split into multiple frequency bands or subbands using an analysis filter bank and then downsampled.
2) Individual subband processing such as coding/compression can then be applied.
3) The subbands are then reconstructed using upsampling and a synthesis filter bank to combine the signals back into the original sampling rate.
4) Perfect reconstruction filter banks can be designed to remove aliasing introduced during downsampling to accurately reconstruct the original signal up to a delay.
This document discusses decimation and interpolation using polyphase filters. It begins by defining decimation and interpolation and describing how decimators use anti-aliasing filters followed by downsamplers, while interpolators use upsamplers followed by anti-imaging filters. It then presents the principles of polyphase decimation and interpolation, showing how they can be represented in both the time and z-transform domains. This allows implementing decimators and interpolators more efficiently using fewer filter operations and less memory than traditional implementations.
The document provides an overview of adaptive filters. It discusses that adaptive filters are digital filters that have self-adjusting characteristics to changes in input signals. They have two main components: a digital filter with adjustable coefficients and an adaptive algorithm. Common adaptive algorithms are LMS and RLS. Adaptive filters are used for applications like noise cancellation, system identification, channel equalization, and signal prediction. The key aspects of adaptive filter theory and algorithms like LMS, RLS, Wiener filters are also covered.
This document discusses baseband shaping for data transmission. It introduces several digital modulation formats used for discrete PAM signals like unipolar, polar, bipolar and Manchester encoding. It then discusses factors that affect transmission like DC component, bandwidth, bit synchronization and error detection. It describes intersymbol interference and its causes like multipath propagation and bandlimited channels. It presents the baseband binary data transmission system model. Nyquist's criterion for zero ISI is defined. Practical solutions like raised cosine filters are discussed. The transmission bandwidth requirement is derived based on the rolloff factor. Finally, it describes what an eye diagram reveals about the system performance.
This document summarizes a presentation on multirate digital signal processing. Multirate systems involve processing signals at different sampling rates, using operations like decimation to lower the sampling rate and interpolation to increase it. Decimation involves downsampling by discarding samples, while interpolation involves upsampling by inserting zeros. These operations are used for applications like sampling rate conversion, audio/video encoding, and communications systems. Key aspects of multirate signal processing discussed include anti-alias filtering, sampling rate conversion using cascaded decimation and interpolation, and choosing optimal filter designs.
The document discusses digital filters and their design process. It explains that the design process involves four main steps: approximation, realization, studying arithmetic errors, and implementation.
For approximation, direct and indirect methods are used to generate a transfer function that satisfies the filter specifications. Realization generates a filter network from the transfer function. Studying arithmetic errors examines how quantization affects filter performance. Implementation realizes the filter in either software or hardware.
The document also outlines the basic building blocks of digital filters, including adders, multipliers, and delay elements. It introduces linear time-invariant digital filters and explains their input-output relationship using difference equations and the z-transform.
This document discusses channel equalization techniques for digital communication systems. It describes four main threats in digital communication channels: inter-symbol interference, multipath propagation, co-channel interference, and noise. It then explains various linear equalization techniques like LMS and NLMS adaptive filters that can be used to mitigate inter-symbol interference. Finally, it discusses the need for non-linear equalizers and how multilayer perceptron neural networks can be used for non-linear channel equalization.
It is sometimes desirable to have circuits capable of selectively filtering one frequency or range of frequencies out of a mix of different frequencies in a circuit. A circuit designed to perform this frequency selection is called a filter circuit, or simply a filter. A common need for filter circuits is in high-performance stereo systems, where certain ranges of audio frequencies need to be amplified or suppressed for best sound quality and power efficiency. You may be familiar with equalizers, which allow the amplitudes of several frequency ranges to be adjusted to suit the listener's taste and acoustic properties of the listening area. You may also be familiar with crossover networks, which block certain ranges of frequencies from reaching speakers. A tweeter (high-frequency speaker) is inefficient at reproducing low-frequency signals such as drum beats, so a crossover circuit is connected between the tweeter and the stereo's output terminals to block low-frequency signals, only passing high-frequency signals to the speaker's connection terminals. This gives better audio system efficiency and thus better performance. Both equalizers and crossover networks are examples of filters, designed to accomplish filtering of certain frequencies.
This document contains 5 problems related to digital signal processing. Problem 1 involves designing a 25-tap finite impulse response (FIR) filter to approximate an ideal bandpass frequency response and plotting the filter's magnitude and phase response. Problem 2 repeats this for a bandstop filter. Problem 3 converts an analog filter to a digital infinite impulse response (IIR) filter using impulse invariance. Problem 4 does the same conversion using bilinear transformation. Problem 5 compares the two conversion methods and discusses their restrictions.
The document reports on a subjective comparison of 13 speech enhancement algorithms conducted by Dynastat, Inc. using the ITU-T P.835 methodology. The algorithms encompassed four classes: spectral subtractive, subspace, statistical-model based, and Wiener. A noisy speech corpus called NOIZEUS was developed with IEEE sentences corrupted by 8 noises at varying SNRs. For the subjective tests, enhanced speech from 20 sentences in 4 noises at 2 SNRs was evaluated based on signal distortion, noise distortion, and overall quality. The statistical-model based methods performed best overall, followed by a multi-band spectral subtraction method. Incorporating noise estimation did not significantly improve performance for some algorithms.
This document provides an overview of digital filters and focuses on finite impulse response (FIR) filters. It defines digital filtering and compares it to analog filtering. It describes different types of digital filters including FIR filters and explains how to design, implement and characterize FIR filters. Key aspects of FIR filters are that they have a finite impulse response, linear phase, and are always stable. Design techniques like windowing methods and Parks-McClellan optimization are covered.
The document discusses the discrete cosine transform (DCT) and compares it to the discrete Fourier transform (DFT). Some key points:
- DCT provides better energy compaction than DFT, making it useful for image/signal compression applications. It transforms a signal from the spatial domain to the frequency domain.
- DCT is a real-valued transform derived from DFT, but is more computationally efficient as it avoids redundant complex calculations for real input data.
- The one-dimensional and two-dimensional DCT transforms are defined. DCT has properties like separability and invertibility like DFT.
- DCT coefficients represent the signal's frequency content, with low frequencies concentrated in
Design And Performance of Finite impulse Response Filter Using Hyperbolic Cos...IDES Editor
In this paper a proposed of design and analysis of
Finite impulse response filter using Hyperbolic Cosine
window (Cosh window for short). This window is very useful
for some applications such as beam forming, filter design,
and speech processing. Digital FIR filter designed by Kaiser
window has a better far-end stop-band attenuation than filter
designed by the other previously well known adjustable
windows such as Dolph-Chebyshev and Saramaki, which are
special cases of Ultraspherical windows, but obtaining a digital
filter which performs higher far-end stop band attenuation
than Kaiser window will be useful. In this paper, the design of
nonrecursive digital FIR filter has been proposed by using
Cosh window. It provides better side lobe roll-off ratio & farend
stop band attenuation than filter designed by well known
Kaiser window, which is the advantage of filter designed by
Cosh window over filter designed by Kaiser window. An
expression for the side lobe & far field level has been developed.
