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.
Research Inventy : International Journal of Engineering and Scienceinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
In this paper, we provide the average bit error probabilities of MQAM and MPSK in the presence of log normal shadowing using Maximal Ratio Combining technique for L diversity branches. We have derived probability of density function (PDF) of received signal to noise ratio (SNR) for L diversity branches in Log Normal fadingfor Maximal Ratio Combining (MRC). We have used Fenton-Wilkinson method to estimate the parameters for a single log-normal distribution that approximates the sum of log-normal random variables (RVs). The results that we provide in this paper are an important tool for measuring the performance ofcommunication links in a log-normal shadowing.
Memory Polynomial Based Adaptive Digital PredistorterIJERA Editor
Digital predistortion (DPD) is a baseband signal processing technique that corrects for impairments in RF
power amplifiers (PAs). These impairments cause out-of-band emissions or spectral regrowth and in-band
distortion, which correlate with an increased bit error rate (BER). Wideband signals with a high peak-to-average
ratio, are more susceptible to these unwanted effects. So to reduce these impairments, this paper proposes the
modeling of the digital predistortion for the power amplifier using GSA algorithm.
Direction of arrival estimation using music algorithmeSAT Journals
Abstract The performance of smart antenna greatly depends on the effectiveness of DOA estimation algorithm. This paper analyzed the performance of MUSIC (Multiple Signal Classification) algorithm for DOA estimation. simulation results shows that MUSIC provide better angular resolution for increasing number of array element, distance between array element and number of samples. All the simulations are carried out using MATLAB. Keywords: DOA (Direction of arrival), MUSIC (Multiple signal classification), ULA(Uniform linear array)
Subspace based doa estimation techniqueseSAT Journals
Abstract The subspace based techniques are used for Direction of Arrival (DOA) estimation in this work. The subspace based techniques are based on using the eigen structure of data covariance matrix. The subspace based techniques includes MUSIC, ROOT-MUSIC, ESPRIT. The aim is to analyze the performance of DOA estimation algorithms in challenging environment, such as low signal to noise ratio, closely spaced sources. The performance of subspace based DOA estimation algorithm is done on Uniform Linear Array (ULA). Simulation result shows the effect of varying parameters will affect DOA estimation. The simulation shows that the MUSIC algorithm has better accuracy as compared to the Root-MUSIC and ESPRIT. Keywords: DOA, MUSIC, Root-MUSIC, and ESPRIT
Evaluation of channel estimation combined with ICI self-cancellation scheme i...ijcsse
Orthogonal Frequency Division Multiplexing (OFDM) is a modulation scheme, which is used in several wireless systems for transferring data at high rate. The multi path fading channel and the frequency offset between the transmitted and received carrier frequencies introduce ICI (Inter Carrier Interference). ICI effects the OFDM symbols and degrades the system performance. This paper proposes a solution: combine channel estimation and ICI self-cancellation to combat against ICI in doubly selective fading channel. The simulation results show the effect of this solution
Research Inventy : International Journal of Engineering and Scienceinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
In this paper, we provide the average bit error probabilities of MQAM and MPSK in the presence of log normal shadowing using Maximal Ratio Combining technique for L diversity branches. We have derived probability of density function (PDF) of received signal to noise ratio (SNR) for L diversity branches in Log Normal fadingfor Maximal Ratio Combining (MRC). We have used Fenton-Wilkinson method to estimate the parameters for a single log-normal distribution that approximates the sum of log-normal random variables (RVs). The results that we provide in this paper are an important tool for measuring the performance ofcommunication links in a log-normal shadowing.
Memory Polynomial Based Adaptive Digital PredistorterIJERA Editor
Digital predistortion (DPD) is a baseband signal processing technique that corrects for impairments in RF
power amplifiers (PAs). These impairments cause out-of-band emissions or spectral regrowth and in-band
distortion, which correlate with an increased bit error rate (BER). Wideband signals with a high peak-to-average
ratio, are more susceptible to these unwanted effects. So to reduce these impairments, this paper proposes the
modeling of the digital predistortion for the power amplifier using GSA algorithm.
Direction of arrival estimation using music algorithmeSAT Journals
Abstract The performance of smart antenna greatly depends on the effectiveness of DOA estimation algorithm. This paper analyzed the performance of MUSIC (Multiple Signal Classification) algorithm for DOA estimation. simulation results shows that MUSIC provide better angular resolution for increasing number of array element, distance between array element and number of samples. All the simulations are carried out using MATLAB. Keywords: DOA (Direction of arrival), MUSIC (Multiple signal classification), ULA(Uniform linear array)
Subspace based doa estimation techniqueseSAT Journals
Abstract The subspace based techniques are used for Direction of Arrival (DOA) estimation in this work. The subspace based techniques are based on using the eigen structure of data covariance matrix. The subspace based techniques includes MUSIC, ROOT-MUSIC, ESPRIT. The aim is to analyze the performance of DOA estimation algorithms in challenging environment, such as low signal to noise ratio, closely spaced sources. The performance of subspace based DOA estimation algorithm is done on Uniform Linear Array (ULA). Simulation result shows the effect of varying parameters will affect DOA estimation. The simulation shows that the MUSIC algorithm has better accuracy as compared to the Root-MUSIC and ESPRIT. Keywords: DOA, MUSIC, Root-MUSIC, and ESPRIT
Evaluation of channel estimation combined with ICI self-cancellation scheme i...ijcsse
Orthogonal Frequency Division Multiplexing (OFDM) is a modulation scheme, which is used in several wireless systems for transferring data at high rate. The multi path fading channel and the frequency offset between the transmitted and received carrier frequencies introduce ICI (Inter Carrier Interference). ICI effects the OFDM symbols and degrades the system performance. This paper proposes a solution: combine channel estimation and ICI self-cancellation to combat against ICI in doubly selective fading channel. The simulation results show the effect of this solution
Enriched Firefly Algorithm for Solving Reactive Power Problemijeei-iaes
In this paper, Enriched Firefly Algorithm (EFA) is planned to solve optimal reactive power dispatch problem. This algorithm is a kind of swarm intelligence algorithm based on the response of a firefly to the light of other fireflies. In this paper, we plan an augmentation on the original firefly algorithm. The proposed algorithm extends the single population FA to the interacting multi-swarms by cooperative Models. The proposed EFA has been tested on standard IEEE 30 bus test system and simulation results show clearly the better performance of the proposed algorithm in reducing the real power loss.
The Baugh-Wooley algorithm is a well-known iterative algorithm for performing multiplication in digital signal processing applications. Decomposition logic is used with Baugh-Wooley algorithm to enhance the speed and to reduce the critical path delay. In this paper a high speed multiplier is designed and implemented using decomposition logic and Baugh-Wooley algorithm. The result is compared with booth
multiplier. FPGA based architecture is presented and design has been implemented using Xilinx 12.3 device.
FPGA IMPLEMENTATION OF HIGH SPEED BAUGH-WOOLEY MULTIPLIER USING DECOMPOSITION...eeiej_journal
The Baugh-Wooley algorithm is a well-known iterative algorithm for performing multiplication in digital signal processing applications. Decomposition logic is used with Baugh-Wooley algorithm to enhance the
speed and to reduce the critical path delay. In this paper a high speed multiplier is designed and implemented using decomposition logic and Baugh-Wooley algorithm. The result is compared with booth multiplier. FPGA based architecture is presented and design has been implemented using Xilinx 12.3
device.
Performance of MMSE Denoise Signal Using LS-MMSE TechniqueIJMER
This paper presents performance of mmse denoises signal using consistent cycle spinning (ccs) and least square (LS) techniques. In the past decade, TV denoise technique is used to reduced the noisy signal. The main drawback is the low quality signal and high MMSE signal. Presently, we
proposed the CCS-MMSE and LS-MMSE technique .The CCS-MMSE technique consists of two steps. They are wavelet based denoise and consistent cycle spinning. The wavelet denoise is powerful decorrelating effect on many signal domains. The consistent cycle spinning is used to estimation the
MMSE in the signal domain. The LS-MMSE is better estimation of MMSE signal domain compare to
CCS-MMSE.The experimental result shows the average MMSE reduction using various techniques.
Improving Performance of Back propagation Learning Algorithmijsrd.com
The standard back-propagation algorithm is one of the most widely used algorithm for training feed-forward neural networks. One major drawback of this algorithm is it might fall into local minima and slow convergence rate. Natural gradient descent is principal method for solving nonlinear function is presented and is combined with the modified back-propagation algorithm yielding a new fast training multilayer algorithm. This paper describes new approach to natural gradient learning in which the number of parameters necessary is much smaller than the natural gradient algorithm. This new method exploits the algebraic structure of the parameter space to reduce the space and time complexity of algorithm and improve its performance.
A Threshold Enhancement Technique for Chaotic On-Off Keying SchemeCSCJournals
In this paper, an improvement for Chaotic ON-OFF (COOK) Keying scheme is proposed. The scheme enhances Bit Error Rate (BER) performance of standard COOK by keeping the signal elements at fixed distance from the threshold irrespective of noise power. Each transmitted chaotic segment is added to its flipped version before transmission. This reduces the effect of noise contribution at correlator of the receiver. The proposed system is tested in Additive White Gaussian Noise (AWGN) channel and compared with the standard COOK under different Eb/No levels. A theoretical estimate of BER is derived and compared with the simulation results. Effect of spreading factor increment in the proposed system is studied. Results show that the proposed scheme has a considerable advantage over the standard COOK at similar average bit energy and with higher values of spreading factors.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
One of the important steps in routing is to find a feasible path based on the state information. In order to support real-time multimedia applications, the feasible path that satisfies one or more constraints has to be computed within a very short time. Therefore, the paper presents a genetic algorithm to solve the paths tree problem subject to cost constraints. The objective of the algorithm is to find the set of edges connecting all nodes such that the sum of the edge costs from the source (root) to each node is minimized. I.e. the path from the root to each node must be a minimum cost path connecting them. The algorithm has been applied on two sample networks, the first network with eight nodes, and the last one with eleven nodes to illustrate its efficiency.
