In ubiquitous computing, or ubiquitous network environment, many sensors and devices are embedded in the environment. Moreover, information appliances and smart space technologies allow users to enjoy high computational power, which also gives freedom and extrication from traditional computing style that often caused users' inconvenience and restriction of their demands and needs. However, due to the characteristics of ubiquitous computing, application's high adaptability to the changes of environments is required. To achieve this high adaptability of applications, we introduce a design approach of "snapshot-based application." The snapshot-based application can store its status as a snapshot. By utilizing the snapshot, an application can be recreated as it is in the status of the snapshot. In this paper, we propose an application framework named "u-Snap," which realizes snapshot-based applications. We have exemplified the effectiveness of this framework with implementation of "u-Photo," which utilize a snapshot of an application as contextual information that is embedded in a photo image.

This letter presents the design and implementation of a video streaming file server system, which has been implemented in the context of a distributed digital multimedia broadcasting environment that has been prototyped. To make a performance analysis of file systems and distributed object services for continuous media (CM) provisioning, we validate the performance of the system against that of a conventional file system, Unix file system, through an experimental evaluation.

Recently, wideband speech communication using 7 kHz-wideband speech coding, as described in ITU-T Recommendations G.722, G.722.1, and G.722.2, has become increasingly necessary for use in advanced IP telephony using PCs, since, for this application, hands-free communication using separate microphones and loudspeakers is indispensable, and in this situation wideband speech is particularly helpful in enhancing the naturalness of communication. An objective quality measurement methodology for wideband-speech coding has been studied, its essential components being an objective quality measure and an input test signal. This paper describes Wideband-PESQ conforming to the draft Annex to ITU-T Recommendation P.862, "Perceptual Evaluation of Speech Quality (PESQ)," as the objective quality measure, by evaluating the consistency between the subjectively evaluated MOS (Mean Opinion Score) and objectively estimated MOS. This paper also describes the verification of artificial voice conforming to Recommendation P.50 "Artificial Voices," as the input test signal for such measurements, by evaluating the consistency between the objectively estimated MOS using a real voice and that obtained using an artificial voice.

Live streaming media are delay sensitive and have limited allowable delays. Current conventional multicast protocols do not have a loss retransmission mechanism. Even though several reliable multicast protocols with retransmission mechanisms have been proposed, the long delay and high packet loss rate make them inefficient for live streaming. This paper proposes a multicast protocol focusing on the allowable delay called the QoS Multicast for Live Streaming (QMLS) protocol. QMLS routers are placed along the multicast tree to detect and retransmit lost packets. We propose a method that enables data recovery to be done immediately after lost packets are detected by the QMLS router and a method that reduces the unnecessary packets sent to end receivers. This paper discusses the mathematical analysis of the proposed protocol and compares it with other multicast protocols. The results reveal that our protocol is more effective in live streaming. Finally, we do a simulation to evaluate its performance and study the effect of consecutive losses. The simulation reveals that consecutive losses can slightly increase losses with our protocol.

The problem of estimating code timings in DS-CDMA systems with multiple antennas is considered in the presence of multipath time-varying fading channels and near-far environments. We present an efficient algorithm for an approximate maximum likelihood approach of jointly estimating the multipath timings of a desired user for DS-CDMA systems that consist of multiple antennas either uncorrelated or fully correlated in space. The procedures of the algorithm to estimate code-timings are developed in order to better exploit the time-varying characteristics of the fading process. In the multipath fading channels, the solution of the proposed algorithms is based on successively optimizing the criterion for increasing numbers of multipath delays. It is shown via simulation results that the modified approaches of the approximate maximum likelihood algorithm much more improve its acquisition performance in the time-varying fading channels. It is seen that the acquisition performance of multiple antennas based acquisition scheme is much better than that of a single antenna based timing estimator in the presence of multipath fading channels and the near-far problem. Furthermore, it is observed that the proposed algorithms outperform the correlator and MUSIC estimator in the multiuser environments with near-far situation on time-varying Rayleigh fading channels.