Simulation & experimental results showing a good agreement
with theory has been provided
This document provides an overview of adaptive filters, including:
- Adaptive filters have coefficients that are adjusted based on input data to optimize performance, unlike fixed filters.
- The LMS algorithm is commonly used to adjust coefficients to minimize the mean square error between the filter output and a desired signal.
- Key applications of adaptive filters include noise cancellation, system identification, channel equalization, and echo cancellation.
Adaptive filters are able to operate in unknown environments and track variations in input signals. They can be used for system identification, prediction, noise cancellation, and channel equalization. The MATLAB Filter Design Toolbox provides an adapfilt function for implementing adaptive filters using algorithms like LMS. The adaptive filter output will approximate the physical system response after convergence by minimizing the difference between the outputs.
This document discusses adaptive equalization techniques used in wireless communications. It begins by describing different types of interference such as co-channel, adjacent channel, and inter-symbol interference that affect wireless transmissions. Equalization is introduced as a technique to counter inter-symbol interference by concentrating dispersed symbol energy back into its time interval. Adaptive equalization is specifically discussed as it can track time-varying mobile channel characteristics using algorithms like zero forcing, least mean squares, and recursive least squares. The key components of an adaptive equalizer including its operating modes in training and tracking are also outlined.
The document discusses adaptive filters, which can automatically adjust their parameters to filter signals whose exact frequency response is unknown. It defines adaptive filters as having an input signal, filter structure, adjustable parameters, and adaptive algorithm. The goal of adaptive filtering is to minimize the error between the filter's output and a desired response. It describes common adaptive filtering problems and solutions like using gradient descent algorithms and the mean squared error cost function to adjust the filter parameters over time and minimize error.
Equalization is a process used to mitigate interference in signals transmitted through dispersive channels. It works by adjusting the balance between frequency components. Common types of interference it addresses are inter-symbol interference, co-channel interference, and adjacent channel interference. Adaptive equalizers can update their parameters periodically during data transmission, while non-adaptive equalizers use fixed parameters after an initial adjustment. Equalization is implemented using equalizer objects in MATLAB that describe the equalizer class and algorithm, which are then applied to the received signal.
Fir filter design (windowing technique)Bin Biny Bino
The window design technique for FIR filters involves choosing an ideal frequency-selective filter with the desired passband and stopband characteristics, and then multiplying or "windowing" its infinite impulse response with an appropriate window function to make it causal and finite. This windowing in the time domain corresponds to convolution in the frequency domain. Common window functions are used to truncate the ideal filter response while maintaining desirable filtering properties. MATLAB code can be used to implement windowed FIR filters.
This document outlines the course details for a digital signal processing course. The main goal of the course is to design digital linear time-invariant filters that are widely used in applications such as audio, communications, radar, and biomedical engineering. Topics that will be covered include sampling of continuous-time signals, discrete-time signals and systems, the z-transform, filter design techniques, discrete Fourier transforms, and applications of digital signal processing. Students will be evaluated based on midterm and final exams, quizzes, assignments, and a project.
This document provides an overview of adaptive filtering techniques. It discusses digital filters and classifications such as linear/nonlinear and finite impulse response (FIR)/infinite impulse response (IIR). It then covers Wiener filters, including how they minimize mean square error. The method of steepest descent is presented as an approach to solve the Wiener-Hopf equations to find optimal filter weights. Finally, it discusses how the least mean squares (LMS) algorithm can be used for adaptive filtering by updating filter weights recursively in the direction that reduces mean square error.
The document discusses digital filter design. It begins by defining digital filters and their purposes, which include signal separation and distortion removal. It then covers the main types of digital filters - finite impulse response (FIR) and infinite impulse response (IIR) filters. FIR filters are implemented non-recursively without feedback, while IIR filters use recursion and feedback. The document outlines FIR filter design methods like windowing and discusses applications of digital filters such as noise suppression, frequency enhancement, and interference removal. In conclusion, digital filters can have linear phase response and are not affected by environmental factors like heat.
Unconstrained Optimization Method to Design Two Channel Quadrature Mirror Fil...CSCJournals
This paper proposes an efficient method for the design of two-channel, quadrature mirror filter (QMF) bank for subband image coding. The choice of filter bank is important as it affects image quality as well as system design complexity. The design problem is formulated as weighted sum of reconstruction error in time domain and passband and stop-band energy of the low-pass analysis filter of the filter bank .The objective function is minimized directly, using nonlinear unconstrained method. Experimental results of the method on images show that the performance of the proposed method is better than that of the already existing methods. The impact of some filter characteristics, such as stopband attenuation, stopband edge, and filter length on the performance of the reconstructed images is also investigated.
Adaptive Trilateral Filter for In-Loop Filteringcsandit
High Efficiency Video Coding (HEVC) has achieved si
gnificant coding efficiency improvement
beyond existing video coding standard by employing
several new coding tools. Deblocking
Filter, Sample Adaptive Offset (SAO) and Adaptive L
oop Filter (ALF) for in-loop filtering are
currently introduced for the HEVC standard. However
, these filters are implemented in spatial
domain despite the fact of temporal correlation wit
hin video sequences. To reduce the artifacts
and better align object boundaries in video, a prop
osed algorithm in in-loop filtering is
proposed. The proposed algorithm is implemented in
HM-11.0 software. This proposed
algorithm allows an average bitrate reduction of ab
out 0.7% and improves the PSNR of the
decoded frame by 0.05%, 0.30% and 0.35% in luminanc
e and chroma.
The document discusses digital filters and their design process. It explains that the design process involves four main steps: approximation, realization, studying arithmetic errors, and implementation.
For approximation, direct and indirect methods are used to generate a transfer function that satisfies the filter specifications. Realization generates a filter network from the transfer function. Studying arithmetic errors examines how quantization affects filter performance. Implementation realizes the filter in either software or hardware.
The document also outlines the basic building blocks of digital filters, including adders, multipliers, and delay elements. It introduces linear time-invariant digital filters and explains their input-output relationship using difference equations and the z-transform.
This document discusses channel equalization techniques for digital communication systems. It describes four main threats in digital communication channels: inter-symbol interference, multipath propagation, co-channel interference, and noise. It then explains various linear equalization techniques like LMS and NLMS adaptive filters that can be used to mitigate inter-symbol interference. Finally, it discusses the need for non-linear equalizers and how multilayer perceptron neural networks can be used for non-linear channel equalization.