Firefly Algorithm to Opmimal Distribution of Reactive Power Compensation Units IJECEIAES
The issue of electric power grid mode of optimization is one of the basic directions in power engineering research. Currently, methods other than classical optimization methods based on various bio-heuristic algorithms are applied. The problems of reactive power optimization in a power grid using bio-heuristic algorithms are considered. These algorithms allow obtaining more efficient solutions as well as taking into account several criteria. The Firefly algorithm is adapted to optimize the placement of reactive power sources as well as to select their values. A key feature of the proposed modification of the Firefly algorithm is the solution for the multi-objective optimization problem. Algorithms based on a bio-heuristic process can find a neighborhood of global extreme, so a local gradient descent in the neighborhood is applied for a more accurate solution of the problem. Comparison of gradient descent, Firefly algorithm and Firefly algorithm with gradient descent is carried out.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Applic...IJTET Journal
An efficient architecture for the implementation of a delayed least mean square adaptive filter. A Novel
partial product Generator is achieving lower adaptation-delay and Area delay consumption and propose a strategy
for optimized balanced pipelining across the time-consuming combinational blocks of the structure. From synthesis
results, the proposed design will offers less area-delay product (ADP) the best of the existing systolic structures, on
average, for filter lengths N =8, 16, and 32. An efficient fixed-point implementation scheme of the proposed
architecture, The EEG(electroencephalogram) is used for recording of electrical activity of the brain .During
recording the EEG is contaminated by various artifacts as PLI(Power line interference), MA(Muscle artifact),
EBA(Eye blink artifact). This paper gives Detail of various artifacts which occur in EEG signal. In this we study
adaptive filter for reducing the EBA (eye blink artifact) noise from the EEG signal and to increase SNR (Signal to
noise ratio).the analytical result matches with the simulation result is showed.
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.
Enriched Firefly Algorithm for Solving Reactive Power Problemijeei-iaes
In this paper, Enriched Firefly Algorithm (EFA) is planned to solve optimal reactive power dispatch problem. This algorithm is a kind of swarm intelligence algorithm based on the response of a firefly to the light of other fireflies. In this paper, we plan an augmentation on the original firefly algorithm. The proposed algorithm extends the single population FA to the interacting multi-swarms by cooperative Models. The proposed EFA has been tested on standard IEEE 30 bus test system and simulation results show clearly the better performance of the proposed algorithm in reducing the real power loss.
The Baugh-Wooley algorithm is a well-known iterative algorithm for performing multiplication in digital signal processing applications. Decomposition logic is used with Baugh-Wooley algorithm to enhance the speed and to reduce the critical path delay. In this paper a high speed multiplier is designed and implemented using decomposition logic and Baugh-Wooley algorithm. The result is compared with booth
multiplier. FPGA based architecture is presented and design has been implemented using Xilinx 12.3 device.
FPGA IMPLEMENTATION OF HIGH SPEED BAUGH-WOOLEY MULTIPLIER USING DECOMPOSITION...eeiej_journal
The Baugh-Wooley algorithm is a well-known iterative algorithm for performing multiplication in digital signal processing applications. Decomposition logic is used with Baugh-Wooley algorithm to enhance the
speed and to reduce the critical path delay. In this paper a high speed multiplier is designed and implemented using decomposition logic and Baugh-Wooley algorithm. The result is compared with booth multiplier. FPGA based architecture is presented and design has been implemented using Xilinx 12.3
device.
Performance of MMSE Denoise Signal Using LS-MMSE TechniqueIJMER
This paper presents performance of mmse denoises signal using consistent cycle spinning (ccs) and least square (LS) techniques. In the past decade, TV denoise technique is used to reduced the noisy signal. The main drawback is the low quality signal and high MMSE signal. Presently, we
proposed the CCS-MMSE and LS-MMSE technique .The CCS-MMSE technique consists of two steps. They are wavelet based denoise and consistent cycle spinning. The wavelet denoise is powerful decorrelating effect on many signal domains. The consistent cycle spinning is used to estimation the
MMSE in the signal domain. The LS-MMSE is better estimation of MMSE signal domain compare to
CCS-MMSE.The experimental result shows the average MMSE reduction using various techniques.
Improving Performance of Back propagation Learning Algorithmijsrd.com
The standard back-propagation algorithm is one of the most widely used algorithm for training feed-forward neural networks. One major drawback of this algorithm is it might fall into local minima and slow convergence rate. Natural gradient descent is principal method for solving nonlinear function is presented and is combined with the modified back-propagation algorithm yielding a new fast training multilayer algorithm. This paper describes new approach to natural gradient learning in which the number of parameters necessary is much smaller than the natural gradient algorithm. This new method exploits the algebraic structure of the parameter space to reduce the space and time complexity of algorithm and improve its performance.
A Threshold Enhancement Technique for Chaotic On-Off Keying SchemeCSCJournals
In this paper, an improvement for Chaotic ON-OFF (COOK) Keying scheme is proposed. The scheme enhances Bit Error Rate (BER) performance of standard COOK by keeping the signal elements at fixed distance from the threshold irrespective of noise power. Each transmitted chaotic segment is added to its flipped version before transmission. This reduces the effect of noise contribution at correlator of the receiver. The proposed system is tested in Additive White Gaussian Noise (AWGN) channel and compared with the standard COOK under different Eb/No levels. A theoretical estimate of BER is derived and compared with the simulation results. Effect of spreading factor increment in the proposed system is studied. Results show that the proposed scheme has a considerable advantage over the standard COOK at similar average bit energy and with higher values of spreading factors.
International Journal of Engineering Research and Applications (IJERA) is a team of researchers not publication services or private publications running the journals for monetary benefits, we are association of scientists and academia who focus only on supporting authors who want to publish their work. The articles published in our journal can be accessed online, all the articles will be archived for real time access.
Our journal system primarily aims to bring out the research talent and the works done by sciaentists, academia, engineers, practitioners, scholars, post graduate students of engineering and science. This journal aims to cover the scientific research in a broader sense and not publishing a niche area of research facilitating researchers from various verticals to publish their papers. It is also aimed to provide a platform for the researchers to publish in a shorter of time, enabling them to continue further All articles published are freely available to scientific researchers in the Government agencies,educators and the general public. We are taking serious efforts to promote our journal across the globe in various ways, we are sure that our journal will act as a scientific platform for all researchers to publish their works online.
One of the important steps in routing is to find a feasible path based on the state information. In order to support real-time multimedia applications, the feasible path that satisfies one or more constraints has to be computed within a very short time. Therefore, the paper presents a genetic algorithm to solve the paths tree problem subject to cost constraints. The objective of the algorithm is to find the set of edges connecting all nodes such that the sum of the edge costs from the source (root) to each node is minimized. I.e. the path from the root to each node must be a minimum cost path connecting them. The algorithm has been applied on two sample networks, the first network with eight nodes, and the last one with eleven nodes to illustrate its efficiency.
Firefly Algorithm to Opmimal Distribution of Reactive Power Compensation Units IJECEIAES
The issue of electric power grid mode of optimization is one of the basic directions in power engineering research. Currently, methods other than classical optimization methods based on various bio-heuristic algorithms are applied. The problems of reactive power optimization in a power grid using bio-heuristic algorithms are considered. These algorithms allow obtaining more efficient solutions as well as taking into account several criteria. The Firefly algorithm is adapted to optimize the placement of reactive power sources as well as to select their values. A key feature of the proposed modification of the Firefly algorithm is the solution for the multi-objective optimization problem. Algorithms based on a bio-heuristic process can find a neighborhood of global extreme, so a local gradient descent in the neighborhood is applied for a more accurate solution of the problem. Comparison of gradient descent, Firefly algorithm and Firefly algorithm with gradient descent is carried out.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Applic...IJTET Journal
An efficient architecture for the implementation of a delayed least mean square adaptive filter. A Novel
partial product Generator is achieving lower adaptation-delay and Area delay consumption and propose a strategy
for optimized balanced pipelining across the time-consuming combinational blocks of the structure. From synthesis
results, the proposed design will offers less area-delay product (ADP) the best of the existing systolic structures, on
average, for filter lengths N =8, 16, and 32. An efficient fixed-point implementation scheme of the proposed
architecture, The EEG(electroencephalogram) is used for recording of electrical activity of the brain .During
recording the EEG is contaminated by various artifacts as PLI(Power line interference), MA(Muscle artifact),
EBA(Eye blink artifact). This paper gives Detail of various artifacts which occur in EEG signal. In this we study
adaptive filter for reducing the EBA (eye blink artifact) noise from the EEG signal and to increase SNR (Signal to
noise ratio).the analytical result matches with the simulation result is showed.
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.
