In this manuscript we present a methodology for reducing the impact of the hosting room reflections in sound field rendering applications based on loudspeaker arrays. The problem is formulated in a least-squares sense. Since matrices involved in the problem are ill-conditioned, it is important to devise a suitable technique for the regularisation of the pseudo-inverse. In this work we adopt a truncated SVD method. The truncation, in particular, aims at reducing the impact of numerical errors and also errors on the knowledge of the sound speed. We include a wide set of experimental results, which validate the proposed technique.

In this paper, we study the feasibility of a batteryless wireless sensor supplied with energy by using microwave power transmission (MPT). If we perform co-channel operation of MPT and wireless local area networks (WLANs) for the sake of spectral efficiency, a time division method for MPT and WLAN communications is required to avoid serious interference from MPT to WLAN data transmissions. In addition, to reduce the power consumption of a sensor, the use of power-save operation of the sensor is desirable. We proposed a scheduling scheme that allocates time for MPT and WLAN communications. Specifically, in the proposed scheduling system, an energy source transmits microwave power to a sensor station except when the sensor station transmits data frames or receives beacon frames. In addition, in the proposed scheduling system, we force the remaining energy of the sensor station to converge to a maximum value by adjusting the time interval of data transmission from the sensor station such that the power consumption of the sensor station is reduced. On the basis of the proposition, we implemented a scheduling system and then confirmed that it performed successfully in the conducted experiments. Finally, we discussed the feasibility of the proposed scheduling scheme by evaluating the coverage and then showed that the scheduling scheme can be applied to closed space or room.

This paper presents a method to seamlessly extend the coverage of energy supply field for wireless sensor networks in order to free sensors from wires and batteries, where the multi-point scheme is employed to overcome path-loss attenuation, while the carrier shift diversity is introduced to mitigate the effect of interference between multiple wave sources. As we focus on the energy transmission part, sensor or communication schemes are out of scope of this paper. To verify the effectiveness of the proposed wireless energy transmission, this paper conducts indoor experiments in which we compare the power distribution and the coverage performance of different energy transmission schemes including conventional single-point, simple multi-point and our proposed multi-point scheme. To easily observe the effect of the standing-wave caused by multipath and interference between multiple wave sources, 3D measurements are performed in an empty room. The results of our experiments together with those of a simulation that assumes a similar antenna setting in free space environment show that the coverage of single-point and multi-point wireless energy transmission without carrier shift diversity are limited by path-loss, standing-wave created by multipath and interference between multiple wave sources. On the other hand, the proposed scheme can overcome power attenuation due to the path-loss as well as the effect of standing-wave created by multipath and interference between multiple wave sources.

In this paper, we propose a joint transmit and receive linear filtering based on minimum mean square error criterion (joint Tx/Rx MMSE filtering) for single-carrier (SC) multiple-input multiple-output (MIMO) transmission. Joint Tx/Rx MMSE filtering transforms the MIMO channel to the orthogonal eigenmodes to avoid the inter-antenna interference (IAI) and performs MMSE based transmit power allocation to sufficiently suppress the inter-symbol interference (ISI) resulting from the severe frequency-selectivity of the channel. Rank adaptation and adaptive modulation are jointly introduced to narrow the gap of received signal-to-interference plus noise power ratio (SINR) among eigenmodes. The superiority of the SC-MIMO transmission with joint Tx/Rx MMSE filtering and joint rank adaptation/adaptive modulation is confirmed by computer simulation.

In the traditional microphone array signal processing, the performance degrades rapidly when the array aperture decreases, which has been a barrier restricting its implementation in the small-scale acoustic system such as digital hearing aids. In this work a new compressed sampling method of miniature microphone array is proposed, which compresses information in the internal of ADC by means of mixture system of hardware circuit and software program in order to remove the redundancy of the different array element signals. The architecture of the method is developed using the Verilog language and has already been tested in the FPGA chip. Experiments of compressed sampling and reconstruction show the successful sparseness and reconstruction for speech sources. Owing to having avoided singularity problem of the correlation matrix of the miniature microphone array, when used in the direction of arrival (DOA) estimation in digital hearing aids, the proposed method has the advantage of higher resolution compared with the traditional GCC and MUSIC algorithms.

To clarify the dependence of arc duration on atmosphere, experiments were conducted under conditions of air, N$_{2}$, Ar, He and CO$_{2}$ with the pressure of 0.1,MPa in a 14,V/28,V/42,V circuit respectively. A quantitative relationship between arc duration and gas parameters such as ionization potential, thermal conductivity was obtained from the experimental data. Besides, the inherent mechanism of influence of atmosphere on arc duration was discussed.