It is sometimes desirable to have circuits capable of selectively filtering one frequency or range of frequencies out of a mix of different frequencies in a circuit. A circuit designed to perform this frequency selection is called a filter circuit, or simply a filter. A common need for filter circuits is in high-performance stereo systems, where certain ranges of audio frequencies need to be amplified or suppressed for best sound quality and power efficiency. You may be familiar with equalizers, which allow the amplitudes of several frequency ranges to be adjusted to suit the listener's taste and acoustic properties of the listening area. You may also be familiar with crossover networks, which block certain ranges of frequencies from reaching speakers. A tweeter (high-frequency speaker) is inefficient at reproducing low-frequency signals such as drum beats, so a crossover circuit is connected between the tweeter and the stereo's output terminals to block low-frequency signals, only passing high-frequency signals to the speaker's connection terminals. This gives better audio system efficiency and thus better performance. Both equalizers and crossover networks are examples of filters, designed to accomplish filtering of certain frequencies.
This document contains 5 problems related to digital signal processing. Problem 1 involves designing a 25-tap finite impulse response (FIR) filter to approximate an ideal bandpass frequency response and plotting the filter's magnitude and phase response. Problem 2 repeats this for a bandstop filter. Problem 3 converts an analog filter to a digital infinite impulse response (IIR) filter using impulse invariance. Problem 4 does the same conversion using bilinear transformation. Problem 5 compares the two conversion methods and discusses their restrictions.
The document reports on a subjective comparison of 13 speech enhancement algorithms conducted by Dynastat, Inc. using the ITU-T P.835 methodology. The algorithms encompassed four classes: spectral subtractive, subspace, statistical-model based, and Wiener. A noisy speech corpus called NOIZEUS was developed with IEEE sentences corrupted by 8 noises at varying SNRs. For the subjective tests, enhanced speech from 20 sentences in 4 noises at 2 SNRs was evaluated based on signal distortion, noise distortion, and overall quality. The statistical-model based methods performed best overall, followed by a multi-band spectral subtraction method. Incorporating noise estimation did not significantly improve performance for some algorithms.
This document provides an overview of digital filters and focuses on finite impulse response (FIR) filters. It defines digital filtering and compares it to analog filtering. It describes different types of digital filters including FIR filters and explains how to design, implement and characterize FIR filters. Key aspects of FIR filters are that they have a finite impulse response, linear phase, and are always stable. Design techniques like windowing methods and Parks-McClellan optimization are covered.
The document discusses the discrete cosine transform (DCT) and compares it to the discrete Fourier transform (DFT). Some key points:
- DCT provides better energy compaction than DFT, making it useful for image/signal compression applications. It transforms a signal from the spatial domain to the frequency domain.
- DCT is a real-valued transform derived from DFT, but is more computationally efficient as it avoids redundant complex calculations for real input data.
- The one-dimensional and two-dimensional DCT transforms are defined. DCT has properties like separability and invertibility like DFT.
- DCT coefficients represent the signal's frequency content, with low frequencies concentrated in
Design And Performance of Finite impulse Response Filter Using Hyperbolic Cos...IDES Editor
In this paper a proposed of design and analysis of
Finite impulse response filter using Hyperbolic Cosine
window (Cosh window for short). This window is very useful
for some applications such as beam forming, filter design,
and speech processing. Digital FIR filter designed by Kaiser
window has a better far-end stop-band attenuation than filter
designed by the other previously well known adjustable
windows such as Dolph-Chebyshev and Saramaki, which are
special cases of Ultraspherical windows, but obtaining a digital
filter which performs higher far-end stop band attenuation
than Kaiser window will be useful. In this paper, the design of
nonrecursive digital FIR filter has been proposed by using
Cosh window. It provides better side lobe roll-off ratio & farend
stop band attenuation than filter designed by well known
Kaiser window, which is the advantage of filter designed by
Cosh window over filter designed by Kaiser window. An
expression for the side lobe & far field level has been developed.
Simulation & experimental results showing a good agreement
with theory has been provided
This document provides an overview of adaptive filters, including:
- Adaptive filters have coefficients that are adjusted based on input data to optimize performance, unlike fixed filters.
- The LMS algorithm is commonly used to adjust coefficients to minimize the mean square error between the filter output and a desired signal.
- Key applications of adaptive filters include noise cancellation, system identification, channel equalization, and echo cancellation.
Adaptive filters are able to operate in unknown environments and track variations in input signals. They can be used for system identification, prediction, noise cancellation, and channel equalization. The MATLAB Filter Design Toolbox provides an adapfilt function for implementing adaptive filters using algorithms like LMS. The adaptive filter output will approximate the physical system response after convergence by minimizing the difference between the outputs.
This document discusses adaptive equalization techniques used in wireless communications. It begins by describing different types of interference such as co-channel, adjacent channel, and inter-symbol interference that affect wireless transmissions. Equalization is introduced as a technique to counter inter-symbol interference by concentrating dispersed symbol energy back into its time interval. Adaptive equalization is specifically discussed as it can track time-varying mobile channel characteristics using algorithms like zero forcing, least mean squares, and recursive least squares. The key components of an adaptive equalizer including its operating modes in training and tracking are also outlined.
The document discusses adaptive filters, which can automatically adjust their parameters to filter signals whose exact frequency response is unknown. It defines adaptive filters as having an input signal, filter structure, adjustable parameters, and adaptive algorithm. The goal of adaptive filtering is to minimize the error between the filter's output and a desired response. It describes common adaptive filtering problems and solutions like using gradient descent algorithms and the mean squared error cost function to adjust the filter parameters over time and minimize error.
Equalization is a process used to mitigate interference in signals transmitted through dispersive channels. It works by adjusting the balance between frequency components. Common types of interference it addresses are inter-symbol interference, co-channel interference, and adjacent channel interference. Adaptive equalizers can update their parameters periodically during data transmission, while non-adaptive equalizers use fixed parameters after an initial adjustment. Equalization is implemented using equalizer objects in MATLAB that describe the equalizer class and algorithm, which are then applied to the received signal.
Fir filter design (windowing technique)Bin Biny Bino
The window design technique for FIR filters involves choosing an ideal frequency-selective filter with the desired passband and stopband characteristics, and then multiplying or "windowing" its infinite impulse response with an appropriate window function to make it causal and finite. This windowing in the time domain corresponds to convolution in the frequency domain. Common window functions are used to truncate the ideal filter response while maintaining desirable filtering properties. MATLAB code can be used to implement windowed FIR filters.
