The document describes an adaptive blind multiuser detection method for asynchronous code division multiple access (CDMA) systems using principal component analysis (PCA) in impulsive noise environments. PCA is used to extract the principal components from the received signal without requiring training sequences or prior knowledge of channel characteristics. The PCA blind multiuser detector provides robust performance compared to knowledge-based detectors when signature waveforms and timing offsets of users are unknown. Simulation results show the proposed PCA method offers substantial gains over traditional subspace methods for multiuser detection.
Further results on the joint time delay and frequency estimation without eige...IJCNCJournal
Joint Time Delay and Frequency Estimation (JTDFE) problem of complex sinusoidal signals received at
two separated sensors is an attractive problem that has been considered for several engineering
applications. In this paper, a high resolution null (noise) subspace method without eigenvalue
decomposition is proposed. The direct data Matrix is replaced by an upper triangular matrix obtained from
Rank-Revealing LU (RRLU) factorization. The RRLU provides accurate information about the rank and the
numerical null space which make it a valuable tool in numerical linear algebra.The proposed novel method
decreases the computational complexity of JTDFE approximately to the half compared with RRQR
methods. The proposed method generates estimates of the unknown parameters which are based on the
observation and/or covariance matrices. This leads to a significant improvement in the computational load.
Computer simulations are included in this paper to demonstrate the proposed method.
This paper proposes modeling and identification of dynamical systems in delta
domain using neural network. The properties of delta operator are used such as greater
numerical robustness in computation and superior coefficients representation in finite word
length in implementation and well ensured numerical conditioning at high sampling
frequency. To formulate the identification scheme delta operator model is recasted into a
realizable neural network structure using the properties of inverse delta operator.
Adaptive Channel Equalization using Multilayer Perceptron Neural Networks wit...IOSRJVSP
This research addresses the problem inter-symbol interference (ISI) using equalization techniques for time dispersive channels with additive white Gaussian noise (AWGN). The channel equalizer is modelled as a non-linear Multilayer Perceptron (MLP) structure. The Back Propagation (BP) algorithm is used to optimize the synaptic weights of the equalizer during the training mode. In the typical BP algorithm, the error signal is propagated from the output layer to the input layer while the learning rate parameter is held constant. In this study, the BP algorithm is modified so as to allow for the learning rate to be variable at each iteration and this achieves a faster convergence. The proposed algorithm is used to train the MLP based decision feedback equalizer (DFE) for time dispersive ISI channels. The equalizer is tested for a random input sequence of BPSK signals and its performance analysed in terms of the Bit Error Rates and speed of convergence. Simulation results show that the proposed algorithm improves the Bit Error Rate (BER) and rate of convergence.
A Novel Algorithm to Estimate Closely Spaced Source DOA IJECEIAES
In order to improve resolution and direction of arrival (DOA) estimation of two closely spaced sources, in context of array processing, a new algorithm is presented. However, the proposed algorithm combines both spatial sampling technic to widen the resolution and a high resolution method which is the Multiple Signal Classification (MUSIC) to estimate the DOA of two closely spaced sources impinging on the far-field of Uniform Linear Array (ULA). Simulations examples are discussed to demonstrate the performance and the effectiveness of the proposed approach (referred as Spatial sampling MUSIC SS-MUSIC) compared to the classical MUSIC method when it’s used alone in this context.
Further results on the joint time delay and frequency estimation without eige...IJCNCJournal
Joint Time Delay and Frequency Estimation (JTDFE) problem of complex sinusoidal signals received at
two separated sensors is an attractive problem that has been considered for several engineering
applications. In this paper, a high resolution null (noise) subspace method without eigenvalue
decomposition is proposed. The direct data Matrix is replaced by an upper triangular matrix obtained from
Rank-Revealing LU (RRLU) factorization. The RRLU provides accurate information about the rank and the
numerical null space which make it a valuable tool in numerical linear algebra.The proposed novel method
decreases the computational complexity of JTDFE approximately to the half compared with RRQR
methods. The proposed method generates estimates of the unknown parameters which are based on the
observation and/or covariance matrices. This leads to a significant improvement in the computational load.
Computer simulations are included in this paper to demonstrate the proposed method.
This paper proposes modeling and identification of dynamical systems in delta
domain using neural network. The properties of delta operator are used such as greater
numerical robustness in computation and superior coefficients representation in finite word
length in implementation and well ensured numerical conditioning at high sampling
frequency. To formulate the identification scheme delta operator model is recasted into a
realizable neural network structure using the properties of inverse delta operator.
Adaptive Channel Equalization using Multilayer Perceptron Neural Networks wit...IOSRJVSP
This research addresses the problem inter-symbol interference (ISI) using equalization techniques for time dispersive channels with additive white Gaussian noise (AWGN). The channel equalizer is modelled as a non-linear Multilayer Perceptron (MLP) structure. The Back Propagation (BP) algorithm is used to optimize the synaptic weights of the equalizer during the training mode. In the typical BP algorithm, the error signal is propagated from the output layer to the input layer while the learning rate parameter is held constant. In this study, the BP algorithm is modified so as to allow for the learning rate to be variable at each iteration and this achieves a faster convergence. The proposed algorithm is used to train the MLP based decision feedback equalizer (DFE) for time dispersive ISI channels. The equalizer is tested for a random input sequence of BPSK signals and its performance analysed in terms of the Bit Error Rates and speed of convergence. Simulation results show that the proposed algorithm improves the Bit Error Rate (BER) and rate of convergence.
A Novel Algorithm to Estimate Closely Spaced Source DOA IJECEIAES
In order to improve resolution and direction of arrival (DOA) estimation of two closely spaced sources, in context of array processing, a new algorithm is presented. However, the proposed algorithm combines both spatial sampling technic to widen the resolution and a high resolution method which is the Multiple Signal Classification (MUSIC) to estimate the DOA of two closely spaced sources impinging on the far-field of Uniform Linear Array (ULA). Simulations examples are discussed to demonstrate the performance and the effectiveness of the proposed approach (referred as Spatial sampling MUSIC SS-MUSIC) compared to the classical MUSIC method when it’s used alone in this context.
In order to improve sensing performance when the noise variance is not known, this paper considers a so-called
blind spectrum sensing technique that is based on eigenvalue models. In this paper, we employed the spiked population
models in order to identify the miss detection probability. At first, we try to estimate the unknown noise variance
based on the blind measurements at a secondary location. We then investigate the performance of detection, in terms
of both theoretical and empirical aspects, after applying this estimated noise variance result. In addition, we study the
effects of the number of SUs and the number of samples on the spectrum sensing performance.
In this paper, a new algorithm for a high resolution
Direction Of Arrival (DOA) estimation method for multiple
wideband signals is proposed. The proposed method proceeds
in two steps. In the first step, the received signals data is
decomposed in a Toeplitz form using the first-order statistics.
