Nguyen and Yamada [NY'13] proposed an adaptive algorithm for fast and stable extraction of the first generalized Hermitian eigen-vector and mentioned the extension to the first r generalized eigenvector extraction...
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ISBN:
(纸本)9781479999880
Nguyen and Yamada [NY'13] proposed an adaptive algorithm for fast and stable extraction of the first generalized Hermitian eigen-vector and mentioned the extension to the first r generalized eigenvector extraction based on the nested orthogonal complement structure [NTY'12]. However, we recently found that the estimates of the eigenvectors are not expressed ideally in the time-varying coordinate system and can change drastically in a certain situation, which may cause numerical instability. In this paper, we propose a new expression of the estimates along with time-varying coordinate system. This modification can be done efficiently with additional multiplications of orthogonal complement matrices. Numerical experiments show that the modified scheme has better stability compared with the original scheme [NTY'12].
Scheduling of scientific workflows in IaaS clouds with payper-use pricing model and multiple types of virtual machines is an important challenge. Most static scheduling algorithms assume that the estimates of task run...
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ISBN:
(纸本)9783319321523;9783319321516
Scheduling of scientific workflows in IaaS clouds with payper-use pricing model and multiple types of virtual machines is an important challenge. Most static scheduling algorithms assume that the estimates of task runtimes are known in advance, while in reality the actual runtime may vary. To address this problem, we propose an adaptive scheduling algorithm for deadline constrained workflows consisting of multiple levels. The algorithm produces a global approximate plan for the whole workflow in a first phase, and a local detailed schedule for the current level of the workflow. By applying this procedure iteratively after each level completes, the algorithm is able to adjust to the runtime variation. For each phase we propose optimization models that are solved using Mixed Integer Programming (MIP) method. The preliminary simulation results using data from Amazon infrastructure, and both synthetic and Montage workflows, show that the adaptive approach has advantages over a static one.
In real world application noise can degrade the performance of the system. It is affected by impulsive noise, as this type of noise highly depends on physical environment. Though adaptive filters are widely used for n...
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ISBN:
(纸本)9781509012770
In real world application noise can degrade the performance of the system. It is affected by impulsive noise, as this type of noise highly depends on physical environment. Though adaptive filters are widely used for noise cancellation, still it requires robust parameter for all type of noise. In this paper, an attempt is taken for reduction of impulsive noise. Initially, the popular LMS algorithm is used for comparison and extends towards Fx-LMS. It shows significant result, but becomes unstable the due to the second order moment does not exist for Gaussian process. Further it has been analyzed with NLMS and the norm is exploited. It is found that WLMS algorithm performs better than all the above algorithms and also robust for impulsive noise. The results show its comparison performance.
An acoustic echo cancellation system is one of the most important breakthrough in the field of adaptive systems. Today acoustic echo cancellers (AEC) are an integral part of full duplex hands-free voice communication....
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ISBN:
(纸本)9783319120126;9783319120119
An acoustic echo cancellation system is one of the most important breakthrough in the field of adaptive systems. Today acoustic echo cancellers (AEC) are an integral part of full duplex hands-free voice communication. Conventional echo cancellers use a linear model to represent the echo path. However many consumer devices include loud-speakers and power amplifiers that generate non-linear distortions. Non-linearity occurs due to the use of low cost electronic loud speakers, microphones and poorly designed enclosures in an AEC system. Non-linearity causes vibration and harmonic distortion and also degrades the speech quality. Double talk detector (DTD) is a key component of an AEC. A DTD is used to sense when the far end signal is corrupted by the near end speech. The DTD freezes the adaptation of model filter to prevent the divergence of the adaptive filter. Various authors have proposed different algorithms for double talk detection. Some of the most popular algorithms are Geigel algorithm, cross-correlation based DTD, normalized cross correlation based DTD, variable impulse response DTD etc. In this paper several double talk detection algorithms in a non-linear platform of an AEC has been discussed.
In networked mobile multitarget tracking systems, parameters such as detection probabilities, clutter rates, and motion model parameters are often unknown and time-varying. Such parameter variability can seriously deg...
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ISBN:
(纸本)9781509024827
In networked mobile multitarget tracking systems, parameters such as detection probabilities, clutter rates, and motion model parameters are often unknown and time-varying. Such parameter variability can seriously degrade the performance of a multitarget tracking system. Here, we propose a Bayesian tracking framework in which the multisensor-multitarget tracking problem is formulated according to the measurement origin uncertainty paradigm and the unknown parameters-in the present case, the detection probabilities at the individual sensors-are modeled as Markov chains. The resulting Bayesian estimation problem is then solved using the belief propagation scheme. This approach results in a multisensor-multitarget tracking method that is able to adapt to the time variations of the detection probabilities. Moreover, the method has a low complexity that scales very well in all relevant system parameters. The performance of the method is assessed using data collected by a mobile underwater wireless sensor network.