This document outlines the course details for a digital signal processing course. The main goal of the course is to design digital linear time-invariant filters that are widely used in applications such as audio, communications, radar, and biomedical engineering. Topics that will be covered include sampling of continuous-time signals, discrete-time signals and systems, the z-transform, filter design techniques, discrete Fourier transforms, and applications of digital signal processing. Students will be evaluated based on midterm and final exams, quizzes, assignments, and a project.
This document provides an overview of adaptive filtering techniques. It discusses digital filters and classifications such as linear/nonlinear and finite impulse response (FIR)/infinite impulse response (IIR). It then covers Wiener filters, including how they minimize mean square error. The method of steepest descent is presented as an approach to solve the Wiener-Hopf equations to find optimal filter weights. Finally, it discusses how the least mean squares (LMS) algorithm can be used for adaptive filtering by updating filter weights recursively in the direction that reduces mean square error.
The document discusses digital filter design. It begins by defining digital filters and their purposes, which include signal separation and distortion removal. It then covers the main types of digital filters - finite impulse response (FIR) and infinite impulse response (IIR) filters. FIR filters are implemented non-recursively without feedback, while IIR filters use recursion and feedback. The document outlines FIR filter design methods like windowing and discusses applications of digital filters such as noise suppression, frequency enhancement, and interference removal. In conclusion, digital filters can have linear phase response and are not affected by environmental factors like heat.
Unconstrained Optimization Method to Design Two Channel Quadrature Mirror Fil...CSCJournals
This paper proposes an efficient method for the design of two-channel, quadrature mirror filter (QMF) bank for subband image coding. The choice of filter bank is important as it affects image quality as well as system design complexity. The design problem is formulated as weighted sum of reconstruction error in time domain and passband and stop-band energy of the low-pass analysis filter of the filter bank .The objective function is minimized directly, using nonlinear unconstrained method. Experimental results of the method on images show that the performance of the proposed method is better than that of the already existing methods. The impact of some filter characteristics, such as stopband attenuation, stopband edge, and filter length on the performance of the reconstructed images is also investigated.
Adaptive Trilateral Filter for In-Loop Filteringcsandit
High Efficiency Video Coding (HEVC) has achieved si
gnificant coding efficiency improvement
beyond existing video coding standard by employing
several new coding tools. Deblocking
Filter, Sample Adaptive Offset (SAO) and Adaptive L
oop Filter (ALF) for in-loop filtering are
currently introduced for the HEVC standard. However
, these filters are implemented in spatial
domain despite the fact of temporal correlation wit
hin video sequences. To reduce the artifacts
and better align object boundaries in video, a prop
osed algorithm in in-loop filtering is
proposed. The proposed algorithm is implemented in
HM-11.0 software. This proposed
algorithm allows an average bitrate reduction of ab
out 0.7% and improves the PSNR of the
decoded frame by 0.05%, 0.30% and 0.35% in luminanc
e and chroma.
High Efficiency Video Coding (HEVC) has achieved significant coding efficiency improvement beyond
existing video coding standard by employing several new coding tools. Deblocking Filter, Sample Adaptive
Offset (SAO) and Adaptive Loop Filter (ALF) for in-loop filtering are currently introduced for the HEVC
standard. However, these filters are implemented in spatial domain despite the fact of temporal correlation
within video sequences. To reduce the artifacts and better align object boundaries in video, a proposed
algorithm in in-loop filtering is proposed. The proposed algorithm is implemented in HM-11.0 software.
This proposed algorithm allows an average bitrate reduction of about 0.7% and improves the PSNR of the
decoded frame by 0.05%, 0.30% and 0.35% in luminance and chroma.
Design of Quadrature Mirror Filter Bank using Particle Swarm Optimization (PSO)IDES Editor
In this paper, the particle swarm optimization
technique is used for the design of a two channel quadrature
mirror filter (QMF) bank. A new method is developed to
optimize the prototype filter response in passband, stopband
and overall filter bank response. The design problem is
formulated as nonlinear unconstrained optimization of an
objective function, which is weighted sum of square of error
in passband, stop band and overall filter bank response at
frequency ( ω=0.5π ). For solving the given optimization
problem, the particle swarm optimization (PSO) technique
is used. As compared to the conventional design techniques,
the proposed method gives better performance in terms of
reconstruction error, mean square error in passband,
stopband, and computational time. Various design examples
are presented to illustrate the benefits provided by the
proposed method.
Analysis of low pdp using SPST in bilateral filterIJTET Journal
A FPGA Implementation of a bilateral filter for image processing is given which does spatial averaging without smoothing edges. Kernel based processing is possible, which means that processing of the entire filter window at one pixel clock cycle. It is also supported by the arrangement of the input data into groups and applied a single clock cycle for a group of pixels. Based on these features, a technique called Spurious Power Suppression Technique (SPST) is implemented in Bilateral Filter to minimize Power Delay Product (PDP). SPST can also dramatically reduce the power by turn off its MSP (Most significant bit) without compromising the computational results to achieve the target parameters. Furthermore, an Original Glitch Diminishing technique is proposed to filter out useless switching power by asserting signals after the data transient period. The SPST can be expanded to a Fine-grain scheme in which the combinational circuits are divided into more than two parts. In Bilateral Filter a kernel of different size can be implemented using SPST which achieve good performance.
1. The document presents a design for a modified Booth recoder using a fused add-multiply (FAM) operator to implement digital signal processing applications more efficiently.
2. It proposes a new recoding technique to decrease the critical path delay and reduce area and power consumption of the FAM unit compared to existing recoding schemes.
3. The technique is also applied to the implementation of finite impulse response (FIR) filters to further optimize hardware usage and achieve faithfully rounded outputs within tight area and power constraints for mobile applications.
D ESIGN A ND I MPLEMENTATION OF D IGITAL F ILTER B ANK T O R EDUCE N O...sipij
The main theme of this paper is to reduce noise fro
m the noisy composite signal and reconstruct the in
put
signals from the composite signal by designing FIR
digital filter bank. In this work, three sinusoidal
signals
of different frequencies and amplitudes are combine
d to get composite signal and a low frequency noise
signal is added with the composite signal to get no
isy composite signal. Finally noisy composite signa
l is
filtered by using FIR digital filter bank to reduce
noise and reconstruct the input signals
The document compares the performance of a Root Raised Cosine matched filter implemented using hybrid-logarithmic arithmetic versus standard binary and floating point arithmetic. Simulations showed that the hybrid logarithmic structure offered superior performance to fixed point solutions while having significantly reduced complexity compared to floating point equivalents. The use of hybrid logarithmic arithmetic also has the potential to reduce power consumption, latency, and hardware complexity for mobile applications.