In the second step, The QR decomposition is applied on the
constructed Toeplitz matrix. Compared with existing schemes,
the proposed scheme provides several advantages. First, it
requires computing the triangular matrix R or the orthogonal
matrix Q to find the DOA; these matrices can be computed
with O(n2) operation. However, most of the existing schemes
required eignvalue decomposition (EVD) for the covariance
matrix or singular value decomposition (SVD) for the data
matrix; using EVD or SVD requires much more complex
computational O(n3) operation. Second, the proposed scheme
is more suitable for high-speed communication since it
requires first-order statistics and a single snapshot. Third,
the proposed scheme can estimate the correlated wideband
signals without using spatial smoothing techniques; whereas,
already-existing schemes do not. Accuracy of the proposed
wideband DOA estimation method is evaluated through
computer simulation in comparison with a conventional
method.
Methodological study of opinion mining and sentiment analysis techniquesijsc
Decision making both on individual and organizational level is always accompanied by the search of
other’s opinion on the same. With tremendous establishment of opinion rich resources like, reviews, forum
discussions, blogs, micro-blogs, Twitter etc provide a rich anthology of sentiments. This user generated
content can serve as a benefaction to market if the semantic orientations are deliberated. Opinion mining
and sentiment analysis are the formalization for studying and construing opinions and sentiments. The
digital ecosystem has itself paved way for use of huge volume of opinionated data recorded. This paper is
an attempt to review and evaluate the various techniques used for opinion and sentiment analysis.
Noisy Speech Enhancement Using Soft Thresholding on Selected Intrinsic Mode F...CSCJournals
In this paper, a new speech enhancement method is introduced. It is essentially based on the Empirical Mode Decomposition technique (EMD) and a soft thresholding approach applied on selected modes. The proposed method is a fully data driven approach. First the noisy speech signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs) by using a time decomposition called sifting process. Second, selected IMFs are soft thresholded and added to the remaining IMFs with the residue to reconstitute the enhanced speech signal. The proposed approach is evaluated using speech signals from NOISEUS database corrupted with additive white Gaussian noise. Our algorithm is compared to other state of the art algorithms.
Performance of Matching Algorithmsfor Signal Approximationiosrjce
IOSR Journal of Electronics and Communication Engineering(IOSR-JECE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of electronics and communication engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in electronics and communication engineering. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Reduction of PAPR and Efficient detection ordering scheme for MIMO Transmissi...IJERA Editor
The technical challenges for communication engineers is the development of best performance wireless
networks with negligible amount of distortions. We have to consider multipath propagation attenuation and
radio spectrum inefficiency. Now a days, In MIMO (Multi Input Multi Output) systems there is a huge demand
for the networks with the high transmission rates and better quality of service which are having low PAPR ratio.
Instead of OFDMA, filter banks are used in massive MIMO to reduce the complexity. But they are error prone
to noise. This base paper discusses about PAPR reduction in MIMO systems using different precoding based
OFDM systems. Mainly, minimization of multi-antenna systems by controlling the transmission power and
reduction of PAPR using ZC (Zadoff-Chu) matrix transform.
MARGINAL PERCEPTRON FOR NON-LINEAR AND MULTI CLASS CLASSIFICATION ijscai
Generalization error of classifier can be reduced by larger margin of separating hyperplane. The proposed classification algorithm implements margin in classical perceptron algorithm, to reduce generalized errors by maximizing margin of separating hyperplane. Algorithm uses the same updation rule with the perceptron, to converge in a finite number of updates to solutions, possessing any desirable fraction of the margin. This solution is again optimized to get maximum possible margin. The algorithm can process linear, non-linear and multi class problems. Experimental results place the proposed classifier equivalent to the support vector machine and even better in some cases. Some preliminary experimental results are briefly discussed.
A Comparative Study of DOA Estimation Algorithms with Application to Tracking...sipij
Tracking the Direction of Arrival (DOA) Estimation of a moving source is an important and challenging
task in the field of navigation, RADAR, SONAR, Wireless Sensor Networks (WSNs) etc. Tracking is carried
out starting from the estimation of DOA, considering the estimated DOA as an initial value, the Kalman
Filter (KF) algorithm is used to track the moving source based on the motion model which governs the
motion of the source. This comparative study deals with analysis, significance of Non-coherent,
Narrowband DOA (Direction of Arrival) Estimation Algorithms in perception to tracking. The DOA
estimation algorithms Multiple Signal Classification (MUSIC), Root-MUSIC& Estimation of Signal
Parameters via Rotational Invariance Technique (ESPRIT) are considered for the purpose of the study, a
comparison in terms of optimality with respect to Signal to Noise Ratio (SNR), number of snapshots and
number of Antenna elements used and Computational complexity is drawn between the chosen algorithms
resulting in an optimum DOA estimate. The optimum DOA Estimate is taken as an initial value for the
Kalman filter tracking algorithm. The Kalman filter algorithm is used to track the optimum DOA Estimate.
Blind Audio Source Separation (Bass): An Unsuperwised Approach IJEEE
Audio processing is an area where signal separation is considered as a fascinating works, potentially offering a vivid range of new scope and experience in professional and personal context. The objective of Blind Audio Source Separation is to separate audio signals from multiple independent sources in an unknown mixing environment. This paper addresses the key challenges in BASS and unsupervised approaches to counter these challenges. Comparative performance analysis of Fast-ICA algorithm and Convex Divergence ICA for Blind Source Separation is presented with the help of experimental result. Result reflects Convex Divergence ICA with α=-1 gives more accurate estimate in comparison of Fast ICA..
Kernal based speaker specific feature extraction and its applications in iTau...TELKOMNIKA JOURNAL
Extraction and classification algorithms based on kernel nonlinear features are popular in the new direction of research in machine learning. This research paper considers their practical application in the iTaukei automatic speaker recognition system (ASR) for cross-language speech recognition. Second, nonlinear speaker-specific extraction methods such as kernel principal component analysis (KPCA), kernel independent component analysis (KICA), and kernel linear discriminant analysis (KLDA) are summarized. The conversion effects on subsequent classifications were tested in conjunction with Gaussian mixture modeling (GMM) learning algorithms; in most cases, computations were found to have a beneficial effect on classification performance. Additionally, the best results were achieved by the Kernel linear discriminant analysis (KLDA) algorithm. The performance of the ASR system is evaluated for clear speech to a wide range of speech quality using ATR Japanese C language corpus and self-recorded iTaukei corpus. The ASR efficiency of KLDA, KICA, and KLDA technique for 6 sec of ATR Japanese C language corpus 99.7%, 99.6%, and 99.1% and equal error rate (EER) are 1.95%, 2.31%, and 3.41% respectively. The EER improvement of the KLDA technique-based ASR system compared with KICA and KPCA is 4.25% and 8.51% respectively.