FPGA IMPLEMENTATION OF NOISE CANCELLATION USING ADAPTIVE ALGORITHMSEditor IJMTER
This paper describes the concept of adaptive noise cancelling. The noise cancellation
using the Recursive Least Squares (RLS) to remove the noise from an input signal. The RLS adaptive
filter uses the reference signal on the Input port and the desired signal on the desired port to
automatically match the filter response in the Noise Filter block. The filtered noise should be completely
subtracted from the "noisy signal” of the input Sine wave & noise input signal, and the "Error Signal"
should contain only the original signal. Finally, the functions of field programmable gate array based
system structure for adaptive noise canceller based on RLS algorithm are synthesized, simulated, and
implemented on Xilinx XC3s200 field programmable gate array using Xilinx ISE tool.
Reducting Power Dissipation in Fir Filter: an AnalysisCSCJournals
In this paper, three existing techniques, Signed Power-of-Two (SPT), Steepest decent and Coefficient segmentation, for power reduction of FIR filters are analyzed. These techniques reduce switching activity which is directly related to the power consumption of a circuit. In an FIR filter, the multiplier consumes maximum power. Therefore, power consumption can be reduced either by by making the filter multiplier-less or by minimizing hamming distance between the coefficients of this multiplier as it directly translates into reduction in power dissipation [8]. The results obtained on four filters (LP) show that hamming distance can be reduced upto 26% and 47% in steepest decent and coefficient segmentation algorithm respectively. Multiplier-less filter can be realized by realizing coefficients in signed power-of-two terms, i.e. by shifting and adding the coefficients, though at the cost of shift operation overhead.
Comparative Analysis of Different Wavelet Functions using Modified Adaptive F...IJERA Editor
The traditional method of wavelet denoising is inefficient in removing the overlap noise between noisy signal
and noise, due to which a modified adaptive filtering based on wavelet transform method is introduced. The
method used in this paper filters out the noise on the basis of wavelet denoising using different wavelet
functions. The simulation results indicate the Signal to Noise ratio (SNR), Mean Square Error (MSE) and signal
error power spectral density comparison plot between different wavelet functions. These comparison results
verified that Daubechies is more efficient than other wavelet functions in filtering out noise in all perspectives.
International Journal of Engineering Research and Development (IJERD)IJERD Editor
journal publishing, how to publish research paper, Call For research paper, international journal, publishing a paper, IJERD, journal of science and technology, how to get a research paper published, publishing a paper, publishing of journal, publishing of research paper, reserach and review articles, IJERD Journal, How to publish your research paper, publish research paper, open access engineering journal, Engineering journal, Mathemetics journal, Physics journal, Chemistry journal, Computer Engineering, Computer Science journal, how to submit your paper, peer reviw journal, indexed journal, reserach and review articles, engineering journal, www.ijerd.com, research journals,
yahoo journals, bing journals, International Journal of Engineering Research and Development, google journals, hard copy of journal
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
IJCER (www.ijceronline.com) International Journal of computational Engineerin...ijceronline
Call for paper 2012, hard copy of Certificate, research paper publishing, where to publish research paper,
journal publishing, how to publish research paper, Call For research paper, international journal, publishing a paper, IJCER, journal of science and technology, how to get a research paper published, publishing a paper, publishing of journal, publishing of research paper, research and review articles, IJCER Journal, How to publish your research paper, publish research paper, open access engineering journal, Engineering journal, Mathematics journal, Physics journal, Chemistry journal, Computer Engineering, Computer Science journal, how to submit your paper, peer review journal, indexed journal, research and review articles, engineering journal, www.ijceronline.com, research journals,
yahoo journals, bing journals, International Journal of Computational Engineering Research, Google journals, hard copy of Certificate,
journal of engineering, online Submission
Low Power Adaptive FIR Filter Based on Distributed ArithmeticIJERA Editor
This paper aims at implementation of a low power adaptive FIR filter based on distributed arithmetic (DA) with
low power, high throughput, and low area. Least Mean Square (LMS) Algorithm is used to update the weight
and decrease the mean square error between the current filter output and the desired response. The pipelined
Distributed Arithmetic table reduces switching activity and hence it reduces power. The power consumption is
reduced by keeping bit-clock used in carry-save accumulation much faster than clock of rest of the operations.
We have implemented it in Quartus II and found that there is a reduction in the total power and the core dynamic
power by 31.31% and 100.24% respectively when compared with the architecture without DA table
Echo Cancellation Algorithms using Adaptive Filters: A Comparative Studyidescitation
An adaptive filter is a filter that self-adjusts its transfer function according to an
optimization algorithm driven by an error signal. Adaptive filter finds its essence in
applications such as echo cancellation, noise cancellation, system identification and many
others. This paper briefly discusses LMS, NLMS and RLS adaptive filter algorithms for
echo cancellation. For the analysis, an acoustic echo canceller is built using LMS, NLMS
and RLS algorithms and the echo cancelled samples are studied using Spectrogram. The
analysis is further extended with its cross-correlation and ERLE (Echo Return Loss
Enhancement) results. Finally, this paper concludes with a better adaptive filter algorithm
for Echo cancellation. The implementation and analysis is done using MATLAB®,
SIMULINK® and SPECTROGRAM V5.0®.
International Journal of Computational Engineering Research(IJCER)ijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
FPGA Design & Simulation Modeling of Baseband Data Transmission SystemIOSR Journals
Abstract: This paper describes a study on a baseband data transmission system developed for undergraduate
students studying communication engineering. Theoretical material, developed in the lectures, is briefly
covered. A practical system is presented with pre-detection filtering being employed to improve the bit error
rate. A simulation of the complete system is carried out on a Sun work station using the MATLAB simulation
package. Simulation and theoretical results are compared.
Performance Analysis of Acoustic Echo Cancellation TechniquesIJERA Editor
Mainly, the adaptive filters are implemented in time domain which works efficiently in most of the applications. But in many applications the impulse response becomes too large, which increases the complexity of the adaptive filter beyond a level where it can no longer be implemented efficiently in time domain. An example of where this can happen would be acoustic echo cancellation (AEC) applications. So, there exists an alternative solution i.e. to implement the filters in frequency domain. AEC has so many applications in wide variety of problems in industrial operations, manufacturing and consumer products. Here in this paper, a comparative analysis of different acoustic echo cancellation techniques i.e. Frequency domain adaptive filter (FDAF), Least mean square (LMS), Normalized least mean square (NLMS) &Sign error (SE) is presented. The results are compared with different values of step sizes and the performance of these techniques is measured in terms of Error rate loss enhancement (ERLE), Mean square error (MSE)& Peak signal to noise ratio (PSNR).
The Use of Java Swing’s Components to Develop a WidgetWaqas Tariq
Widget is a kind of application provides a single service such as a map, news feed, simple clock, battery-life indicators, etc. This kind of interactive software object has been developed to facilitate user interface (UI) design. A user interface (UI) function may be implemented using different widgets with the same function. In this article, we present the widget as a platform that is generally used in various applications, such as in desktop, web browser, and mobile phone. We also describe a visual menu of Java Swing’s components that will be used to establish widget. It will assume that we have successfully compiled and run a program that uses Swing components.
3D Human Hand Posture Reconstruction Using a Single 2D ImageWaqas Tariq
Passive sensing of the 3D geometric posture of the human hand has been studied extensively over the past decade. However, these research efforts have been hampered by the computational complexity caused by inverse kinematics and 3D reconstruction. In this paper, our objective focuses on 3D hand posture estimation based on a single 2D image with aim of robotic applications. We introduce the human hand model with 27 degrees of freedom (DOFs) and analyze some of its constraints to reduce the DOFs without any significant degradation of performance. A novel algorithm to estimate the 3D hand posture from eight 2D projected feature points is proposed. Experimental results using real images confirm that our algorithm gives good estimates of the 3D hand pose. Keywords: 3D hand posture estimation; Model-based approach; Gesture recognition; human- computer interface; machine vision.
Camera as Mouse and Keyboard for Handicap Person with Troubleshooting Ability...Waqas Tariq
Camera mouse has been widely used for handicap person to interact with computer. The utmost important of the use of camera mouse is must be able to replace all roles of typical mouse and keyboard. It must be able to provide all mouse click events and keyboard functions (include all shortcut keys) when it is used by handicap person. Also, the use of camera mouse must allow users troubleshooting by themselves. Moreover, it must be able to eliminate neck fatigue effect when it is used during long period. In this paper, we propose camera mouse system with timer as left click event and blinking as right click event. Also, we modify original screen keyboard layout by add two additional buttons (button “drag/ drop” is used to do drag and drop of mouse events and another button is used to call task manager (for troubleshooting)) and change behavior of CTRL, ALT, SHIFT, and CAPS LOCK keys in order to provide shortcut keys of keyboard. Also, we develop recovery method which allows users go from camera and then come back again in order to eliminate neck fatigue effect. The experiments which involve several users have been done in our laboratory. The results show that the use of our camera mouse able to allow users do typing, left and right click events, drag and drop events, and troubleshooting without hand. By implement this system, handicap person can use computer more comfortable and reduce the dryness of eyes.
A Proposed Web Accessibility Framework for the Arab DisabledWaqas Tariq
The Web is providing unprecedented access to information and interaction for people with disabilities. This paper presents a Web accessibility framework which offers the ease of the Web accessing for the disabled Arab users and facilitates their lifelong learning as well. The proposed framework system provides the disabled Arab user with an easy means of access using their mother language so they don’t have to overcome the barrier of learning the target-spoken language. This framework is based on analyzing the web page meta-language, extracting its content and reformulating it in a suitable format for the disabled users. The basic objective of this framework is supporting the equal rights of the Arab disabled people for their access to the education and training with non disabled people. Key Words : Arabic Moon code, Arabic Sign Language, Deaf, Deaf-blind, E-learning Interactivity, Moon code, Web accessibility , Web framework , Web System, WWW.