We analyze the outage probability of the multiuser two-way relay network (TWRN) where the N-th best mobile user (MU) out of M MUs and the base station (BS) exchange messages with the aid of an amplify-and-forward relay. In the analysis, we focus on the practical unbalanced Nakagami-m fading between the MUs-relay link and the relay-BS link. We also consider both perfect and outdated channel state information (CSI) between the MUs and the relay. We first derive tight closed-form lower bounds on the outage probability. We then derive compact expressions for the asymptotic outage probability to explicitly characterize the network performance in the high signal-to-noise ratio regime. Based on our asymptotic results, we demonstrate that the diversity order is determined by both Nakagami-m fading parameters, M, and N when perfect CSI is available. When outdated CSI is available, the diversity order is determined by Nakagami-m fading parameters only. In addition, we quantify the contributions of M, N, and the outdated CSI to the outage probability via the array gain.

In this paper, a new database suitable for HMM-based automatic Filipino speech recognition is described for the purpose of training a domain-independent, large-vocabulary continuous speech recognition system. Although it is known that high-performance speech recognition systems depend on a superior speech database used in the training stage, due to the lack of such an appropriate database, previous reports on Filipino speech recognition had to contend with serious data sparsity issues. In this paper we alleviate such sparsity through appropriate data analysis that makes the evaluation results more reliable. The best system is identified through its low word-error rate to a cross-validation set containing almost three hours of unknown speech data. Language-dependent problems are discussed, and their impact on accuracy was analyzed. The approach is currently data driven, however it serves as a competent baseline model for succeeding future developments.

It is known that the IP Multimedia Subsystem (IMS) provides various telecom services e.g., VoIP, instant messaging, and video conferencing. In the IMS, these services are provided with session initiation protocol (SIP) handled by call/session control function servers (CSCFs). Completing the SIP signaling call flow without delay is vital to provide satisfactory services to the users. For service continuity, previous studies redundantly synchronized session states of CSCFs with backup servers. This paper proposes an IMS restoration system that selectively stores the session states. This is achieved by utilizing the retransmission mechanism of SIP. Time-based simulation emulating the process of backup servers shows that the proposed system reduces the number of backup servers to less than 38% compared to the previous studies, without degrading the service quality.

Differential games are considered an extension of optimal control problems, which are used to formulate centralized control problems in smart grids. Optimal control theory is used to study systems consisting of one agent with one objective, whereas differential games are used to formulate systems consisting of multiple agents with multiple objectives. Therefore, a differential-game-theoretic approach is appropriate for formulating decentralized demand-side energy management systems where there are multiple decision-making entities interacting with each other. Moreover, in many smart grid applications, we need to obtain information for control via communication systems. To formulate the influence of communication availability, differential game theory is also promising because the availability of communication is considered as part of an information structure (i.e., feedback or open-loop) in differential games. The feedback information structure is adopted when information for control can be obtained, whereas the open-loop information structure is applied when the information cannot be obtained because of communication failure. This paper proposes a comprehensive framework for evaluating the performance of demand-side actors in a demand-side management system using each control scheme according to both communication availability and sampling frequency. Numerical analysis shows that the proposed comprehensive framework allows for an analysis of trade-off for decentralized and centralized control schemes.

The aim of this study is to present electrooculogram signals that can be used for human computer interface efficiently. Establishing an efficient alternative channel for communication without overt speech and hand movements is important to increase the quality of life for patients suffering from Amyotrophic Lateral Sclerosis or other illnesses that prevent correct limb and facial muscular responses. In this paper, we introduce the gaze estimation system of electrooculogram signals. Using this system, the electrooculogram signals can be recorded when the patients focused on each direct. All these recorded signals could be analyzed using math-method and the mathematical model will be set up. Gaze estimation can be recognized using electrooculogram signals follow these models.

This paper presents a novel filter to keep from over-smoothing the edges and corners and rectify the outliers in the flow field after each incremental computation step, which plays a key role during the process of estimating flow field. This filter works according to the spatial-temporal derivatives distance of the input image and velocity field distance, whose principle is more reasonable in filtering mechanism for optical flow than other existing nonlinear filters. Moreover, we regard the spatial-temporal derivatives as new powerful descriptions of different motion layers or regions and give a detailed explanation. Experimental results show that our proposed method achieves better performance.

A new real-time sound field transmission system is presented. To construct this system, a large listening area needs to be reproduced at not less than a constant height. Additionally, the driving signals of the loudspeakers should be obtained only from received signals of microphones. Wave field reconstruction (WFR) filtering for linear arrays of microphones and loudspeakers is considered to be suitable for this kind of system. An experimental system was developed to show the feasibility of real-time sound field transmission using the WFR filter. Experiments to measure the reproduced sound field and a subjective listening test of sound localization were conducted to evaluate the proposed system. Although the reproduced sound field included several artifacts such as spatial aliasing and faster amplitude decay, the experimental results indicated that the proposed system was able to provide sound localization accuracy for virtual sound sources comparable to that for real sound sources in a large listening area.