An Ant Algorithm for Solving QoS Multicast Routing ProblemCSCJournals
Abstract: Many applications require send information from a source to multiple destinations through a communication network. To support these applications, it is necessary to determine a multicast tree of minimal cost to connect the source node to the destination nodes subject to delay constraints. Based on the Ant System algorithm, we present an ant algorithm to find the multicast tree that minimizes the total cost. In the proposed algorithm, the k shortest paths from the source node to the destination nodes are used for genotype representation. The expermintal results show that the algorithm can find optimal solution quickly and has a good scalability.
Medienübergreifende Repositorien - mehr als nur DokumentenserverRalf Claußnitzer
Um den weltweiten, freien und schnellen Zugriff auf Preprint-Artikel zu ermöglichen, entstand 1991 das heute weltweit bekannte Repository-System der Physik: ArXiv.org. Mit der Zeit griffen viele Einrichtungen diese Idee auf und entwickelten sie zum praktischen Konzept des Open Access Publikationsservers mit standardisierten und zertifizierten Schnittstellen weiter.
Zunächst nur für die Sammlung von ausgewählten Hochschulschriften vorgesehen, entwickelten sich diese Dienste zu Publikationsplattformen für Volltexte, zunehmend aber auch zu Sammlungen von multimedialen Dateiformaten. Dies ermöglichte für bestimmte Materialien zunächst die Abkehr von den klassischen Druckausgaben. Aber auch das Konzept des Dokumentenservers bleibt nicht vom Medienwandel verschont, der den Fokus vom Volltext auf zusammenhängend publizierte digitale Objekte verschiebt und heute auch nachträglich digitalisierte Werke sowie Forschungsdaten umfasst. Dieser Paradigmenwechsel bedeutet für Repositorien die Notwendigkeit zur Erneuerung ihrer Infrastruktur und bibliothekarischen Workflows. Die moderne Bibliothek muss hier ansetzen und das Dienstkonzept Dokumentenserver mit dem der Digitalen Sammlungen und Mediatheken vereinen, moderne Schnittstellen schaffen und den Zielkonflikt zwischen umfassender Sammlung und Langzeitarchivierung weitestgehend auflösen.
Das an der SLUB gestartete Projekt "Medienübergreifendes Repository“ hat zum Ziel, die sächsische Publikationsplattform Qucosa in diesem Sinne technologisch neu auszurichten. Im Zuge der Entwicklung soll eine Gesamtlösung für alle digitalen Objekte gefunden werden, die es ermöglicht, die vorhandenen Workflows miteinander zu vernetzen und unterschiedliche Präsentationsformate zu unterstützen. Der Vortrag gibt einen Überblick über die Anforderungen an eine solche Technologie sowie über die sich ergebenden Möglichkeiten und berichtet vom Stand der Systementwicklung sowie den Möglichkeiten der Nachnutzung der quelloffenen Gesamtlösung.
In order to improve sensing performance when the noise variance is not known, this paper considers a so-called
blind spectrum sensing technique that is based on eigenvalue models. In this paper, we employed the spiked population
models in order to identify the miss detection probability. At first, we try to estimate the unknown noise variance
based on the blind measurements at a secondary location. We then investigate the performance of detection, in terms
of both theoretical and empirical aspects, after applying this estimated noise variance result. In addition, we study the
effects of the number of SUs and the number of samples on the spectrum sensing performance.
In this paper, a new algorithm for a high resolution
Direction Of Arrival (DOA) estimation method for multiple
wideband signals is proposed. The proposed method proceeds
in two steps. In the first step, the received signals data is
decomposed in a Toeplitz form using the first-order statistics.
In the second step, The QR decomposition is applied on the
constructed Toeplitz matrix. Compared with existing schemes,
the proposed scheme provides several advantages. First, it
requires computing the triangular matrix R or the orthogonal
matrix Q to find the DOA; these matrices can be computed
with O(n2) operation. However, most of the existing schemes
required eignvalue decomposition (EVD) for the covariance
matrix or singular value decomposition (SVD) for the data
matrix; using EVD or SVD requires much more complex
computational O(n3) operation. Second, the proposed scheme
is more suitable for high-speed communication since it
requires first-order statistics and a single snapshot. Third,
the proposed scheme can estimate the correlated wideband
signals without using spatial smoothing techniques; whereas,
already-existing schemes do not. Accuracy of the proposed
wideband DOA estimation method is evaluated through
computer simulation in comparison with a conventional
method.
Methodological study of opinion mining and sentiment analysis techniquesijsc
Decision making both on individual and organizational level is always accompanied by the search of
other’s opinion on the same. With tremendous establishment of opinion rich resources like, reviews, forum
discussions, blogs, micro-blogs, Twitter etc provide a rich anthology of sentiments. This user generated
content can serve as a benefaction to market if the semantic orientations are deliberated. Opinion mining
and sentiment analysis are the formalization for studying and construing opinions and sentiments. The
digital ecosystem has itself paved way for use of huge volume of opinionated data recorded. This paper is
an attempt to review and evaluate the various techniques used for opinion and sentiment analysis.
Noisy Speech Enhancement Using Soft Thresholding on Selected Intrinsic Mode F...CSCJournals
In this paper, a new speech enhancement method is introduced. It is essentially based on the Empirical Mode Decomposition technique (EMD) and a soft thresholding approach applied on selected modes. The proposed method is a fully data driven approach. First the noisy speech signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs) by using a time decomposition called sifting process. Second, selected IMFs are soft thresholded and added to the remaining IMFs with the residue to reconstitute the enhanced speech signal. The proposed approach is evaluated using speech signals from NOISEUS database corrupted with additive white Gaussian noise. Our algorithm is compared to other state of the art algorithms.
Performance of Matching Algorithmsfor Signal Approximationiosrjce
IOSR Journal of Electronics and Communication Engineering(IOSR-JECE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of electronics and communication engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in electronics and communication engineering. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Reduction of PAPR and Efficient detection ordering scheme for MIMO Transmissi...IJERA Editor
The technical challenges for communication engineers is the development of best performance wireless
networks with negligible amount of distortions. We have to consider multipath propagation attenuation and
radio spectrum inefficiency. Now a days, In MIMO (Multi Input Multi Output) systems there is a huge demand
for the networks with the high transmission rates and better quality of service which are having low PAPR ratio.
Instead of OFDMA, filter banks are used in massive MIMO to reduce the complexity. But they are error prone
to noise. This base paper discusses about PAPR reduction in MIMO systems using different precoding based
OFDM systems. Mainly, minimization of multi-antenna systems by controlling the transmission power and
reduction of PAPR using ZC (Zadoff-Chu) matrix transform.