Real Time Blinking Detection Based on Gabor FilterWaqas Tariq
New method of blinking detection is proposed. The utmost important of blinking detections method is robust against different users, noise, and also change of eye shape. In this paper, we propose blinking detections method by measuring of distance between two arcs of eye (upper part and lower part). We detect eye arcs by apply Gabor filter onto eye image. As we know that Gabor filter has advantage on image processing application since it able to extract spatial localized spectral features, such line, arch, and other shape are more easily detected. After two of eye arcs are detected, we measure the distance between both by using connected labeling method. The open eye is marked by the distance between two arcs is more than threshold and otherwise, the closed eye is marked by the distance less than threshold. The experiment result shows that our proposed method robust enough against different users, noise, and eye shape changes with perfectly accuracy.
Computer Input with Human Eyes-Only Using Two Purkinje Images Which Works in ...Waqas Tariq
A method for computer input with human eyes-only using two Purkinje images which works in a real time basis without calibration is proposed. Experimental results shows that cornea curvature can be estimated by using two light sources derived Purkinje images so that no calibration for reducing person-to-person difference of cornea curvature. It is found that the proposed system allows usersf movements of 30 degrees in roll direction and 15 degrees in pitch direction utilizing detected face attitude which is derived from the face plane consisting three feature points on the face, two eyes and nose or mouth. Also it is found that the proposed system does work in a real time basis.
Toward a More Robust Usability concept with Perceived Enjoyment in the contex...Waqas Tariq
Mobile multimedia service is relatively new but has quickly dominated people¡¯s lives, especially among young people. To explain this popularity, this study applies and modifies the Technology Acceptance Model (TAM) to propose a research model and conduct an empirical study. The goal of study is to examine the role of Perceived Enjoyment (PE) and what determinants can contribute to PE in the context of using mobile multimedia service. The result indicates that PE is influencing on Perceived Usefulness (PU) and Perceived Ease of Use (PEOU) and directly Behavior Intention (BI). Aesthetics and flow are key determinants to explain Perceived Enjoyment (PE) in mobile multimedia usage.
Collaborative Learning of Organisational KnolwedgeWaqas Tariq
This paper presents recent research into methods used in Australian Indigenous Knowledge sharing and looks at how these can support the creation of suitable collaborative envi- ronments for timely organisational learning. The protocols and practices as used today and in the past by Indigenous communities are presented and discussed in relation to their relevance to a personalised system of knowledge sharing in modern organisational cultures. This research focuses on user models, knowledge acquisition and integration of data for constructivist learning in a networked repository of or- ganisational knowledge. The data collected in the repository is searched to provide collections of up-to-date and relevant material for training in a work environment. The aim is to improve knowledge collection and sharing in a team envi- ronment. This knowledge can then be collated into a story or workflow that represents the present knowledge in the organisation.
Our research aims to propose a global approach for specification, design and verification of context awareness Human Computer Interface (HCI). This is a Model Based Design approach (MBD). This methodology describes the ubiquitous environment by ontologies. OWL is the standard used for this purpose. The specification and modeling of Human-Computer Interaction are based on Petri nets (PN). This raises the question of representation of Petri nets with XML. We use for this purpose, the standard of modeling PNML. In this paper, we propose an extension of this standard for specification, generation and verification of HCI. This extension is a methodological approach for the construction of PNML with Petri nets. The design principle uses the concept of composition of elementary structures of Petri nets as PNML Modular. The objective is to obtain a valid interface through verification of properties of elementary Petri nets represented with PNML.
Development of Sign Signal Translation System Based on Altera’s FPGA DE2 BoardWaqas Tariq
The main aim of this paper is to build a system that is capable of detecting and recognizing the hand gesture in an image captured by using a camera. The system is built based on Altera’s FPGA DE2 board, which contains a Nios II soft core processor. Image processing techniques and a simple but effective algorithm are implemented to achieve this purpose. Image processing techniques are used to smooth the image in order to ease the subsequent processes in translating the hand sign signal. The algorithm is built for translating the numerical hand sign signal and the result are displayed on the seven segment display. Altera’s Quartus II, SOPC Builder and Nios II EDS software are used to construct the system. By using SOPC Builder, the related components on the DE2 board can be interconnected easily and orderly compared to traditional method that requires lengthy source code and time consuming. Quartus II is used to compile and download the design to the DE2 board. Then, under Nios II EDS, C programming language is used to code the hand sign translation algorithm. Being able to recognize the hand sign signal from images can helps human in controlling a robot and other applications which require only a simple set of instructions provided a CMOS sensor is included in the system.
An overview on Advanced Research Works on Brain-Computer InterfaceWaqas Tariq
A brain–computer interface (BCI) is a proficient result in the research field of human- computer synergy, where direct articulation between brain and an external device occurs resulting in augmenting, assisting and repairing human cognitive. Advanced works like generating brain-computer interface switch technologies for intermittent (or asynchronous) control in natural environments or developing brain-computer interface by Fuzzy logic Systems or by implementing wavelet theory to drive its efficacies are still going on and some useful results has also been found out. The requirements to develop this brain machine interface is also growing day by day i.e. like neuropsychological rehabilitation, emotion control, etc. An overview on the control theory and some advanced works on the field of brain machine interface are shown in this paper.
Exploring the Relationship Between Mobile Phone and Senior Citizens: A Malays...Waqas Tariq
There is growing ageing phenomena with the rise of ageing population throughout the world. According to the World Health Organization (2002), the growing ageing population indicates 694 million, or 223% is expected for people aged 60 and over, since 1970 and 2025.The growth is especially significant in some advanced countries such as North America, Japan, Italy, Germany, United Kingdom and so forth. This growing older adult population has significantly impact the social-culture, lifestyle, healthcare system, economy, infrastructure and government policy of a nation. However, there are limited research studies on the perception and usage of a mobile phone and its service for senior citizens in a developing nation like Malaysia. This paper explores the relationship between mobile phones and senior citizens in Malaysia from the perspective of a developing country. We conducted an exploratory study using contextual interviews with 5 senior citizens of how they perceive their mobile phones. This paper reveals 4 interesting themes from this preliminary study, in addition to the findings of the desirable mobile requirements for local senior citizens with respect of health, safety and communication purposes. The findings of this study bring interesting insight to local telecommunication industries as a whole, and will also serve as groundwork for more in-depth study in the future.
Principles of Good Screen Design in WebsitesWaqas Tariq
Visual techniques for proper arrangement of the elements on the user screen have helped the designers to make the screen look good and attractive. Several visual techniques emphasize the arrangement and ordering of the screen elements based on particular criteria for best appearance of the screen. This paper investigates few significant visual techniques in various web user interfaces and showcases the results for better understanding and their presence.
Virtual teams are used more and more by companies and other organizations to receive benefits. They are a great way to enable teamwork in situations where people are not sitting in the same physical place at the same time. As companies seek to increase the use of virtual teams, a need exists to explore the context of these teams, the virtuality of a team and software that may help these teams working virtualy. Virtual teams have the same basic principles as traditional teams, but there is one big difference. This difference is the way the team members communicate. Instead of using the dynamics of in-office face-to-face exchange, they now rely on special communication channels enabled by modern technologies, such as e-mails, faxes, phone calls and teleconferences, virtual meetings etc. This is why this paper is focused on the issues regarding virtual teams, and how these teams are created and progressing in Albania.
Cognitive Approach Towards the Maintenance of Web-Sites Through Quality Evalu...Waqas Tariq
It is a well established fact that the Web-Applications require frequent maintenance because of cutting– edge business competitions. The authors have worked on quality evaluation of web-site of Indian ecommerce domain. As a result of that work they have made a quality-wise ranking of these sites. According to their work and also the survey done by various other groups Futurebazaar web-site is considered to be one of the best Indian e-shopping sites. In this research paper the authors are assessing the maintenance of the same site by incorporating the problems incurred during this evaluation. This exercise gives a real world maintainability problem of web-sites. This work will give a clear picture of all the quality metrics which are directly or indirectly related with the maintainability of the web-site.
USEFul: A Framework to Mainstream Web Site Usability through Automated Evalua...Waqas Tariq
A paradox has been observed whereby web site usability is proven to be an essential element in a web site, yet at the same time there exist an abundance of web pages with poor usability. This discrepancy is the result of limitations that are currently preventing web developers in the commercial sector from producing usable web sites. In this paper we propose a framework whose objective is to alleviate this problem by automating certain aspects of the usability evaluation process. Mainstreaming comes as a result of automation, therefore enabling a non-expert in the field of usability to conduct the evaluation. This results in reducing the costs associated with such evaluation. Additionally, the framework allows the flexibility of adding, modifying or deleting guidelines without altering the code that references them since the guidelines and the code are two separate components. A comparison of the evaluation results carried out using the framework against published evaluations of web sites carried out by web site usability professionals reveals that the framework is able to automatically identify the majority of usability violations. Due to the consistency with which it evaluates, it identified additional guideline-related violations that were not identified by the human evaluators.