This paper presents a dynamic adaptation algorithm of joint source-channel code rate for enhancing voice transmission over LTE network. In order to assess the speech quality, we use the Wideband (WB) E-model. In this ...
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ISBN:
(纸本)9781509024940
This paper presents a dynamic adaptation algorithm of joint source-channel code rate for enhancing voice transmission over LTE network. In order to assess the speech quality, we use the Wideband (WB) E-model. In this model, both end-to-end delay and packet loss are taken into account. The goal of this paper is to find out the best suboptimal solution for improving voice traffic over LTE network with some constraints on allowed maximum end-to-end delay, allowed maximum packet loss and minimum required bandwidth. The best suboptimal choice is channel code rate corresponding to each mode of the AMR-WB codec that minimizes redundant bits generated by channel coding with an acceptable MOS reduction. Besides, this algorithm can be integrated with rate control in AMR-WB codec to offer the required mode of LTE network. Our results show that the MOS degradation is not significant, but the percent of reduced redundant bits to be very considerable. This will requires less bandwidth, thus, more mobile users can be served. The algorithm has simple computational operations, it can be applied to real-time voice communications.
In this paper, we investigate active noise control over large 2D spatial regions when the noise source is sparsely distributed. The l(1) relaxation technique originated from compressive sensing is adopted and based on...
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ISBN:
(纸本)9781479999880
In this paper, we investigate active noise control over large 2D spatial regions when the noise source is sparsely distributed. The l(1) relaxation technique originated from compressive sensing is adopted and based on that we develop the algorithm for two cases: multipoint noise cancellation and wave domain noise cancellation. This results in two new variants (i) zero-attracting multi-point complex FxLMS and (ii) zero-attracting wave domain complex FxLMS. Both approaches use a feedback control system, where a microphone array is distributed over the boundary of the control region to measure the residual noise signals and a loudspeaker array is placed outside the microphone array to generate the anti-noise signals. Simulation results demonstrate the performance and advantages of the proposed methods in terms of convergence rate and spatial noise reduction levels.
In recent years, sun tracking systems are being increasingly developed to improve the efficiency of PV panels. Due to problems of tracking methods which uses optical sensors, new methods with less practical problems a...
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ISBN:
(纸本)9783642257339
In recent years, sun tracking systems are being increasingly developed to improve the efficiency of PV panels. Due to problems of tracking methods which uses optical sensors, new methods with less practical problems are developing day by day. In this article, we suggest a new method to reduce the problems which exist with sun tracking. In this method, instead of using optical sensor, an adaptive algorithm is used, where recursive equations are employed to calculate Azimuth and Elevation angles of sun. We applied this method on a two-axis sun tracker system with fixed panel on 45 system. Then calculate the efficiency of PV panels. Finally, by comparing the Results of this method with respect to method of optical sensors, it has been shown that the efficiency of this method is higher than methods which used optical sensors.
In speech enhancement algorithm based on generalized sidelobe canceller (GSC), when there is error in direction estimating, the target speech will not be blocked by blocking matrix (BM) module completely. Then in the ...
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ISBN:
(纸本)9781509006540
In speech enhancement algorithm based on generalized sidelobe canceller (GSC), when there is error in direction estimating, the target speech will not be blocked by blocking matrix (BM) module completely. Then in the later multiple-input canceller (MC) module, the target speech will be eliminated, which will cause the leakage of the target speech. In this paper, a new optimization algorithm is proposed for the leakage of the speech caused by the error of signal direction of arrival (DOA). The blocking matrix would be adjusted adaptively according to the characteristics of the correlation between the final output of GSC and the output of BM module. This way, the estimated direction can be closer to the real target speech direction in the blocking matrix in order to reduce the leakage of the target speech and the leakage of the target speech in multiple-input canceller will be reduced. The simulation results show that the proposed algorithm have better speech enhancement performances in both objective and subjective evaluations.
We consider the problem of proper learning of a boolean halfspace with integer weights {0, 1,..., t}, from membership queries only. The best known algorithm for this problem is an adaptive algorithm that asks n(0(t5))...
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ISBN:
(纸本)9783319116617
We consider the problem of proper learning of a boolean halfspace with integer weights {0, 1,..., t}, from membership queries only. The best known algorithm for this problem is an adaptive algorithm that asks n(0(t5)) membership queries, while the best lower bound for the number of membership queries is n(Omega(t)) [1]. In this paper we close this gap and give an adaptive proper learning algorithm with two rounds, and asking n(0(t)) membership queries. We also give a non-adaptive proper learning algorithm that asks n(0(t3)) membership queries. (C) 2016 Elsevier B.V. All rights reserved.
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