MARGINAL PERCEPTRON FOR NON-LINEAR AND MULTI CLASS CLASSIFICATION ijscai
Generalization error of classifier can be reduced by larger margin of separating hyperplane. The proposed classification algorithm implements margin in classical perceptron algorithm, to reduce generalized errors by maximizing margin of separating hyperplane. Algorithm uses the same updation rule with the perceptron, to converge in a finite number of updates to solutions, possessing any desirable fraction of the margin. This solution is again optimized to get maximum possible margin. The algorithm can process linear, non-linear and multi class problems. Experimental results place the proposed classifier equivalent to the support vector machine and even better in some cases. Some preliminary experimental results are briefly discussed.
A Comparative Study of DOA Estimation Algorithms with Application to Tracking...sipij
Tracking the Direction of Arrival (DOA) Estimation of a moving source is an important and challenging
task in the field of navigation, RADAR, SONAR, Wireless Sensor Networks (WSNs) etc. Tracking is carried
out starting from the estimation of DOA, considering the estimated DOA as an initial value, the Kalman
Filter (KF) algorithm is used to track the moving source based on the motion model which governs the
motion of the source. This comparative study deals with analysis, significance of Non-coherent,
Narrowband DOA (Direction of Arrival) Estimation Algorithms in perception to tracking. The DOA
estimation algorithms Multiple Signal Classification (MUSIC), Root-MUSIC& Estimation of Signal
Parameters via Rotational Invariance Technique (ESPRIT) are considered for the purpose of the study, a
comparison in terms of optimality with respect to Signal to Noise Ratio (SNR), number of snapshots and
number of Antenna elements used and Computational complexity is drawn between the chosen algorithms
resulting in an optimum DOA estimate. The optimum DOA Estimate is taken as an initial value for the
Kalman filter tracking algorithm. The Kalman filter algorithm is used to track the optimum DOA Estimate.
Blind Audio Source Separation (Bass): An Unsuperwised Approach IJEEE
Audio processing is an area where signal separation is considered as a fascinating works, potentially offering a vivid range of new scope and experience in professional and personal context. The objective of Blind Audio Source Separation is to separate audio signals from multiple independent sources in an unknown mixing environment. This paper addresses the key challenges in BASS and unsupervised approaches to counter these challenges. Comparative performance analysis of Fast-ICA algorithm and Convex Divergence ICA for Blind Source Separation is presented with the help of experimental result. Result reflects Convex Divergence ICA with α=-1 gives more accurate estimate in comparison of Fast ICA..
Kernal based speaker specific feature extraction and its applications in iTau...TELKOMNIKA JOURNAL
Extraction and classification algorithms based on kernel nonlinear features are popular in the new direction of research in machine learning. This research paper considers their practical application in the iTaukei automatic speaker recognition system (ASR) for cross-language speech recognition. Second, nonlinear speaker-specific extraction methods such as kernel principal component analysis (KPCA), kernel independent component analysis (KICA), and kernel linear discriminant analysis (KLDA) are summarized. The conversion effects on subsequent classifications were tested in conjunction with Gaussian mixture modeling (GMM) learning algorithms; in most cases, computations were found to have a beneficial effect on classification performance. Additionally, the best results were achieved by the Kernel linear discriminant analysis (KLDA) algorithm. The performance of the ASR system is evaluated for clear speech to a wide range of speech quality using ATR Japanese C language corpus and self-recorded iTaukei corpus. The ASR efficiency of KLDA, KICA, and KLDA technique for 6 sec of ATR Japanese C language corpus 99.7%, 99.6%, and 99.1% and equal error rate (EER) are 1.95%, 2.31%, and 3.41% respectively. The EER improvement of the KLDA technique-based ASR system compared with KICA and KPCA is 4.25% and 8.51% respectively.
An Ant Algorithm for Solving QoS Multicast Routing ProblemCSCJournals
Abstract: Many applications require send information from a source to multiple destinations through a communication network. To support these applications, it is necessary to determine a multicast tree of minimal cost to connect the source node to the destination nodes subject to delay constraints. Based on the Ant System algorithm, we present an ant algorithm to find the multicast tree that minimizes the total cost. In the proposed algorithm, the k shortest paths from the source node to the destination nodes are used for genotype representation. The expermintal results show that the algorithm can find optimal solution quickly and has a good scalability.
Medienübergreifende Repositorien - mehr als nur DokumentenserverRalf Claußnitzer
Um den weltweiten, freien und schnellen Zugriff auf Preprint-Artikel zu ermöglichen, entstand 1991 das heute weltweit bekannte Repository-System der Physik: ArXiv.org. Mit der Zeit griffen viele Einrichtungen diese Idee auf und entwickelten sie zum praktischen Konzept des Open Access Publikationsservers mit standardisierten und zertifizierten Schnittstellen weiter.
Zunächst nur für die Sammlung von ausgewählten Hochschulschriften vorgesehen, entwickelten sich diese Dienste zu Publikationsplattformen für Volltexte, zunehmend aber auch zu Sammlungen von multimedialen Dateiformaten. Dies ermöglichte für bestimmte Materialien zunächst die Abkehr von den klassischen Druckausgaben. Aber auch das Konzept des Dokumentenservers bleibt nicht vom Medienwandel verschont, der den Fokus vom Volltext auf zusammenhängend publizierte digitale Objekte verschiebt und heute auch nachträglich digitalisierte Werke sowie Forschungsdaten umfasst. Dieser Paradigmenwechsel bedeutet für Repositorien die Notwendigkeit zur Erneuerung ihrer Infrastruktur und bibliothekarischen Workflows. Die moderne Bibliothek muss hier ansetzen und das Dienstkonzept Dokumentenserver mit dem der Digitalen Sammlungen und Mediatheken vereinen, moderne Schnittstellen schaffen und den Zielkonflikt zwischen umfassender Sammlung und Langzeitarchivierung weitestgehend auflösen.
Das an der SLUB gestartete Projekt "Medienübergreifendes Repository“ hat zum Ziel, die sächsische Publikationsplattform Qucosa in diesem Sinne technologisch neu auszurichten. Im Zuge der Entwicklung soll eine Gesamtlösung für alle digitalen Objekte gefunden werden, die es ermöglicht, die vorhandenen Workflows miteinander zu vernetzen und unterschiedliche Präsentationsformate zu unterstützen. Der Vortrag gibt einen Überblick über die Anforderungen an eine solche Technologie sowie über die sich ergebenden Möglichkeiten und berichtet vom Stand der Systementwicklung sowie den Möglichkeiten der Nachnutzung der quelloffenen Gesamtlösung.
An approach for software effort estimation using fuzzy numbers and genetic al...csandit
One of the most critical tasks during the software development life cycle is that of estimating the
effort and time involved in the development of the software product. Estimation may be
performed by many ways such as: Expert judgments, Algorithmic effort estimation, Machine
learning and Analogy-based estimation. In which Analogy-based software effort estimation is
the process of identifying one or more historical projects that are similar to the project being
developed and then using the estimates from them. Analogy-based estimation is integrated with
Fuzzy numbers in order to improve the performance of software project effort estimation during
the early stages of a software development lifecycle. Because of uncertainty associated with
attribute measurement and data availability, fuzzy logic is introduced in the proposed model.