Robot Arm Utilized Having Meal Support System Based on Computer Input by Huma...Waqas Tariq
A robot arm utilized having meal support system based on computer input by human eyes only is proposed. The proposed system is developed for handicap/disabled persons as well as elderly persons and tested with able persons with several shapes and size of eyes under a variety of illumination conditions. The test results with normal persons show the proposed system does work well for selection of the desired foods and for retrieve the foods as appropriate as usersf requirements. It is found that the proposed system is 21% much faster than the manually controlled robotics.
Dynamic Construction of Telugu Speech Corpus for Voice Enabled Text EditorWaqas Tariq
In recent decades speech interactive systems have gained increasing importance. Performance of an ASR system mainly depends on the availability of large corpus of speech. The conventional method of building a large vocabulary speech recognizer for any language uses a top-down approach to speech. This approach requires large speech corpus with sentence or phoneme level transcription of the speech utterances. The transcriptions must also include different speech order so that the recognizer can build models for all the sounds present. But, for Telugu language, because of its complex nature, a very large, well annotated speech database is very difficult to build. It is very difficult, if not impossible, to cover all the words of any Indian language, where each word may have thousands and millions of word forms. A significant part of grammar that is handled by syntax in English (and other similar languages) is handled within morphology in Telugu. Phrases including several words (that is, tokens) in English would be mapped on to a single word in Telugu.Telugu language is phonetic in nature in addition to rich in morphology. That is why the speech technology developed for English cannot be applied to Telugu language. This paper highlights the work carried out in an attempt to build a voice enabled text editor with capability of automatic term suggestion. Main claim of the paper is the recognition enhancement process developed by us for suitability of highly inflecting, rich morphological languages. This method results in increased speech recognition accuracy with very much reduction in corpus size. It also adapts Telugu words to the database dynamically, resulting in growth of the corpus.
An Improved Approach for Word Ambiguity RemovalWaqas Tariq
Word ambiguity removal is a task of removing ambiguity from a word, i.e. correct sense of word is identified from ambiguous sentences. This paper describes a model that uses Part of Speech tagger and three categories for word sense disambiguation (WSD). Human Computer Interaction is very needful to improve interactions between users and computers. For this, the Supervised and Unsupervised methods are combined. The WSD algorithm is used to find the efficient and accurate sense of a word based on domain information. The accuracy of this work is evaluated with the aim of finding best suitable domain of word. Keywords: Human Computer Interaction, Supervised Training, Unsupervised Learning, Word Ambiguity, Word sense disambiguation
Parameters Optimization for Improving ASR Performance in Adverse Real World N...Waqas Tariq
From the existing research it has been observed that many techniques and methodologies are available for performing every step of Automatic Speech Recognition (ASR) system, but the performance (Minimization of Word Error Recognition-WER and Maximization of Word Accuracy Rate- WAR) of the methodology is not dependent on the only technique applied in that method. The research work indicates that, performance mainly depends on the category of the noise, the level of the noise and the variable size of the window, frame, frame overlap etc is considered in the existing methods. The main aim of the work presented in this paper is to use variable size of parameters like window size, frame size and frame overlap percentage to observe the performance of algorithms for various categories of noise with different levels and also train the system for all size of parameters and category of real world noisy environment to improve the performance of the speech recognition system. This paper presents the results of Signal-to-Noise Ratio (SNR) and Accuracy test by applying variable size of parameters. It is observed that, it is really very hard to evaluate test results and decide parameter size for ASR performance improvement for its resultant optimization. Hence, this study further suggests the feasible and optimum parameter size using Fuzzy Inference System (FIS) for enhancing resultant accuracy in adverse real world noisy environmental conditions. This work will be helpful to give discriminative training of ubiquitous ASR system for better Human Computer Interaction (HCI). Keywords: ASR Performance, ASR Parameters Optimization, Multi-Environmental Training, Fuzzy Inference System for ASR, ubiquitous ASR system, Human Computer Interaction (HCI)
June 3, 2024 Anti-Semitism Letter Sent to MIT President Kornbluth and MIT Cor...Levi Shapiro
Letter from the Congress of the United States regarding Anti-Semitism sent June 3rd to MIT President Sally Kornbluth, MIT Corp Chair, Mark Gorenberg
Dear Dr. Kornbluth and Mr. Gorenberg,
The US House of Representatives is deeply concerned by ongoing and pervasive acts of antisemitic
harassment and intimidation at the Massachusetts Institute of Technology (MIT). Failing to act decisively to ensure a safe learning environment for all students would be a grave dereliction of your responsibilities as President of MIT and Chair of the MIT Corporation.
This Congress will not stand idly by and allow an environment hostile to Jewish students to persist. The House believes that your institution is in violation of Title VI of the Civil Rights Act, and the inability or
unwillingness to rectify this violation through action requires accountability.
Postsecondary education is a unique opportunity for students to learn and have their ideas and beliefs challenged. However, universities receiving hundreds of millions of federal funds annually have denied
students that opportunity and have been hijacked to become venues for the promotion of terrorism, antisemitic harassment and intimidation, unlawful encampments, and in some cases, assaults and riots.
The House of Representatives will not countenance the use of federal funds to indoctrinate students into hateful, antisemitic, anti-American supporters of terrorism. Investigations into campus antisemitism by the Committee on Education and the Workforce and the Committee on Ways and Means have been expanded into a Congress-wide probe across all relevant jurisdictions to address this national crisis. The undersigned Committees will conduct oversight into the use of federal funds at MIT and its learning environment under authorities granted to each Committee.
• The Committee on Education and the Workforce has been investigating your institution since December 7, 2023. The Committee has broad jurisdiction over postsecondary education, including its compliance with Title VI of the Civil Rights Act, campus safety concerns over disruptions to the learning environment, and the awarding of federal student aid under the Higher Education Act.
• The Committee on Oversight and Accountability is investigating the sources of funding and other support flowing to groups espousing pro-Hamas propaganda and engaged in antisemitic harassment and intimidation of students. The Committee on Oversight and Accountability is the principal oversight committee of the US House of Representatives and has broad authority to investigate “any matter” at “any time” under House Rule X.
• The Committee on Ways and Means has been investigating several universities since November 15, 2023, when the Committee held a hearing entitled From Ivory Towers to Dark Corners: Investigating the Nexus Between Antisemitism, Tax-Exempt Universities, and Terror Financing. The Committee followed the hearing with letters to those institutions on January 10, 202
2024.06.01 Introducing a competency framework for languag learning materials ...Sandy Millin
http://sandymillin.wordpress.com/iateflwebinar2024
Published classroom materials form the basis of syllabuses, drive teacher professional development, and have a potentially huge influence on learners, teachers and education systems. All teachers also create their own materials, whether a few sentences on a blackboard, a highly-structured fully-realised online course, or anything in between. Despite this, the knowledge and skills needed to create effective language learning materials are rarely part of teacher training, and are mostly learnt by trial and error.
Knowledge and skills frameworks, generally called competency frameworks, for ELT teachers, trainers and managers have existed for a few years now. However, until I created one for my MA dissertation, there wasn’t one drawing together what we need to know and do to be able to effectively produce language learning materials.
This webinar will introduce you to my framework, highlighting the key competencies I identified from my research. It will also show how anybody involved in language teaching (any language, not just English!), teacher training, managing schools or developing language learning materials can benefit from using the framework.
MATATAG CURRICULUM: ASSESSING THE READINESS OF ELEM. PUBLIC SCHOOL TEACHERS I...NelTorrente
In this research, it concludes that while the readiness of teachers in Caloocan City to implement the MATATAG Curriculum is generally positive, targeted efforts in professional development, resource distribution, support networks, and comprehensive preparation can address the existing gaps and ensure successful curriculum implementation.
it describes the bony anatomy including the femoral head , acetabulum, labrum . also discusses the capsule , ligaments . muscle that act on the hip joint and the range of motion are outlined. factors affecting hip joint stability and weight transmission through the joint are summarized.
Read| The latest issue of The Challenger is here! We are thrilled to announce that our school paper has qualified for the NATIONAL SCHOOLS PRESS CONFERENCE (NSPC) 2024. Thank you for your unwavering support and trust. Dive into the stories that made us stand out!
A Strategic Approach: GenAI in EducationPeter Windle
Artificial Intelligence (AI) technologies such as Generative AI, Image Generators and Large Language Models have had a dramatic impact on teaching, learning and assessment over the past 18 months. The most immediate threat AI posed was to Academic Integrity with Higher Education Institutes (HEIs) focusing their efforts on combating the use of GenAI in assessment. Guidelines were developed for staff and students, policies put in place too. Innovative educators have forged paths in the use of Generative AI for teaching, learning and assessments leading to pockets of transformation springing up across HEIs, often with little or no top-down guidance, support or direction.
This Gasta posits a strategic approach to integrating AI into HEIs to prepare staff, students and the curriculum for an evolving world and workplace. We will highlight the advantages of working with these technologies beyond the realm of teaching, learning and assessment by considering prompt engineering skills, industry impact, curriculum changes, and the need for staff upskilling. In contrast, not engaging strategically with Generative AI poses risks, including falling behind peers, missed opportunities and failing to ensure our graduates remain employable. The rapid evolution of AI technologies necessitates a proactive and strategic approach if we are to remain relevant.
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.
A workshop hosted by the South African Journal of Science aimed at postgraduate students and early career researchers with little or no experience in writing and publishing journal articles.