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
DESIGN REALIZATION AND PERFORMANCE EVALUATION OF AN ACOUSTIC ECHO CANCELLATIO...sipij
Nowadays, in the field of communications, AEC (acoustic echo cancellation) is truly essential with respect
to the quality of multimedia transmission. In this paper, we designed and developed an efficient AEC based
on adaptive filters to improve quality of service in telecommunications against the phenomena of acoustic
echo, which is indeed a problem in hands-free communications.The main advantage of the proposed algorithm is its capacity of tracking non-stationary signals such as acoustic echo. In this work the acoustic echo cancellation (AEC) is modeled using a digital signal
processing technique especially Simulink Blocksets. The algorithm’s code is generated in Matlab Simulink
programming environment. At simulation level, results of simulink implementation prove that module
behavior is realistic when it comes to cancellation of echo in hands free communication using adaptive algorithm.Results obtained with our algorithm in terms of ERLE criteria are confronted to IUT-T recommendation
G.168.
1) The document describes the design of a compensation filter for a Generalized Comb Filter (GCF) using a Maximally-Flat minimization technique.
2) The coefficients of the proposed compensation filter are obtained by solving two linear equations to minimize passband droop.
3) The compensation filter operates at a lower rate than the GCF and significantly reduces passband droop when cascaded with the GCF.
An Efficient Reconfigurable Filter Design for Reducing Dynamic PowerEditor IJCATR
This paper presents an architectural view of designing a digital filter. The main idea is to design a reconfigurable filter for reducing dynamic
power consumption. By considering the input variation’s we reduce the order of the filter considering the coefficient are fixed. The filter is implemented
using mentor graphics using TSMC .18um technology. The power consumption is decreased in the rate of 16% from the conventional model with a slight
increase in area overhead. If the filter coefficients are fixed then the power can be reduced up to 18% and the area overhead can also be reduced from the
reconfigurable architecture.
Matlab Implementation of Baseline JPEG Image Compression Using Hardware Optim...inventionjournals
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
Design of an Adaptive Hearing Aid Algorithm using Booth-Wallace Tree MultiplierWaqas Tariq
The paper presents FPGA implementation of a spectral sharpening process suitable for speech enhancement and noise reduction algorithms for digital hearing aids. Booth and Booth Wallace multiplier is used for implementing digital signal processing algorithms in hearing aids. VHDL simulation results confirm that Booth Wallace multiplier is hardware efficient and performs faster than Booth’s multiplier. Booth Wallace multiplier consumes 40% less power compared to Booth multiplier. A novel digital hearing aid using spectral sharpening filter employing booth Wallace multiplier is proposed. The results reveal that the hardware requirement for implementing hearing aid using Booth Wallace multiplier is less when compared with that of a booth multiplier. Furthermore it is also demonstrated that digital hearing aid using Booth Wallace multiplier consumes less power and performs better in terms of speed.
The document describes a novel FPGA implementation of an image scaling processor using bilinear interpolation. The proposed method uses sharpening and clamp filters as pre-filters to the bilinear interpolator in order to improve image quality during scaling. The bilinear interpolation algorithm was chosen due to its lower complexity compared to other methods. The design was implemented in Verilog and synthesized for an FPGA, achieving a maximum frequency of 215.64MHz for 64x64 grayscale images. The hardware resources required were moderately lower than other algorithms.
International Journal of Engineering Inventions (IJEI) provides a multidisciplinary passage for researchers, managers, professionals, practitioners and students around the globe to publish high quality, peer-reviewed articles on all theoretical and empirical aspects of Engineering and Science.
The peer-reviewed International Journal of Engineering Inventions (IJEI) is started with a mission to encourage contribution to research in Science and Technology. Encourage and motivate researchers in challenging areas of Sciences and Technology.
A digital calibration algorithm with variable amplitude dithering for domain-...VLSICS Design
This document summarizes a digital calibration algorithm with variable-amplitude dithering for domain-extended pipeline analog-to-digital converters (ADCs). The algorithm aims to overcome limitations of existing dither-based calibration techniques by allowing for higher dither amplitudes without reducing the input signal amplitude. This improves convergence speed. The algorithm is also suitable for domain-extended pipeline ADCs as they provide more redundancy space to correct comparator offsets. Simulation results on a 12-bit, 100 MS/s pipeline ADC show improved static and dynamic performance after calibration using the proposed algorithm.
This document presents a new adaptive algorithm for an adaptive decision feedback equalizer (ADFE) that has lower computational complexity than existing algorithms. The proposed block-based normalized least mean square (BBNLMS) algorithm with set-membership filtering for the ADFE achieves similar bit error rate performance and convergence speed as conventional algorithms like set-membership normalized least mean square (SM-NLMS), but with significantly fewer computations. Simulation results show the new algorithm provides comparable equalization performance to SM-NLMS while realizing about a 70% reduction in computational operations, especially at high signal-to-noise ratios, making it suitable for high-speed decision feedback equalization applications.
RK7(5) is a numerical method was used to minimize the local truncation error using selection step size algorithm.
Time multiplexing CNN simulator was modied using RK7(5) to improve the performance of the simulator and the
quality of the output image for edge detection. The results showed better performance than those in the literature.
Implementation of Low-Complexity Redundant Multiplier Architecture for Finite...ijcisjournal
In the present work, a low-complexity Digit-Serial/parallel Multiplier over Finite Field is proposed. It is
employed in applications like cryptography for data encryption and decryptionto deal with discrete
mathematical andarithmetic structures. The proposedmultiplier utilizes a redundant representation because
of their free squaring and modular reduction. The proposed 10-bit multiplier is simulated and synthesized
using Xilinx VerilogHDL. It is evident from the simulation results that the multiplier has significantly low
area and power when compared to the previous structures using the same representation.
Similar to Two-Channel Quadrature Mirror Filter Bank Design using FIR Polyphase Component (20)
Power System State Estimation - A ReviewIDES Editor
This document provides a review of power system state estimation techniques. It discusses both static and dynamic state estimation algorithms. For static state estimation, it covers weighted least squares, decoupled, and robust estimation methods. Weighted least squares is commonly used but can have numerical instability issues. Decoupled state estimation approximates the gain matrix for faster computation. Robust estimation uses M-estimators and other techniques to handle outliers and bad data. Dynamic state estimation applies Kalman filtering, leapfrog algorithms, and other methods to continuously monitor system states over time.
Artificial Intelligence Technique based Reactive Power Planning Incorporating...IDES Editor
This document summarizes a research paper that proposes using artificial intelligence techniques and FACTS controllers for reactive power planning in real-time power transmission systems. The paper formulates the reactive power planning problem and incorporates flexible AC transmission system (FACTS) devices like static VAR compensators (SVC), thyristor controlled series capacitors (TCSC), and unified power flow controllers (UPFC). Evolutionary algorithms like evolutionary programming (EP) and differential evolution (DE) are applied to find the optimal locations and settings of the FACTS controllers to minimize losses and costs. Simulation results on IEEE 30-bus and 72-bus Indian test systems show that UPFC performs best in reducing losses compared to SVC and TCSC.