But hardly a historical project is exactly same as the project being estimated due to some
distance associated in similarity distance. This means that the most similar project still has a
similarity distance with the project being estimated in most of the cases. Therefore, the effort
needs to be adjusted when the most similar project has a similarity distance with the project
being estimated. To adjust the reused effort, we build an adjustment mechanism whose
algorithm can derive the optimal adjustment on the reused effort using Genetic Algorithm. The
proposed model Combine the fuzzy logic to estimate software effort in early stages with Genetic
algorithm based adjustment mechanism may result to near the correct effort estimation.
Combined feature extraction techniques and naive bayes classifier for speech ...csandit
Speech processing and consequent recognition are important areas of Digital Signal Processing
since speech allows people to communicate more natu-rally and efficiently. In this work, a
speech recognition system is developed for re-cognizing digits in Malayalam. For recognizing
speech, features are to be ex-tracted from speech and hence feature extraction method plays an
important role in speech recognition. Here, front end processing for extracting the features is
per-formed using two wavelet based methods namely Discrete Wavelet Transforms (DWT) and
Wavelet Packet Decomposition (WPD). Naive Bayes classifier is used for classification purpose.
After classification using Naive Bayes classifier, DWT produced a recognition accuracy of
83.5% and WPD produced an accuracy of 80.7%. This paper is intended to devise a new
feature extraction method which produces improvements in the recognition accuracy. So, a new
method called Dis-crete Wavelet Packet Decomposition (DWPD) is introduced which utilizes
the hy-brid features of both DWT and WPD. The performance of this new approach is evaluated
and it produced an improved recognition accuracy of 86.2% along with Naive Bayes classifier.
Analytical study of feature extraction techniques in opinion miningcsandit
Although opinion mining is in a nascent stage of development but still the ground is set for
dense growth of researches in the field. One of the important activities of opinion mining is to
extract opinions of people based on characteristics of the object under study. Feature extraction
in opinion mining can be done by various ways like that of clustering, support vector machines
etc. This paper is an attempt to appraise the various techniques of feature extraction. The first
part discusses various techniques and second part makes a detailed appraisal of the major
techniques used for feature extraction
Intersymbol interference caused by multipath in band limited frequency selective time dispersive channels distorts the transmitted signal, causing bit error at receiver. ISI is the major obstacle to high speed data transmission over wireless channels. Channel estimation is a technique used to combat the intersymbol interference. The objective of this paper is to improve channel estimation accuracy in MIMO-OFDM system by using modified variable step size leaky Least Mean Square (MVSSLLMS) algorithm proposed for MIMO OFDM System. So we are going to analyze Bit Error Rate for different signal to noise ratio, also compare the proposed scheme with standard LMS channel estimation method.
RESOLVING CYCLIC AMBIGUITIES AND INCREASING ACCURACY AND RESOLUTION IN DOA ES...csandit
A method to resolve cyclic ambiguities and increase the accuracy and the resolution in the
direction-of-arrival (DOA) estimation using the Estimation of Signal Parameters via Rotational
Invariance Technique (ESPRIT)algorithm is proposed. It is based on rotating the array and
sampling the received signal at multiple positions. Using this approach, the gain in accuracy
and resolution is addressed as function of the mean and variance of the DOA. Simulations
results are provided as a means of verifying this analysis.
CHANNEL ESTIMATION AND MULTIUSER DETECTION IN ASYNCHRONOUS SATELLITE COMMUNIC...ijwmn
In this paper, we propose a new method of channel estimation for asynchronous additive white Gaussian noise channels in satellite communications. This method is based on signals correlation and multiuser interference cancellation which adopts a successive structure. Propagation delays and signals amplitudes are jointly estimated in order to be used for data detection at the receiver. As, a multiuser detector, a single stage successive interference cancellation (SIC) architecture is analyzed and integrated to the channel estimation technique and the whole system is evaluated. The satellite access method adopted is the direct sequence code division multiple access (DS CDMA) one. To evaluate the channel estimation and the detection technique, we have simulated a satellite uplink with an asynchronous multiuser access.
Performance Analysis of Ultra Wideband Receivers for High Data Rate Wireless ...graphhoc
For high data rate ultra wideband communication system, performance comparison of Rake, MMSE and Rake-MMSE receivers is attempted in this paper. Further a detail study on Rake-MMSE time domain equalizers is carried out taking into account all the important parameters such as the effect of the number of Rake fingers and equalizer taps on the error rate performance. This receiver combats inter-symbol interference by taking advantages of both the Rake and equalizer structure. The bit error rate performances are investigated using MATLAB simulation on IEEE 802.15.3a defined UWB channel models. Simulation results show that the bit error rate probability of Rake-MMSE receiver is much better than Rake receiver and MMSE equalizer. Study on non-line of sight indoor channel models illustrates that bit error rate performance of Rake-MMSE (both LE and DFE) improves for CM3 model with smaller spread compared to CM4 channel model. It is indicated that for a MMSE equalizer operating at low to medium SNR values, the number of Rake fingers is the dominant factor to improve system performance, while at high SNR values the number of equalizer taps plays a more significant role in reducing the error rate.
Performance analysis of adaptive filter channel estimated MIMO OFDM communica...IJECEIAES
Advanced Communication Systems are wideband systems to support multiple applications such as audio, video and data so and so forth. These systems require high spectral efficiency and data rates. In addition, they should provide multipath fading and inter-symbol interference (ISI) free transmission. Multiple input multiple output orthogonal frequency division multiplexing (MIMO OFDM) meets these requirements Hence, MIMOOFDM is the most preferable technique for long term evaluation advanced (LTEA). The primary objective of this paper is to control bit error rate (BER) by proper channel coding, pilot carriers, adaptive filter channel estimation schemes and space time coding (STC). A combination of any of these schemes results in better BER performance over individual schemes. System performance is analyzed for various digital modulation schemes. In this paper, adaptive filter channel estimated MIMO OFDM system is proposed by integrating channel coding, adaptive filter channel estimation, digital modulation and space time coding. From the simulation results, channel estimated 2×2 MIMO OFDM system shows superior performance over individual schemes.