Delivering Micro-Credentials in Technical and Vocational Education and TrainingAG2 Design
Explore how micro-credentials are transforming Technical and Vocational Education and Training (TVET) with this comprehensive slide deck. Discover what micro-credentials are, their importance in TVET, the advantages they offer, and the insights from industry experts. Additionally, learn about the top software applications available for creating and managing micro-credentials. This presentation also includes valuable resources and a discussion on the future of these specialised certifications.
For more detailed information on delivering micro-credentials in TVET, visit this https://tvettrainer.com/delivering-micro-credentials-in-tvet/
Design of an Adaptive Hearing Aid Algorithm using Booth-Wallace Tree Multiplier
1. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 1
Design of an Adaptive Hearing Aid Algorithm using Booth-
Wallace Tree Multiplier
Jitendra Kumar Das jkdas12@gmail.com
Asst. Professor, Dept. of Electronics and Telecommunication
Engineering, Synergy Institute of Engineering & Technology,
Dhenkanal, 759001, Orissa, India
Dr. K. K. Mahapatra kmaha2@rediffmail.com
Professor, Dept. of Electronics and Communication,
NIT Rourkela, 769008, Orissa, India
Abstract
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.
Keywords: Booth Multiplier, Booth Wallace Multiplier, Adaptive Lattice Filte
1. INTRODUCTION
The decimation filter used in designing of a hearing aid has two major disadvantages which
are
a. It requires more area for designing in FPGA.
b. As it is highly serial in nature, it requires more output latency.
Due to above reasons it consumes more power and we are specifically interested on lowering
the power consumption of digital hearing aids. In this context we have used our own
customized multiplier while maintaining the overall signal quality [6-7] to lower the power
consumption.
Hearing impairment is often accompanied with reduced frequency selectivity which leads to a
decreased speech intelligibility in noisy environments [2-5]. One possibility to alleviate this
deficiency is the spectral sharpening for speech enhancement based on adaptive filtering [8-
9] by which the intelligibility of the speech signal is maintained. Due to area constraints, such
algorithms are usually implemented in totally time-multiplexed architectures, in which multiple
operations are scheduled to run on a few processing units. This work discusses the power
2. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 2
consumption in an FPGA implementation of the speech enhancement algorithm. It points out
that power consumption can be reduced using Booth Wallace multiplier [11]. Several
implementations of the algorithm, differing only in the degree of resource sharing are
investigated aiming at power-efficiency maximization. At first an overview of the algorithm is
given. Next the realized architectures using booth-Wallace tree multiplier are presented.
2. WALLACE TREE MULTIPLIER
Wallace trees are irregular in the sense that the informal description does not specify a
systematic method for the compressor interconnections [10]. However, it is an efficient
implementation of adding partial products in parallel [12]. The Wallace tree operates in three
steps:
Multiply: The multiplication is carried out using Booth Algorithm [11-14] which will
generate n/2 partial product where n is number of bits of the multiplicand. The partial
products are generated using the Booth recoding table given in table 1.
Mri+1 Mr Mri-1 Recoded output
0 0 0 0
0 0 1 Y
0 1 0 Y
0 1 1 +2Y
1 0 0 -2Y
1 0 1 -Y
1 1 0 -Y
1 1 1 0
TABLE 1: Booth recoding table
Addition: As long as there are more than 3 wires with the same weights add a
following layer. Take 3 wires of same weight and input them into a full adder. The
result will be an output wire of same weight. If there are two wires of same weight,
add them using half-adder and if only one is left, connect it to the next layer.
Group the wires in two numbers and add in a conventional adder. A typical Wallace
tree architecture is shown in figure1 below. In the diagram AB0-AB7 represents the
partial products.
FIGURE 1: Wallace tree multiplier
Wallace multipliers consist of AND-gates, Carry Save Adders and a Carry Propagate Adder or
Carry Look-ahead Adder.
The n-bit CSA consists of disjoint full adders (FA’s). It consumes three-bit input vectors and
produces two outputs, i.e., n-bit sum vector S and n-bit carry vector C. Unlike the normal
adders [e.g. ripple-carry adder (RCA) and carry-look ahead adder (CLA)], a CSA contains no
3. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 3
carry propagation. Consequently, the CSA has the same propagation delay as only one FA
delay and the delay is constant for any value of n. For sufficiently large n, the CSA
implementation becomes much faster and also relatively smaller in size than the
implementation of normal adders. In Wallace multiplier carry save adders are used, and one
carry propagate adder is used as shown in the figure 2. The basic idea in Wallace multiplier is
that all the partial products are added at the same time instead of adding one at a time. This
speeds up the multiplication process.
FIGURE 2: Implementation of n bit CSA operation
3. ADAPTIVE LATTICE FILTER
The adaptive lattice filter [6-7, 18-20] consists of three parts and these are
a. Adaptive decorrelator
b. Analysis filter (lattice filter)
c. Synthesis filter(lattice structure)
The Adaptive Decorrelator
An adaptive filter is a filter that adjusts its transfer function according to an optimizing
algorithm. Because of the complexity of the optimizing algorithms, most adaptive filters are
digital filters that perform digital signal processing and adapt their performance based on the
input signal used.
The adaptive process involves the use of a cost function, which is a criterion for optimum
performance so that the filter coefficients adjusted to minimize the cost on the next iteration
[1]. The block diagram for such filter is presented in figure 3 that serves as a foundation for
particular adaptive filter realizations, such as Least Mean Squares (LMS), Recursive Least
Squares (RLS) or steepest descent algorithm etc. The idea behind the block diagram is that a
variable filter extracts an estimate of the desired signal.
4. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 4
FIGURE 3: Block diagram of an Adaptive filter
The structure presented in figure 3 is described now. We take the following assumptions:
1. The input signal is the sum of a desired signal d(n) and interfering noise v(n)
x(n) = d(n) + v(n) (1)
2. The variable filter has a Finite Impulse Response (FIR) structure. For such structures the
impulse response is equal to the filter coefficients. The coefficients for a filter of order p are
defined as
Wn(0)=[wn(0),wn(1),…wn(p)]
T
(2)
3. The error signal or cost function is the difference between the desired and the estimated
signal.
(n)dd(n)e(n)
^
−= (4)
The variable filter estimates the desired signal by convolving the input signal with the impulse
response. In vector notation this is expressed as
(n)X(n)Wd(n) T
= (5)
where
x(n)=[x(n), x(n-1),………..,x(n-p)]T
(6)
is an input signal vector. Moreover, the variable filter updates the filter coefficients at every
time instant
Wn+1=Wn+ ∆Wn (7)
where ∆wn is a correction factor for the filter coefficients. The adaptive algorithm generates
this correction factor based on the input and error signals. LMS and RLS define two different
coefficient update algorithms. The speech signal to be transmitted is spectrally masked by
noise. By using an adaptive filter, we can attempt to minimize the error by finding the
correlation between the noise at the signal microphone and the (correlated) noise at the
reference microphone. In this particular case the error does not tend to zero as we note the
signal d(k) = x(k) + n(k) whereas the input signal to the filter is x(k) and n(k) does not contain
any speech [2]. Therefore it is not possible to "subtract" any speech when forming e(k)=d(k)-
d(n) . Hence in minimising the power of the error signal e(k) we note that only the noise is
removed and e(k) =~ x(k).
Figure 4 depicts the structure of the adaptive gradient lattice decorrelator. The illustration
shows three stages only with indices 1, i and m. good results typically require a filter order m
where m can vary from 8 to 10 for speech sampled at 8 kHz. The output signal with vanishing
autocorrelation is computed on the upper signal path by subtracting from the input sample
suitable fractions of the signal values on the lower path. The multipliers K1,K2………..Km are
iteratively computed as in equation 8.
Ki[n]:=Ki[n-1]+∆Ki[n] (8)
5. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 5
At every sampling interval n. the details of this process are illustrated in figure 4 for the i-th
stage. Input and output values on upper and lower signal path to and from the i
th
stage
contribute to the computation of the update value ∆Ki.
Both output values are multiplied with the input values on the opposite path and then summed
up to form the numerator in the subsequent computation of the update value. The
denominator σ
2
is iteratively computed.
σ
2
:=e. σ
2
[n-1]+ ∆ σ
2
[n] (9)
The incremental value equals the sum of the two squared input values. The iterative
computation of the denominator defines an exponentially decaying window which
progressively decreases the influence of past contributions.
x
+
x
z-1
XXXX
+
+
+
z-1
z-1
x
+/
z-1
K
K
z-1 +
FIGURE 4: Adaptive gradient lattice filter
The computationally expensive division yields fast converging filter coefficients ki independent
of the varying input signal power level. This remarkable property is indispensable for good
enhancement results. It is also clear contrast to simpler algorithms replacing the division by a
multiplication with a small convergence constant 0<µ<<1. The longest delay through a string
of lattice filters extends from the output of the storage element in the first lattice filter, through
a multiplication with the first reflection coefficient, and then through an addition for each stage
of the lattice filter until the output is produced in the final stage. For a large number of lattice
filter stages, this longest delay can be reduced by a lattice filter optimization for speed which
defers the final carry propagating addition until after the final lattice filter stage. This requires
the transmission of an additional value between lattice filter stages. The multiplication process
is speeded up using booth multiplier and the accumulation process is done faster using the
Wallace multiplier.