Design and Performance Analysis of Genetic based PID-PSS with SVC in a Multi-...IDES Editor
Damping of power system oscillations with the help
of proposed optimal Proportional Integral Derivative Power
System Stabilizer (PID-PSS) and Static Var Compensator
(SVC)-based controllers are thoroughly investigated in this
paper. This study presents robust tuning of PID-PSS and
SVC-based controllers using Genetic Algorithms (GA) in
multi machine power systems by considering detailed model
of the generators (model 1.1). The effectiveness of FACTSbased
controllers in general and SVC-based controller in
particular depends upon their proper location. Modal
controllability and observability are used to locate SVC–based
controller. The performance of the proposed controllers is
compared with conventional lead-lag power system stabilizer
(CPSS) and demonstrated on 10 machines, 39 bus New England
test system. Simulation studies show that the proposed genetic
based PID-PSS with SVC based controller provides better
performance.
Optimal Placement of DG for Loss Reduction and Voltage Sag Mitigation in Radi...IDES Editor
This paper presents the need to operate the power
system economically and with optimum levels of voltages has
further led to an increase in interest in Distributed
Generation. In order to reduce the power losses and to improve
the voltage in the distribution system, distributed generators
(DGs) are connected to load bus. To reduce the total power
losses in the system, the most important process is to identify
the proper location for fixing and sizing of DGs. It presents a
new methodology using a new population based meta heuristic
approach namely Artificial Bee Colony algorithm(ABC) for
the placement of Distributed Generators(DG) in the radial
distribution systems to reduce the real power losses and to
improve the voltage profile, voltage sag mitigation. The power
loss reduction is important factor for utility companies because
it is directly proportional to the company benefits in a
competitive electricity market, while reaching the better power
quality standards is too important as it has vital effect on
customer orientation. In this paper an ABC algorithm is
developed to gain these goals all together. In order to evaluate
sag mitigation capability of the proposed algorithm, voltage
in voltage sensitive buses is investigated. An existing 20KV
network has been chosen as test network and results are
compared with the proposed method in the radial distribution
system.
Line Losses in the 14-Bus Power System Network using UPFCIDES Editor
Controlling power flow in modern power systems
can be made more flexible by the use of recent developments
in power electronic and computing control technology. The
Unified Power Flow Controller (UPFC) is a Flexible AC
transmission system (FACTS) device that can control all the
three system variables namely line reactance, magnitude and
phase angle difference of voltage across the line. The UPFC
provides a promising means to control power flow in modern
power systems. Essentially the performance depends on proper
control setting achievable through a power flow analysis
program. This paper presents a reliable method to meet the
requirements by developing a Newton-Raphson based load
flow calculation through which control settings of UPFC can
be determined for the pre-specified power flow between the
lines. The proposed method keeps Newton-Raphson Load Flow
(NRLF) algorithm intact and needs (little modification in the
Jacobian matrix). A MATLAB program has been developed to
calculate the control settings of UPFC and the power flow
between the lines after the load flow is converged. Case studies
have been performed on IEEE 5-bus system and 14-bus system
to show that the proposed method is effective. These studies
indicate that the method maintains the basic NRLF properties
such as fast computational speed, high degree of accuracy and
good convergence rate.
Study of Structural Behaviour of Gravity Dam with Various Features of Gallery...IDES Editor
The size and shape of opening in dam causes the
stress concentration, it also causes the stress variation in the
rest of the dam cross section. The gravity method of the analysis
does not consider the size of opening and the elastic property
of dam material. Thus the objective of study is comprises of
the Finite Element Method which considers the size of
opening, elastic property of material, and stress distribution
because of geometric discontinuity in cross section of dam.
Stress concentration inside the dam increases with the opening
in dam which results in the failure of dam. Hence it is
necessary to analyses large opening inside the dam. By making
the percentage area of opening constant and varying size and
shape of opening the analysis is carried out. For this purpose
a section of Koyna Dam is considered. Dam is defined as a
plane strain element in FEM, based on geometry and loading
condition. Thus this available information specified our path
of approach to carry out 2D plane strain analysis. The results
obtained are then compared mutually to get most efficient
way of providing large opening in the gravity dam.
Assessing Uncertainty of Pushover Analysis to Geometric ModelingIDES Editor
Pushover Analysis a popular tool for seismic
performance evaluation of existing and new structures and is
nonlinear Static procedure where in monotonically increasing
loads are applied to the structure till the structure is unable
to resist the further load .During the analysis, whatever the
strength of concrete and steel is adopted for analysis of
structure may not be the same when real structure is
constructed and the pushover analysis results are very sensitive
to material model adopted, geometric model adopted, location
of plastic hinges and in general to procedure followed by the
analyzer. In this paper attempt has been made to assess
uncertainty in pushover analysis results by considering user
defined hinges and frame modeled as bare frame and frame
with slab modeled as rigid diaphragm and results compared
with experimental observations. Uncertain parameters
considered includes the strength of concrete, strength of steel
and cover to the reinforcement which are randomly generated
and incorporated into the analysis. The results are then
compared with experimental observations.
Secure Multi-Party Negotiation: An Analysis for Electronic Payments in Mobile...IDES Editor
This document summarizes and analyzes secure multi-party negotiation protocols for electronic payments in mobile computing. It presents a framework for secure multi-party decision protocols using lightweight implementations. The main focus is on synchronizing security features to avoid agreement manipulation and reduce user traffic. The paper describes negotiation between an auctioneer and bidders, showing multiparty security is better than existing systems. It analyzes the performance of encryption algorithms like ECC, XTR, and RSA for use in the multiparty negotiation protocols.
Selfish Node Isolation & Incentivation using Progressive ThresholdsIDES Editor
The problems associated with selfish nodes in
MANET are addressed by a collaborative watchdog approach
which reduces the detection time for selfish nodes thereby
improves the performance and accuracy of watchdogs[1]. In
the related works they make use of credit based systems, reputation
based mechanisms, pathrater and watchdog mechanism
to detect such selfish nodes. In this paper we follow an approach
of collaborative watchdog which reduces the detection
time for selfish nodes and also involves the removal of such
selfish nodes based on some progressively assessed thresholds.
The threshold gives the nodes a chance to stop misbehaving
before it is permanently deleted from the network.
The node passes through several isolation processes before it
is permanently removed. Another version of AODV protocol
is used here which allows the simulation of selfish nodes in
NS2 by adding or modifying log files in the protocol.