Channel Equalization of WCDMA Downlink System Using Finite Length MMSE-DFEIOSR Journals
Abstract: The performance of WCDMA system deteriorates in the presence of multipath fading environment. Fading destroys the orthogonality and is responsible for multiple access interference (MAI). Though conventional rake receiver provides reasonable performance in the WCDMA downlink system due to path diversity, but it does not restores the orthogonality. Linear equalizer restores orthogonality and suppresses MAI, but it is not efficient, since its performance depends on the spectral characteristics of the channel. To overcome this, Minimum Mean Square Error- Decision Feedback Equalizer (MMSE-DFE) with a linear, anticausal feedforward filter, causal feedback filter and simple detector is proposed in this paper. The filter taps of finite length DFE is derived using the cholesky factorization theory, capable of suppressing noise, Intersymbol Interference (ISI) and MAI. This paper describes the WCDMA downlink system using finite length MMSE-DFE and takes into consideration the effects of interference which includes additive white gaussian noise, multipath fading, ISI and MAI. Furthermore, the performance is compared with conventional rake receiver and MMSE and the simulation results are shown. Keywords – MMSE, MMSE-DFE, rake receiver, WCDMA
Channel Equalization of WCDMA Downlink System Using Finite Length MMSE-DFEIOSR Journals
The performance of WCDMA system deteriorates in the presence of multipath fading environment.
Fading destroys the orthogonality and is responsible for multiple access interference (MAI). Though
conventional rake receiver provides reasonable performance in the WCDMA downlink system due to path
diversity, but it does not restores the orthogonality. Linear equalizer restores orthogonality and suppresses
MAI, but it is not efficient, since its performance depends on the spectral characteristics of the channel. To
overcome this, Minimum Mean Square Error- Decision Feedback Equalizer (MMSE-DFE) with a linear,
anticausal feedforward filter, causal feedback filter and simple detector is proposed in this paper. The filter taps
of finite length DFE is derived using the cholesky factorization theory, capable of suppressing noise,
Intersymbol Interference (ISI) and MAI. This paper describes the WCDMA downlink system using finite length
MMSE-DFE and takes into consideration the effects of interference which includes additive white gaussian
noise, multipath fading, ISI and MAI. Furthermore, the performance is compared with conventional rake
receiver and MMSE and the simulation results are shown.
BER Performance of Antenna Array-Based Receiver using Multi-user Detection in...ijwmn
Antenna promises to provide significant increases in system capacity and performance in
wireless systems. In this paper, a simplified, near-optimum array receiver is proposed,
which is based on the angular gain of the spatial filter. This detection is then analyzed by
calculating the exact error probability.The proposed model confirms the benefits of adaptive
antennas in reducing the overall interference level (intercell/intracell) and to find an
accurate approximation of the error probability. We extend the method that has been
proposed for propagation over Nakagami-m fading channels, the model shows good
agreements with simulation results.
Method for Converter Synchronization with RF InjectionCSCJournals
This paper presents an injection method for synchronizing analog to digital converters (ADC). This approach can eliminate the need for precision routed discrete synchronization signals of current technologies, such as JESD204. By eliminating the setup and hold time requirements at the conversion (or near conversion) clock rate, higher sample rate systems can be synchronized. Measured data from an existing multiple ADC conversion system was used to evaluate the method. Coherent beams were simulated to measure the effectiveness of the method. The results show near theoretical coherent processing gain.
Performance Analysis of Adaptive DOA Estimation Algorithms For Mobile Applica...IJERA Editor
Spatial filtering for mobile communications has attracted a lot of attention over the last decade and is cur-rently considered a very promising technique that will help future cellular networks achieve their ambi-tious goals. One way to accomplish this is via array signal processing with algorithms which estimate the Direction-Of-Arrival (DOA) of the received waves from the mobile users. This paper evaluates the per-formance of a number of DOA estimation algorithms. In all cases a linear antenna array at the base station is assumed to be operating typical cellular environment.
Multi carrier equalization by restoration of redundanc y (merry) for adaptive...IJNSA Journal
This paper proposes a new blind adaptive channel shortening approach for multi-carrier systems. The
performance of the discrete Fourier transform-DMT (DFT-DMT) system is investigated with the proposed
DST-DMT system over the standard carrier serving area (CSA) loop1. Enhanced bit rates demonstrated
and less complexity also involved by the simulation of the DST-DMT system.
In this paper, a novel architecture of RNS based 1D Lifting Integer Wavelet Transform (IWT) has been introduced. Advantage of Residue Number System (RNS) based Lifting Scheme over RNS based Filter Bank and non-binary IWT has been discussed. The performance of traditional predicts and updates stage of binary Lifting Scheme (LS) for Discrete Wavelet Transform (DWT) generates huge carry propagation
delay, power and complexity. As a result non binary number system is becoming popular in the field of Digital Signal Processing (DSP) due to its efficient performance. In this paper also a new fixed number ROM based RNS division circuit has been proposed. The proposed architecture has been validated on Xilinx Vertex5 FPGA platform and the corresponding result and reports are shown in here.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
Performance Analysis of Group-Blind Multiuser Detectors for Synchronous CDMAidescitation
Blind multiuser detectors are attractive for the
suppression of interference in a CDMA environment. This
paper deals with the performance of group blind multiuser
detector for synchronous CDMA is analyzed. The blind multi
user detectors are Direct Matrix Inversion(DMI),Subspace
and group blind multiuser detector. The performance
analysis is performed by means of the Signal to Interference
Noise Ratio(SINR) and Bit Error Rate(BER). The numerical
results are plotted as variation of SINR Vs SNR, K and M,
SINR with respect to correlation coefficient( ) and BER
Vs Number of samples(M) for three detectors using
MATLAB software. The gain rises in group blind multiuser
detector over the DMI and subspace detectors. The
comparison is carried out for equicorrelated signals for
mathematical simplicity.
Similar to Adaptive blind multiuser detection under impulsive noise using principal components (20)
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GraphRAG is All You need? LLM & Knowledge GraphGuy Korland
Guy Korland, CEO and Co-founder of FalkorDB, will review two articles on the integration of language models with knowledge graphs.
1. Unifying Large Language Models and Knowledge Graphs: A Roadmap.
https://arxiv.org/abs/2306.08302
2. Microsoft Research's GraphRAG paper and a review paper on various uses of knowledge graphs:
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All of this illustrated with link prediction over knowledge graphs, but the argument is general.
Slack (or Teams) Automation for Bonterra Impact Management (fka Social Soluti...Jeffrey Haguewood
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JMeter webinar - integration with InfluxDB and GrafanaRTTS
Watch this recorded webinar about real-time monitoring of application performance. See how to integrate Apache JMeter, the open-source leader in performance testing, with InfluxDB, the open-source time-series database, and Grafana, the open-source analytics and visualization application.
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Length: 30 minutes
Session Overview
-------------------------------------------
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- What out-of-the-box solutions are available for real-time monitoring JMeter tests?
- What are the benefits of integrating InfluxDB and Grafana into the load testing stack?
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Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
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Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
2. 124 Computer Science & Information Technology (CS & IT)
subspace method (SS) [2] is known for its better performance. However, the batch algorithm [2]
requires singular value decomposition (SVD), which is inconvenient for adaptive implementation.
Further, it requires accurate rank estimation of the correlation matrix, which is not easy in an
inherently noisy environment [4]-[9].