The Analysis Filter
The analysis filter H(z)=[1-A(z/β)] is illustrated in figure 5. Its structure [1, 8-9] is similar to that
of the adaptive decorrelator shown in figure 4. The only difference is the multiplication with the
filter parameter β following every shift element z
-1
on the lower signal path. Furthermore, the
analysis filter does not need a separate circuitry for coefficient update. It instead requires and
6. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 6
therefore copies the filter coefficients K1, K2…..Km computed by the unmodified (β =1) filter
structure, i.e., the adaptive decorrelator.
FIGURE 5: Analysis filter (1-A(z/), x(n) is input and y’(n) is output
FIGURE 6: Single stage of analysis filter
The two mathematical equations for the single stage analysis filter as shown in figure 6 are
fm(n) = fm-1(n)-kmbm-1(n-1) (10)
bm(n) = bm-1(n-1)-kmfm-1(n) (11)
Some characteristics of the Lattice predictor are
It is the most efficient structure for generating simultaneously the forward and
backward prediction errors.
The lattice structure is modular: increasing the order of the filter requires adding
only one extra module, leaving all other modules the same.
The various stages of a lattice are decoupled from each other in the following
sense: The memory of the lattice (storing b0(n ¡ 1); : : : ; bm¡1(n ¡ 1)) contains
orthogonal variables, thus the information contained in x(n) is splitted in m pieces,
which reduces gradually the redundancy of the signal.
The similar structure of the lattice filter stages makes the filter suitable for VLSI
implementation.
Lattice filters typically find use in such applications as predictive filtering, adaptive filtering,
and speech processing. One desirable feature of lattice filters are their use of reflection
coefficients as the filter parameter. Algorithms exist to compute reflection coefficients to
obtain the optimal linear filter for a given filter order. Reflection coefficients have the additional
property that for some applications, the optimal reflection coefficients remain unchanged
when going from a lower order filter to a higher order filter. Thus, when adding additional filter
stages, only the reflection coefficients for the added stages need to be computed. The
7. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 7
modification with filter parameter causes the analysis filter to produce an output signal with
reduced formants instead of a signal with completely flat spectral envelope as produced by
the adaptive decorrelator.
The Synthesis Filter
When considering IIR filters, the direct form filter is the common structure of choice. This is
true, in general, because when designing an algorithm which adapts the parameters ak and bk,
the coefficients of the difference equation, described below, are manipulated directly.
Yk+a1Yk-1+- - - - - - - - - - - - +amYk-m=b0Uk+b1Uk-1+- - - - - - +bmUn-m (12)
Some problems exist in using the direct form filter for adaptive applications. First of all,
ensuring stability of a time-varying direct form filter can be a major difficulty. It is often
computationally a burden because the polynomial, A(z), made up of the ak parameters, must
be checked to see if it is minimum phase at each iteration. Even if the stability was assured
during adaptation, round off error causing limit cycles can plague the filter. Parallel and
cascade forms are often used as alternatives for direct form filters. These consist of an
interconnection of first and second order filter sections, whose sensitivity to round off errors
tends to be less drastic than for the direct form filter. Since the filter is broken down into a
factored form, the round off error associated with each factorization only affects that term. In
the direct form filter, the factors are lumped together so that round off error in each term
affects all of the factors in turn.
A larger problem exists for both parallel and cascade forms: the mapping from transfer
function space to parameter space is not unique. Whenever the mapping from the transfer
function space to the parameter space is not unique, additional saddle points in the error
surface appear that would not be present if the mapping had been unique. The addition of
these saddle points can slow down the convergence speed if the parameter trajectories
wander close to these saddle points. For this reason, these filter forms are considered
unsuitable for adaptive filtering.
A tapped-state lattice form has many of the desirable properties associated with common
digital filters and avoids the problems discussed above. Due to the computational structure,
the round off error in this filter is inherently low.
Direct implementation of the IIR filter can lead to instabilities if it is quantized. The filter is
stable using the following structure [1, 7-9, 14-16]. The structure of the synthesis filter
H(z)=[1-A(z/γ)]
-1
is shown in figure 7. The synthesis filter also requires and copies the filter
coefficients K1, K2,……Km from the adaptive decorrelator at every sampling interval. The
structure in figure 4.7 also shows a synthesis filter modified by the multiplication with the filter
parameter γ succeeding every shift element z
-1
on the lower signal path. The unmodified
synthesis filter (γ=1) restores the original formants in the output when a signal with flat
spectral envelope is fed to its input. The modification with a parameter value less than unity
causes the synthesis filter to produce an output signal with partially restored formants only.
The spectral sharpening effect results from a suitable choice of both filter parameters
0<β<γ<1. Experiments with one adaptive filter only failed in producing satisfactory speech
enhancement results.
8. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 8
FIGURE 7: Synthesis filter, y’(n) input and y(n) is output
FIGURE 8: Single stage of sysnthesi filter.
The two mathematical equations for the single stage synthesis filter are shown below.
fm(n) = fm-1(n)-kmbm-1(n-1) (13)
gm(n) = gm-1(n-1)-kmfm-1(n) (14)
The computational complexity of a digital filter structure is given by the total number of
multipliers and the total number of two input adders required for its implementation which
roughly provides an indication of its cost of implementation. The synthesis filter is stable if the
magnitudes of all multiplier coefficients in the realization are less than unity i.e. - 1<Km<1 for
m=M, M-1, M-2, …1, 0.
4. HEARING AID DESIGN
4.1 Spectral Sharpening For Speech Enhancement
Speech enhancement usually results from adaptively filtering the noise reference signals and
subsequently subtracting them from the primary input. However, a procedure for speech
enhancement based on a single audio path is presented here. It is therefore applicable for
real world situations. An example of such a situation is using hearing aid equipment. The
hearing impaired person could place additional microphones close to noise sources only
rarely. Current hearing aid equipment are used for filtering and amplifying the speech signal,
this suggests that hearing impairment is just a more or less reduced sensitivity to sound
pressure in various frequency intervals. This view however neglects the loss of frequency
discrimination which can be efficiently compensated by the spectral sharpening technique
presented. The idea of spectral sharpening originates from the adaptive post filtering method
in modern speech coding schemes at bit rates around 8 kb/s and lower [1]. With these
algorithms speech is encoded segment by segment. The linear prediction filter is any way
computed in every speech segment for the encoding process as
A(z)=a1z-1
+ a2z-2
+…..+ amz-m
(15)
and post- filtering with the transfer function
9. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 9
)/(1
)/(1
)(
γ
β
zA
zA
zH
−
−
= (16)
and constant filter parameters 0<β<γ<1 is subsequently performed with a moderate
computational increase.
Figure 9 shows the block diagram of spectral sharpening of speech sharpening [8-9] for
speech enhancement. The speech signal x[n] from the microphone splits into three distinct
paths. The signal on the lowest path passes through the analysis filter [1-A(z/β)] and
subsequently through the synthesis filter [1-A(z/γ)]-1
. Both filters are implemented as lattice
filters with the analysis and synthesis structures respectively. They both require the identical
set of reflection coefficients K1, K2,…..Km. where m represents the number of stages which is
updated in every sampling interval by the adaptive decorrelator shown on the middle path of
figure 4. The filter parameters β and γ do not vary with time.
FIGURE 9: Block diagram of Spectral Sharpening for Speech Enhancement.
A high pass filter 1-αz
-1
is shown in front of the adaptive decorrelator, where x=1 may be
chosen for simplicity. The high pass filter is used in order to compensate the spectral tilt of
natural speech: the average power of the speech signal decreases above 1 KHz at a rate of ~
10 db per octave. The adaptive transfer function in equation (16) enhances this spectral tilt
even more when the filter coefficients K1,K2,……..Km are computed from the speech signal
x[n] directly. Efficient speech enhancement requires however that the various formants are
more or less uniformly emphasized, regardless of their relative power level. This is possible
with the use of the high pass filter. It compensates at least partially the original spectral tilt.
The decorrelator on the middle signal path of the figure is an adaptive gradient lattice filter. It
produces an output signal with vanishing autocorrelation by updating its filter coefficients in
every sampling interval to the continuously changing input signal characteristics. The output
signal is not required in this application, however. The updated filter coefficients K1, K2……Km
are of interest only for the use in the analysis and synthesis filter.
4.2 Spectral Sharpening For Noise Reduction
The block diagram of the spectral sharpening process for noise reduction is illustrated in
figure 10. The arrangement of adaptive decorrelator, analysis and synthesis filters agrees with
the previous block diagram in figure 9, however there various differences like
1. no loudness control,.
2. the input signal x[n] goes directly to the adaptive decorrelator, and
10. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 10
3. a high pass filter precedes the analysis and synthesis filters.
1
1
1
)1(
)( −
−
−
−
=
az
zb
zHhp
(17)
The reasons for these differences are as follows.
As mentioned in the previous section the spectral sharpening process
)/(1
)/(1
)(
γ
β
za
za
zH
−
−
= (18)
introduces a signal dependent amplification, signal segments with strong formant structure
are amplified more than segments with a rather flat spectral envelop. In the sequel it is
assumed that back ground noise is the major source for signal degradation and that its
spectrum reveals relatively flat resonances only. Speech segments with strong resonances
clearly profit in this situation. They experience a remarkable amplification compared to noisy
segments. The loudness compensation of the previous block diagram is consequently omitted
in order to preserve this effect.
FIGURE 10: Block diagram of Spectral Sharpening by Noise Reduction
Best results require that the input signal is directly fed to the adaptive decorrelator. Only
negligible amplification is then applied to noisy signal segments as a consequence of their
assumed approximately flat spectrum. The spectral sharpening process further enhances the
spectral tilt of speech when the filter parameters are estimated from the speech signal without
prior compensation.