Various OSI Layer Attacks and Countermeasure to Enhance the Performance of WS...IDES Editor
Wireless sensor networks are networks having non
wired infrastructure and dynamic topology. In OSI model each
layer is prone to various attacks, which halts the performance
of a network .In this paper several attacks on four layers of
OSI model are discussed and security mechanism is described
to prevent attack in network layer i.e wormhole attack. In
Wormhole attack two or more malicious nodes makes a covert
channel which attracts the traffic towards itself by depicting a
low latency link and then start dropping and replaying packets
in the multi-path route. This paper proposes promiscuous mode
method to detect and isolate the malicious node during
wormhole attack by using Ad-hoc on demand distance vector
routing protocol (AODV) with omnidirectional antenna. The
methodology implemented notifies that the nodes which are
not participating in multi-path routing generates an alarm
message during delay and then detects and isolate the
malicious node from network. We also notice that not only
the same kind of attacks but also the same kind of
countermeasures can appear in multiple layer. For example,
misbehavior detection techniques can be applied to almost all
the layers we discussed.
Responsive Parameter based an AntiWorm Approach to Prevent Wormhole Attack in...IDES Editor
The recent advancements in the wireless technology
and their wide-spread deployment have made remarkable
enhancements in efficiency in the corporate and industrial
and Military sectors The increasing popularity and usage of
wireless technology is creating a need for more secure wireless
Ad hoc networks. This paper aims researched and developed
a new protocol that prevents wormhole attacks on a ad hoc
network. A few existing protocols detect wormhole attacks but
they require highly specialized equipment not found on most
wireless devices. This paper aims to develop a defense against
wormhole attacks as an Anti-worm protocol which is based on
responsive parameters, that does not require as a significant
amount of specialized equipment, trick clock synchronization,
no GPS dependencies.
Cloud Security and Data Integrity with Client Accountability FrameworkIDES Editor
This document summarizes a proposed cloud security and data integrity framework that provides client accountability. The framework aims to address issues like lack of user control over cloud data, need for data transparency and tracking, and ensuring data integrity. It proposes using JAR (Java Archive) files for data sharing due to benefits like portability. The framework incorporates client-side verification using MD5 hashing, digital signature-based authentication of JAR files, and use of HMAC to ensure data integrity. It also uses password-based encryption of log files to keep them tamper-proof. The framework is intended to provide both accountability and security for data sharing in cloud environments.
Genetic Algorithm based Layered Detection and Defense of HTTP BotnetIDES Editor
A System state in HTTP botnet uses HTTP protocol
for the creation of chain of Botnets thereby compromising
other systems. By using HTTP protocol and port number 80,
attacks can not only be hidden but also pass through the
firewall without being detected. The DPR based detection
leads to better analysis of botnet attacks [3]. However, it
provides only probabilistic detection of the attacker and also
time consuming and error prone. This paper proposes a Genetic
algorithm based layered approach for detecting as well as
preventing botnet attacks. The paper reviews p2p firewall
implementation which forms the basis of filtering.
Performance evaluation is done based on precision, F-value
and probability. Layered approach reduces the computation
and overall time requirement [7]. Genetic algorithm promises
a low false positive rate.
Enhancing Data Storage Security in Cloud Computing Through SteganographyIDES Editor
This document summarizes a research paper that proposes a method for enhancing data security in cloud computing through steganography. The method hides user data in digital images stored on cloud servers. When data needs to be accessed, it is extracted from the images. The document outlines the cloud architecture and security issues addressed. It then describes the proposed system architecture, security model, and data storage and retrieval process. Data is partitioned and hidden in multiple images to improve security. The goal is to prevent unauthorized access to user data stored on cloud servers.
The main tasks of a Wireless Sensor Network
(WSN) are data collection from its nodes and communication
of this data to the base station (BS). The protocols used for
communication among the WSN nodes and between the WSN
and the BS, must consider the resource constraints of nodes,
battery energy, computational capabilities and memory. The
WSN applications involve unattended operation of the network
over an extended period of time. In order to extend the lifetime
of a WSN, efficient routing protocols need to be adopted. The
proposed low power routing protocol based on tree-based
network structure reliably forwards the measured data towards
the BS using TDMA. An energy consumption analysis of the
WSN making use of this protocol is also carried out. It is
found that the network is energy efficient with an average
duty cycle of 0:7% for the WSN nodes. The OmNET++
simulation platform along with MiXiM framework is made
use of.
Permutation of Pixels within the Shares of Visual Cryptography using KBRP for...IDES Editor
The security of authentication of internet based
co-banking services should not be susceptible to high risks.
The passwords are highly vulnerable to virus attacks due to
the lack of high end embedding of security methods. In order
for the passwords to be more secure, people are generally
compelled to select jumbled up character based passwords
which are not only less memorable but are also equally prone
to insecurity. Multiple use of distributed shares has been
studied to solve the problem of authentication by algorithms
based on thresholding of pixels in image processing and visual
cryptography concepts where the subset of shares is considered
for the recovery of the original image for authentication using
correlation function[1][2].The main disadvantage in the above
study is the plain storage of shares and also one of the shares
is being supplied to the customer, which will lead to the
possibility of misuse by a third party. This paper proposes a
technique for scrambling of pixels by key based random
permutation (KBRP) within the shares before the
authentication has been attempted. Total number of shares to
be created is dependent on the multiplicity of ownership of
the account. By this method the problem of uncertainty among
the customers with regard to security, storage, retrieval of
holding of half of the shares is minimized.
This paper presents a trifocal Rotman Lens Design
approach. The effects of focal ratio and element spacing on
the performance of Rotman Lens are described. A three beam
prototype feeding 4 element antenna array working in L-band
has been simulated using RLD v1.7 software. Simulated
results show that the simulated lens has a return loss of –
12.4dB at 1.8GHz. Beam to array port phase error variation
with change in the focal ratio and element spacing has also
been investigated.
Band Clustering for the Lossless Compression of AVIRIS Hyperspectral ImagesIDES Editor
Hyperspectral images can be efficiently compressed
through a linear predictive model, as for example the one
used in the SLSQ algorithm. In this paper we exploit this
predictive model on the AVIRIS images by individuating,
through an off-line approach, a common subset of bands, which
are not spectrally related with any other bands. These bands
are not useful as prediction reference for the SLSQ 3-D
predictive model and we need to encode them via other
prediction strategies which consider only spatial correlation.
We have obtained this subset by clustering the AVIRIS bands
via the clustering by compression approach. The main result
of this paper is the list of the bands, not related with the
others, for AVIRIS images. The clustering trees obtained for
AVIRIS and the relationship among bands they depict is also
an interesting starting point for future research.