All the subbspace tracking algorithms were found with computational complexity around
2
( )O m ,
or where d is the dimension of the signal subspace, in each recursion [3]. However, the
total computation of the blind channel identification is still due to the second step, i.e.,
using the estimated subspace vectors to optimize channel estimation. The second step cannot be
repeatedly implemented with reduced computations in an obvious manner[11]. Least square(LS)
method is a simple way to estimate a Channel ,it’s efficiency is not significant . Principal
component analysis does not assume on the independence of the data vectors. When many
combinations are required PCA identifies linear combinations with the largest variances , it
considers variances in decreasing order of importance. In PCA signal to noise ratio of signals
used for differential side channel analysis is improved. By applying PCA to the frequency
spectrum noise and signals can be separated.
These detectors employ the PCA algorithm [9] at the outputs of a bank of matched filters. Main
purpose is to retrive the original sequence from the received signal that is corrupted by other
users and MAI, without the help of training sequences and a priori knowledge of the channel.
The paper is organized as follows :Section 2 describes the System model followed by section 3
with the description of the algorithm used. In section 4 contains results and conclusions.
2. SYSTEM MODELLING
Consider a DS-CDMA channel operating with coherent binary phase shift keying (BPSK)
modulation format with L users sharing the same bandwidth. If L users are active, the received
base band, continuous-time signal is a superposition of all L signals[4],
∞≤≤−∞+= ttztXty ),()()( (1)
where X(t) is the useful signal comprised of the data signals of L active users and z(t) the additive
white Gaussian noise. The useful signal X(t) is taken as the expression
)()()(
1
01
ll
M
i
l
L
l
l iTtxibwtX τ−−= ∑∑
−
== (2)
where M stands for the number of data symbols per user in the data frame of interest, T is the
symbol interval, wl is the received signal amplitude, lτ is the delay, }1,.....,1,0,)({ −= Miibl is the
symbol stream, and )(txl , { }Tt0txl ≤≤);(
is the normalized signaling waveform of the lth user. It
is assumed that the signaling interval of each user is T seconds, and the input alphabet is
antipodal binary: ]1,1[ +− . Further, direct-sequence spread-spectrum (DS-SS) multiple access
signatures may be written as a succession of the chip pulses as
( )O md
3
( )O m
3. Computer Science & Information Technology (CS & IT) 125
],0[),()(
1
0
TtjTtptx c
N
j
l
jl ∈−= ∑
−
=
ϕ
(3)
where N is the processing gain, ( )l
N
ll
ppp 110 ,..., − is the signature sequence of s1′± assigned to the lth
user, and ϕ is the normalized chip waveform of duration cT
, with TNTc =
.
A. Synchronous Case
For the synchronous case (i.e., 0....21 ==== lτττ ) of model (1), to demodulate the i th symbols
of the L users, { }L
ll ib 1
)( = , it is sufficient to consider the received signal during the i th signaling
interval, i.e.,
])1(,[),()()()(
1
TiiTttziTtxibwty ll
L
l
l +∈+−= ∑= (4)
Near the receiver, y(t) is the received signal it is filtered by a chip-matched filter and then
sampled at the chip rate. The resulting discrete-time signal sample corresponding to the j th chip
of the i th symbol is given by
dtjTiTttyiy
c
c
TjiT
jTiT
cj )()()(
)1(
∫
++
+
−−= ϕ
(or)
)()(
1
izxibw l
l
jl
L
l
l += ∑= (5)
(6)
where is the normalized signature sequence of the l th
user, and is the channel ambient noise sample vector at the i th symbol
interval. The term
{ })(iz j
is assumed as a sequence of independent and identically distributed
(i.i.d.) random variables with a non Gaussian distribution. In this paper, the pdf of impulsive
noise model is modeled as commonly used two-term Gaussian mixture model for the additive
noise samples,{ })(iz j
, which has the form
(7)
with , 10 ≤≤ ε , and 1≥κ . term represents the nominal background noise and the
term represents an impulsive component, with ε representing the probability of
1
( ) ( ) ( )
L
l l l
l
i w b i i
=
= +∑y x z
0 1 1 0 1 1
1
, ,..., , ,..,
T Tl l l l l l
l N Nx x x p p p
N
− −
= = x
[ ]0 1 1( ) ( ), ( ),..., ( )
T
Ni z i z i z i−=z
2 2
(1 ) (0, ) (0, )f v vε ϖ εϖ κ= − +
0v >
2
(0, )vϖ
2
(0, )vϖ κ
4. 126 Computer Science & Information Technology (CS & IT)
occurrence of impulses. This model serves as an approximation to the more fundamental
Middleton Class A noise model and also has been used extensively to model physical noise
arising in radio and acoustic channels [4,10, 11].
B. Extension to Asynchronous Case
This algorithm is applicable for the users when they are symbol and chip asynchronous. After
some algebra, asynchronous signal model given in (1) can be written as [5]
(8)
where
(9)
where Hm denotes the )1( +× MLNM Hermitian cross correlation matrix of the composite
waveforms associated with the symbols in b and the matrices and are defined as
(10)
with
(11)
and . The parameter m is called the smoothing factor and is chosen such
that the Sylvester matrix has full column rank. If the signature waveforms of all users as well
as the relative delays are known to the receiver, then the Sylvester matrix can be constructed.
Here, using PCA, principal components are calculated for received ym (i) as in eq(4)
The output matrix is is found as the covariance matrix
( ) ( ) ( )m m m m mi i i= +y Η W b z
( 1)
(1) (0) ...
: ... ... :
: ... ... :
... (1) (0)
m
Nm L m× +
≅
H H 0
H
0 H H
(0)H (1)H
[ ]1
(0)
...
(1)
L
≅
H
h h
H
1
1
(1 )
l l
l l
m m
l l l l l
N m N m
π π
+
− − −
≅ − +
0 0
h x x
0 0
&l l
l l l
c c
m m
T T
τ τ
π
= = −
mH
mH
5. Computer Science & Information Technology (CS & IT) 127
(1) (2) .... ( )
(0) (1) .... ( 1)
( 1) (0) .... ( 2)
(1)
y y y t
y y y t
y
y y y t
y
− =
− −
M M O
(12)
C = Covariance(y) (13)
T
C U V= Σ (14)
3. PRINCIPAL COMPONENT ANALYSIS ALGORITHM
PCA is a technique which is generally used for reducing the dimensionality of multivariate data
sets i.e. reducing the number of dimensions, without much loss of information .The covariance
matrix of n random variables x is Σ, the principal components are defined as
z = Ax
Where A is n by n orthogonal matrix with rows of that are eigen vectors of covariance matrix as
the rows of z is the vector of n principal components (PCs) . The Eigen values of Σ are
proportional to the fraction of the total variance accounted for by the corresponding eigenvectors,
so the PCs explaining most of the variance in the original variables can be identified. Regularly
some of the original variables are correlated like a small group of the PCs describes a large space
of the variance of the main data.