The high pass filter which preceded the adaptive decorrelator in the figure 9 has been shifted
to the bottom signal path in figure 10 in order to avoid the scheme from producing a dull
sound.
4.3 High Pass Filter
In signal processing, there are many instances in which an input signal to a system contains
extra unnecessary content or additional noise which can degrade the quality of the desired
signal. In such cases we may remove or filter out the useless samples. For example, in the
case of the telephone system, there is no reason to transmit very high frequencies since most
speech falls within the band of 700 to 3,400 Hz. Therefore, in this case, all frequencies above
and below that band are filtered out. The frequency band between 700 and 3,400 Hz, which
isn’t filtered out, is known as the pass band, and the frequency band that is blocked out is
known as the stop band. FIR, Finite Impulse Response, filters are one of the primary types of
filters used in Digital Signal Processing [1, 14-16]. FIR filters are said to be finite because they
do not have any feedback. Therefore, if an impulse is sent through the system (a single spike)
then the output would invariably become zero as soon as the impulse runs through the filter.
11. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 11
There are a few terms that are used to describe the behaviour and performance of FIR filter.
These are
• Filter Coefficients - The set of constants, also called tap weights, used to multiply
against delayed sample values. For an FIR filter, the filter coefficients are, by
definition, the impulse response of the filter.
• Impulse Response – A filter’s time domain output sequence when the input is an
impulse. An impulse is a single unity-valued sample followed and preceded by zero
valued samples. For an FIR filter the impulse response of a FIR filter is the set of filter
coefficients.
• Tap – The number of FIR taps, typically N, tells us a couple things about the filter.
Most importantly it tells us the amount of memory needed, the number of calculations
required, and the amount of "filtering" that it can do. Basically, the more taps in a filter
results in better stop band attenuation (less of the part we want filtered out), less
ripple (less variations in the pass band), and steeper roll off (a shorter transition
between the pass band and the stop band).
• Multiply-Accumulate (MAC) – In the context of FIR Filters, a "MAC" is the operation of
multiplying a coefficient by the corresponding delayed data sample and accumulating
the result. There is usually one MAC per tap.
FIGURE 11: General causal FIR filter structure
Figure 11 gives the signal flow graph for a general finite-impulse-response filter (FIR). Such a
filter is also called a transversal filter, or a tapped delay line. The implementation is one
example of a direct-form implementation of a digital filter. The impulse response h(n) is
obtained at the output when the input signal is the impulse signal =[1 0 0 0…]. If the kth tap
is denoted bk, then it is obvious from figure 11 above that the impulse response signal is given
by
>
≤≤
<
=
Mn
Mnb
n
nh n
,0
0,
0,0
)( (19)
In other words, the impulse response simply consists of the tap coefficients, pretended and
appended by zeros.
12. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 12
5. EXPERIMENTAL RESULTS AND CONCLUSION
5.1 Spectral Sharpening For Speech Enhancement In Matlab
Validation of the proposed filter was conducted using simulation tool. MATLAB v7.01 was
used as platform. Speech signal constituting of 26000 samples at 8k samples/ sec was
generated using wave recorder of windows and captured as a Matlab data file. This
constitutes 3.25s of voice data. The waveform generated is presented in figure 12. The peak
value of the signal generated is 275 mV. This signal formed the input to the hearing aid
appliance. The output of a single stage is presented in figure 13 for β=0.04, γ=0.6 and µ=0.98.
From the observation of the filter output it is seen that the output amplitude is nearly 600 mV.
The single stage output with the filter parameters, β=0.4, γ=0.6 and µ=0.98 is presented in
figure 14. In this case, peak amplitude is 390 mV which constitute a gain less than 2.
FIGURE 12: Waveform of the 3.25s speech input
FIGURE 12: Waveform of the 3.25 second hearing aid output using parameters β=0.04,γ=0.6,µ=0.98
FIGURE 13: Waveform of the 3.25 second hearing aid output using parameters =0.4,=0.6,=0.98
13. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 13
Following this; the performance of a 8 stage filter is observed. The filter output for β=0.04,
γ=0.6, µ=0.98 and β=0.4, γ=0.6 and µ=0.98 are presented in figure 15 and figure 16
respectively. From figure 12 and figure 15, it is seen that output is more than double.
Considering the superior performance of the 8 stage filter output over single stage filter, a 8
stage is used for hardware implementation.
FIGURE 15: Waveform of the 3.25 second hearing aid output using parameters
FIGURE 16: Waveform of the 3.25 second hearing aid output using parameters =0.4,=0.6,=0.98
5.2 FPGA Based Simulation Results.
MULTIPLIERS
The table below compares the cell usage of the three multipliers (SHIFT/ADD, BOOTH’S and
BOOTH WALLACE multiplier) for 8 bit by 8 bit multiplication and 16 bit by 16 bit multiplication.
From the table we can see that the booth Wallace multiplier uses less hardware compared to
that of the shift/add multiplier and booth multiplier. The details are given table 2.
Cell
Usage
Shift/add
multiplier
(8x8)
Shift/add
multiplier
(16x16)
Booth
multiplier
(8x8)
Booth
multiplier
(16x16)
Booth
Wallace
multiplier
(8x8)
Booth
Wallace
multiplier
(16x16)
BELS 240 1000 333 975 167 697
LUT-1 1 1 0 0 0 0
LUT-2 14 1 37 36 5 9
LUT-2 34 186 28 66 51 234
LUT-4 74 290 116 399 83 328
MUXCY 56 240 64 228 0 0
MUXF5 11 27 14 2 28 126
XORCY 49 225 61 219 0 0
14. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 14
TABLE 2: Cell used for the three Multipliers in virtex2p
Cells used Slices 4-LUT IO Delay Power consumption
Booth multiplier 109 192 32 24.41 ns 5 mW
Booth Wallace
multiplier
76 139 32 20.62 ns 3 mW
TABLE 3: Power consumption and Delay for two multipliers with 8x8 bits
From the table 3 we can see that the using the booth Wallace multiplier consumes less power
compared to the booth multiplier and also that booth Wallace multiplier is faster than booth
multiplier. Hence the Booth Wallace multiplier [17-20] is used for hearing aid design in VHDL
in this investigation. The table 4 gives area used by the two multipliers. From this it can be
seen that the booth Wallace tree multiplier uses less hardware than other.
Cells used Slices Slice
flip flops
4-LUT Logic Shift
registers
IO
Booth multiplier 2684 183 5003 4979 24 32
Booth Wallace Multiplier 2583 196 4885 4866 19 32
TABLE 4: Cell usage for hearing aid component in virtex2p
5.3 Spectral Sharpening for Speech Enhancement in VHDL
The amplitude values of the speech signal sampled at 8 kS/s is rounded to 8 bits and stored
in a text file for VHDL simulation. The hearing aid is designed in VHDL and is tested using
different multipliers. The first 250 samples are taken as input for the hearing aid in VHDL .The
output obtained through simulation is stored in a text file. The text file is read in MATLAB and
is plotted as shown in the figure 18. The parameters used in VHDL are β=0.04, γ=0.6, µ=0.98.
FIGURE 14: Comparision of input speech signal with output using vhdl for 250 samples
15. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 15
FIGURE 15: Comparison of the MATLAB output speech signal with the output obtained using VHDL for
250 samples
From the figure we can see that the output obtained using VHDL is slightly less in magnitude
than MATLAB output. This is due to rounding of the values and due to fixed point
multiplication. But from the figure 19 we see that the VHDL output follows the MATLAB output.
The table 5 below shows the resource utilization summary and power consumed by the two
designs using different multipliers.
Design name Slice
used
out of
3008
4-LUTs
used out
of 6016
Slice FFs
used out
of 6016
Shift
registers
Logics IO used
out of 348
Power
consumption
High pass filter using
Booth multiplier
241
(7%)
446
(7.4%)
41
(<1%)
- - 25
(7%)
-
Synthesis filter 175
(5.8%)
323
(5.3%)
32
(<1%)
- - 41
(11%)
-
Decorrelator 164
(5.4%)
292
(4.9%)
104
(1.7%)
- - 40
(11%)
-
Hearing aid using
Booth multiplier
2684
(89%)
5003
(83%)
183
(3%)
24 4979 32
(9.5%)
40 mW
Hearing aid using
Booth Wallace
multiplier
2583
(85%)
4885
(81%)
196
(3.3%)
19 4866 33
(9.5%)
30 mW
TABLE 5: FPGA resources used and power of hearing aid design
6. Conclusions
All the papers referred so far do not convey any information about the power consumption
and only architectural part is discussed. Our emphasis is to design an adaptive algorithm
based on Booth-Wallace tree multiplier which consumes less power with respect to the use of
Booth multiplier suitable for hearing aid application.. with this effort we the whole system in
figure 10 is implemented using Booth-Wallace tree multiplier and power calculation of the
whole system is done using Xilinx XPower Analyser. From the figure 19 it can be seen that
our design output matches the Matlab output. Also referring to table 5 we can see that the
power consumed by the hearing aid with booth Wallace tree multiplier is less than the hearing
aid using booth multiplier which is about 25% lesser than the latter. So we can conclude that
the hearing aid using booth Wallace tree multiplier consumes less power in this case.
16. Jitendra Kumar Das & Dr. K. K. Mahapatra
International Journal of Logic and Computation (IJLP), Volume (1): Issue (1) 16
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