Microelectronic Circuit Analogous to Hydrogen Bonding Network in Active Site ...IDES Editor
A microelectronic circuit of block-elements
functionally analogous to two hydrogen bonding networks is
investigated. The hydrogen bonding networks are extracted
from â-lactamase protein and are formed in its active site.
Each hydrogen bond of the network is described in equivalent
electrical circuit by three or four-terminal block-element.
Each block-element is coded in Matlab. Static and dynamic
analyses are performed. The resultant microelectronic circuit
analogous to the hydrogen bonding network operates as
current mirror, sine pulse source, triangular pulse source as
well as signal modulator.
Texture Unit based Monocular Real-world Scene Classification using SOM and KN...IDES Editor
In this paper a method is proposed to discriminate
real world scenes in to natural and manmade scenes of similar
depth. Global-roughness of a scene image varies as a function
of image-depth. Increase in image depth leads to increase in
roughness in manmade scenes; on the contrary natural scenes
exhibit smooth behavior at higher image depth. This particular
arrangement of pixels in scene structure can be well explained
by local texture information in a pixel and its neighborhood.
Our proposed method analyses local texture information of a
scene image using texture unit matrix. For final classification
we have used both supervised and unsupervised learning using
K-Nearest Neighbor classifier (KNN) and Self Organizing
Map (SOM) respectively. This technique is useful for online
classification due to very less computational complexity.
Introduction to AI for Nonprofits with Tapp NetworkTechSoup
Dive into the world of AI! Experts Jon Hill and Tareq Monaur will guide you through AI's role in enhancing nonprofit websites and basic marketing strategies, making it easy to understand and apply.
ISO/IEC 27001, ISO/IEC 42001, and GDPR: Best Practices for Implementation and...PECB
Denis is a dynamic and results-driven Chief Information Officer (CIO) with a distinguished career spanning information systems analysis and technical project management. With a proven track record of spearheading the design and delivery of cutting-edge Information Management solutions, he has consistently elevated business operations, streamlined reporting functions, and maximized process efficiency.
Certified as an ISO/IEC 27001: Information Security Management Systems (ISMS) Lead Implementer, Data Protection Officer, and Cyber Risks Analyst, Denis brings a heightened focus on data security, privacy, and cyber resilience to every endeavor.
His expertise extends across a diverse spectrum of reporting, database, and web development applications, underpinned by an exceptional grasp of data storage and virtualization technologies. His proficiency in application testing, database administration, and data cleansing ensures seamless execution of complex projects.
What sets Denis apart is his comprehensive understanding of Business and Systems Analysis technologies, honed through involvement in all phases of the Software Development Lifecycle (SDLC). From meticulous requirements gathering to precise analysis, innovative design, rigorous development, thorough testing, and successful implementation, he has consistently delivered exceptional results.
Throughout his career, he has taken on multifaceted roles, from leading technical project management teams to owning solutions that drive operational excellence. His conscientious and proactive approach is unwavering, whether he is working independently or collaboratively within a team. His ability to connect with colleagues on a personal level underscores his commitment to fostering a harmonious and productive workplace environment.
Date: May 29, 2024
Tags: Information Security, ISO/IEC 27001, ISO/IEC 42001, Artificial Intelligence, GDPR
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Training: ISO/IEC 27001 Information Security Management System - EN | PECB
ISO/IEC 42001 Artificial Intelligence Management System - EN | PECB
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Webinars: https://pecb.com/webinars
Article: https://pecb.com/article
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Executive Directors Chat Leveraging AI for Diversity, Equity, and InclusionTechSoup
Let’s explore the intersection of technology and equity in the final session of our DEI series. Discover how AI tools, like ChatGPT, can be used to support and enhance your nonprofit's DEI initiatives. Participants will gain insights into practical AI applications and get tips for leveraging technology to advance their DEI goals.
বাংলাদেশের অর্থনৈতিক সমীক্ষা ২০২৪ [Bangladesh Economic Review 2024 Bangla.pdf] কম্পিউটার , ট্যাব ও স্মার্ট ফোন ভার্সন সহ সম্পূর্ণ বাংলা ই-বুক বা pdf বই " সুচিপত্র ...বুকমার্ক মেনু 🔖 ও হাইপার লিংক মেনু 📝👆 যুক্ত ..
আমাদের সবার জন্য খুব খুব গুরুত্বপূর্ণ একটি বই ..বিসিএস, ব্যাংক, ইউনিভার্সিটি ভর্তি ও যে কোন প্রতিযোগিতা মূলক পরীক্ষার জন্য এর খুব ইম্পরট্যান্ট একটি বিষয় ...তাছাড়া বাংলাদেশের সাম্প্রতিক যে কোন ডাটা বা তথ্য এই বইতে পাবেন ...
তাই একজন নাগরিক হিসাবে এই তথ্য গুলো আপনার জানা প্রয়োজন ...।
বিসিএস ও ব্যাংক এর লিখিত পরীক্ষা ...+এছাড়া মাধ্যমিক ও উচ্চমাধ্যমিকের স্টুডেন্টদের জন্য অনেক কাজে আসবে ...
Macroeconomics- Movie Location
This will be used as part of your Personal Professional Portfolio once graded.
Objective:
Prepare a presentation or a paper using research, basic comparative analysis, data organization and application of economic information. You will make an informed assessment of an economic climate outside of the United States to accomplish an entertainment industry objective.
How to Build a Module in Odoo 17 Using the Scaffold MethodCeline George
Odoo provides an option for creating a module by using a single line command. By using this command the user can make a whole structure of a module. It is very easy for a beginner to make a module. There is no need to make each file manually. This slide will show how to create a module using the scaffold method.
Assessment and Planning in Educational technology.pptxKavitha Krishnan
In an education system, it is understood that assessment is only for the students, but on the other hand, the Assessment of teachers is also an important aspect of the education system that ensures teachers are providing high-quality instruction to students. The assessment process can be used to provide feedback and support for professional development, to inform decisions about teacher retention or promotion, or to evaluate teacher effectiveness for accountability purposes.
This presentation includes basic of PCOS their pathology and treatment and also Ayurveda correlation of PCOS and Ayurvedic line of treatment mentioned in classics.
Main Java[All of the Base Concepts}.docxadhitya5119
This is part 1 of my Java Learning Journey. This Contains Custom methods, classes, constructors, packages, multithreading , try- catch block, finally block and more.
Exploiting Artificial Intelligence for Empowering Researchers and Faculty, In...Dr. Vinod Kumar Kanvaria
Exploiting Artificial Intelligence for Empowering Researchers and Faculty,
International FDP on Fundamentals of Research in Social Sciences
at Integral University, Lucknow, 06.06.2024
By Dr. Vinod Kumar Kanvaria