For two-dimensional IIR digital filters described by the Fornasini-Marchesini second model, the problem of jointly optimizing high-order error feedback and realization to minimize the effects of roundoff noise at the filter output subject to l2-scaling constraints is investigated. The problem at hand is converted into an unconstrained optimization problem by using linear-algebraic techniques. The unconstrained optimization problem is then solved iteratively by applying an efficient quasi-Newton algorithm with closed-form formulas for key gradient evaluation. Finally, a numerical example is presented to illustrate the validity and effectiveness of the proposed technique.

Wide area virtualization of wireless transceivers by centrally managed software radio systems is a way to efficiently share the resources for supporting a variety of wireless protocols. In order to enable wide-area virtualization of wireless transceivers, the authors have developed a mechanism to deliver the radio space information which is quantized broadband radio wave information including the radio signals to the transceivers. Delivery mechanism consists of a distribution server which distributes radio space corresponding to the request of the client such as the center frequency and the bandwidth and a client which uses the radio space information. Accumulation of the distribution servers which deliver radio space information simultaneously to a large number of clients will contribute to build an infrastructure for any clients ubiquitously distributed over the globe. In this paper, scale-out architecture of a distribution server is proposed to deliver unlimitedly broadband radio space information to unlimited number of clients. Experimental implementation indicates the architecture to be a scale-out solution, while the number of clients is restricted by the computer resources of the distribution server. The band pass filter processing for individual client in the distribution server consumes the dominant part of the processing power, and the number of CPU cores is the upper limit of clients supportable for the distribution server in the current operating system implementation. The logical increase of the number of CPU cores by hardware multithreading does not contribute to relax this limit. We also discuss the guidance architecture or building server derived from these conclusions.

In a sector of a single cell, due to the fading characteristic of wireless channels, several decode-and-forward relay stations are deployed to form a two-hop relay-assisted multicast system. We propose two schemes for the system, the first scheme combines the use of space-time code and distributed space-time code (DSTC), and the second one combines the use of DSTC and maximum ratio combining. We give an outage probability analysis for both of them. Based on this analysis, we manage to maximize the spectral efficiency under a preset outage probability confinement by finding out the optimal power allocation and relay location strategies. We use genetic algorithms to verify our analysis and numerical results show that the schemes proposed by us significantly outperform the scheme in previous work. We also show the effect of path loss exponent on the optimal strategy.

We propose a new algorithm to suppress both stationary background noise and nonstationary directional interference noise in a speech enhancement system that employs the generalized sidelobe canceller. Our approach builds on advances in generalized sidelobe canceller design involving the transfer function ratio. Our system is composed of three stages. The first stage estimates the transfer function ratio on the acoustic path, from the nonstationary directional interference noise source to the microphones, and the powers of the stationary background noise components. Secondly, the estimated powers of the stationary background noise components are used to execute spectral subtraction with respect to input signals. Finally, the estimated transfer function ratio is used for speech enhancement on the primary channel, and an adaptive filter reduces the residual correlated noise components of the signal. These algorithmic improvements give consistently better performance than the transfer function generalized sidelobe canceller when input signal-to-noise ratio is 10 dB or lower.

When ultra-high-frequency (UHF) method is applied in partial discharge (PD) detection for GIS, the propagation process and rules of electromagnetic (EM) wave need to be understood clearly for conducting diagnosis and assessment about the real insulation status. The preceding researches are mainly concerning about the radial component of the UHF signal, but the propagation of the signal components in axial and radial directions and that perpendicular to the radial direction of the GIS tank are rarely considered. So in this paper, for a 252,kV GIS with T-shaped structure (TS), the propagation and attenuation of PD-induced EM wave in different circumferential angles and directions are investigated profoundly in time and frequency domain based on Finite Difference Time Domain (FDTD) method. The attenuation rules of the peak to peak value (Vpp) and cumulative energy are concluded. By comparing the results of straight branch and T branch, the influence of T-shaped structure over the propagation of different signal components are summarized. Moreover, the new circumferential and axial location methods proposed in the previous work are verified to be still applicable. This paper discusses the propagation mechanism of UHF signal in T-shaped tank, which provides some referential significance towards the utilization of UHF technique and better implementation of PD detection.

In this letter, we propose an improved single image haze removal algorithm using image segmentation. It can effectively resolve two common problems of conventional algorithms which are based on dark channel prior: halo artifact and wrong estimation of atmospheric light. The process flow of our algorithm is as follows. First, the input hazy image is over-segmented. Then, the segmentation results are used for improving the conventional dark channel computation which uses fixed local patches. Also, the segmentation results are used for accurately estimating the atmospheric light. Finally, from the improved dark channel and atmospheric light, an accurate transmission map is computed allowing us to recover a high quality haze-free image.

Registration consistency (RC) stands out as a widely-used automatic measure from existing image registration evaluation measures. However the original RC neglects the influence brought by the image intensity variation, leading to several problems. This letter proposes a rectified registration consistency, which takes both image intensity variation and geometrical transformation into consideration. Therefore the geometrical transformation is evaluated more by decreasing the influence of intensity variation. Experiments on real image pairs demonstrated the superiority of the proposed measure over the original RC.