The data matrix X of size m x n is considered as received signal.
1 1 1 1( ) [ ( ), ( ), ( ), ( )]X t x t x t x t x t= KK (15)
is the time ordered collection of a single matrix to which PCA can be applied. The means of the
xi are removed and the covariance matrix computed by
1 T
X X
n
∑ = (16)
∑ is an m x m square matrix, Eigen values, and corresponding eigenvectors will be calculated,
First eigenvectors are calculated from the covariance matrix, second step is to arrange them by
Eigen values in descending order .This gives you the components in order of significance. The
lesser Eigen values are ignored there by reducing the data. Principal components are ordered
eigenvectors of the covariance matrix. The PCs were obtained using
zj=aj x j=1,2, …….n
The PCs are a linear transformation with transformation coefficients given by the eigenvectors .
PCA is solved by using covariance matrix and singular value decomposition (SVD) methods.
6. 128 Computer Science & Information Technology (CS & IT)
Let X be an arbitrary n × m matrix and
T
X X . be a rank r, square, symmetric m × m matrix.
1 2 3
ˆ ˆ ˆ ˆ{ , , ...... }rv v v v is the set of orthonormal m×1 eigenvectors with associated eigenvalues for
1 2 3{ , , ,...... }rλ λ λ λ the symmetric matrix
T
X X .
ˆ ˆ( )T
i i iX X v vλ= (17)
i iσ λ=
are positive, real and named as the singular values
1 2 3
ˆ ˆ ˆ ˆ{ , , ...... }ru u u u is the group of n×1 vectors defined by
[ ]1ˆ ˆi i
i
u X v
σ=
1
ˆ ˆ
0
i j
i j
u u
i j
=
=
≠ Eigenvectors are orthonormal. (18)
ˆi iX v σ=
The scalar expansion of SVD is
ˆ ˆi i iX v uσ=
X multiplied by an eigenvector of
T
X X .is equal to a scalar times another vector. The set of
eigenvectors 1 2 3
ˆ ˆ ˆ ˆ{ , , ...... }rv v v v and the set of vectors are 1 2 3
ˆ ˆ ˆ ˆ{ , , ...... }ru u u u both orthonormal sets
and bases in r dimensional space.
1 0
0 0
γ
σ
σ
∑ =
%
K
O
M M
L (19)
1 2 3 γσ σ σ σ≥ ≥ ≥ %K
are the rank-ordered set of singular values. Likewise we construct
accompanying orthogonal matrices,
[ ]1 2 3
ˆ ˆ ˆ ˆ, , ...... rV v v v v=
(20)
[ ]1 2 3
ˆ ˆ ˆ ˆ, , ...... rU u u u u=
(21)
Matrix expansion of SVD
XV U= ∑ (22)
7. Computer Science & Information Technology (CS & IT) 129
where each column of V and U perform the scalar version of the decomposition (Equation 3).
Because V is orthogonal, we can multiply both sides by V−1 = ்ܸ
to arrive at the final form of
the decomposition.
T
XV U V= ∑
Implementation of PCA is as follows
The description of PCA using the covariance method is as follows:
The purpose is to transform one data set X of dimension M to an another data set Y of smaller
dimension L.
Step 1: Get some data and organize the data set.
Step 2: Subtract the mean:
For better performance of PCA , subtract the mean from each of the data dimensions.
The mean subtracted is equal to the average across each dimension.
Step 3: Calculate the covariance matrix.
Step 4: Calculate the eigenvectors and Eigen values of the covariance matrix.
Step 5: Choosing components and forming a feature vector:
The eigenvector containing the maximum Eigen value is considered as the principle
component of the data set.
Feature Vector = (eig1, eig2, eig3, eig4,………………, eign) .
Step 6: Deriving the new data set:
4. EXPERIMENTAL RESULTS
In simulations, first a synchronous system with 6 users is considered. The spreading sequence of
each user with length 31 is considered (as shown in the following experiments). Figure 1(a-f)
shows the bit error rate and signal to noise ratio comparisions for 6 different users which are
estimated using corresponding principal componets. And the received users powers are all equal
for the considered detectors (i.e., in the case of tight power control). The noise distribution
parameters for are fixed as and , respectively. These plots demonstrate the
performance gains achieved by the robust blind multiuser detector with PCA in impulsive noise.
Next, an asynchronous system with 6 users and the noise distribution parameters ε =0.01 & 0.1
and 100=κ are considered. The delays of the 6 users are randomly generated. Fig 1( a-f) show
the transmitter and estimated symbols for user 1.and Fig 2 show the BER for 6 different users
with different Principal compon. Fig 3. Shows the details of transmitted and received symbols
for of user one. Fig 4 gives the details of proposed method and subspace method. It is observed
from these plots that the robust blind multiuser detector with the proposed PCA technique
outperforms the traditional subspace method based linear MMSE detector in impulsive noise.
The performance gain increases as the SNR increased.
0.01&0.1ε = 100κ =
8. 130 Computer Science & Information Technology (CS & IT)
(a) (b)
(c) (d)
(e) (f)
Fig 1. BER and SNR Plots for different users using first 6 Principal Components
Fig 2 BER for 6 different users with different Principal components
-10 -5 0 5 10 15
10
-3
10
-2
10
-1
10
0
SNR
BER
-10 -8 -6 -4 -2 0 2 4 6 8 10
10
-2
10
-1
10
0
SNR
BER
-10 -5 0 5 10 15
10
-3
10
-2
10
-1
10
0
SNR
BER
user1
user2
user3
user4
user5
user6
-10 -5 0 5 10 15
10
-3
10
-2
10
-1
10
0
SNR
BER
-10 -5 0 5 10 15
10
-3
10
-2
10
-1
10
0
SNR
BER
-10 -5 0 5 10 15
10
-3
10
-2
10
-1
10
0
SNR
BER
-10 -8 -6 -4 -2 0 2 4 6 8 10
10
-2
10
-1
10
0
SNR
BER
9. Computer Science & Information Technology (CS & IT) 131
Fig 3. Transmitted and received symbols for user 1
5. CONCLUSIONS
In this paper, Multiuser detection has been done using principal components in asynchronous
non-Gaussian channels is proposed and analyzed. Even the process of computing principal
components is complex, simulation results show that the robust blind multiuser detector with the
proposed PCA technique outperforms the traditional subspace method based linear MMSE
detector in both synchronous and asynchronous CDMA non-Gaussian channels.
REFERENCES
[1] S. Verdu, Multiuser Detection, Cambridge University Press, Cambridge, 1998.
[2] E. Moulines, P. Duhamel, J. Cardoso, and S. Mayrargue, “Subspace Methods for the Blind
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Transmitted Symbols of User 1
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Received Noisy Signals
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Equalized symbols of User 1
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Scaled Equalized Symbols
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