Recently, many algorithms have been proposed for fast full search motion estimation. Among them, successive elimination algorithm (SEA) and its modified algorithms significantly speed up the performance of the full search algorithm. By introducing the inequality equation between the norm and the mean absolute difference (MAD) of two matching blocks, the SEA can successively eliminate invalid candidate blocks without any loss in estimation accuracy. In this paper, we propose a partial norm based early rejection algorithm (PNERA) for fast block motion estimation. The proposed algorithm employs the sum of partial norms from several subblocks of the block. Applying the sum of partial norms to the inequality equation, we can significantly reduce the computational complexity of the full search algorithm. In an attempt to reduce the computational load further, the modified algorithms using partial norm distortion elimination (PNDE) and subsampling methods are also proposed. Experimental results show that the proposed algorithm is about 4 to 9 times faster than the original exhaustive full search, and is about 3 to 4 times faster than the SEA.

The present letter describes the estimation of the upper bounds of the correlation functions of a class of zero-correlation-zone sequences constructed from an arbitrary Hadamard matrix.

Limited energy is a big challenge for large scale wireless sensor networks (WSN). Previous research works show that modulation scaling is an efficient technique to reduce energy consumption. However, the impacts of using modulation scaling on packet delivery latency and loss are not considered, which may have adverse effects on the application qualities. In this paper, we study this problem and propose control schemes to minimize energy consumption while ensuring application qualities. We first analyze the relationships of modulation scaling and energy consumption, end-to-end delivery latency and packet loss ratio. With the analytical model, we develop a centralized control scheme to adaptively adjust the modulation levels, in order to minimize energy consumption and ensure the application qualities. To improve the scalability of the centralized control scheme, we also propose a distributed control scheme. In this scheme, the sink will send the differences between the required and measured application qualities to the sensors. The sensors will update their modulation levels with the local information and feedback from the sink. Experimental results show the effectiveness of energy saving and QoS guarantee of the control schemes. The control schemes can adapt efficiently to the time-varying requirements on application qualities.

In this paper, we propose a spatio-temporal equalizer for the space-time block coded transmission over the frequency selective fading channels with the presence of co-channel interference (CCI). The proposed equalizer, based on the tapped delay line adaptive array (TDLAA), performs signal equalization and CCI suppression simultaneously using the minimum mean square error (MMSE) method. It is to show that our scheme outperforms the previous two-stage combined adaptive antenna and delayed decision feedback sequence estimator (DDFSE) approach. We also show that performance can be further improved if the synchronization between the preceding and delayed paths is achieved.

In this report, we propose a tracking algorithm of speaker direction using microphones located at vertices of an equilateral triangle. The method realizes tracking by minimizing a performance index that consists of the cross spectra at three different microphone pairs in the triangular array. We adopt the steepest descent method to minimize it, and for guaranteeing global convergence to the correct direction with high accuracy, we alter the performance index during the adaptation depending on the convergence state. Through some computer simulation and experiments in a real acoustic environment, we show the effectiveness of the proposed method.

High-resolution spectrum estimation techniques have been extensively studied in recent publications. Knowledge of the noise variance is vital for spectrum estimation from noise-corrupted observations. This paper presents the use of noise compensation and data extrapolation for spectrum estimation. We assume that the observed data sequence can be represented by a set of autoregressive parameters. A recently proposed iterative algorithm is then used for noise variance estimation while autoregressive parameters are used for data extrapolation. We also present analytical results to show the exponential decay characteristics of the extrapolated samples and the frequency domain smoothing effect of data extrapolation. Some statistical results are also derived. The proposed noise-compensated data extrapolation approach is applied to both the autoregressive and FFT-based spectrum estimation methods. Finally, simulation results show the superiority of the method in terms of bias reduction and resolution improvement for sinusoids buried in noise.

This paper considers the problem of recursive filtering for linear discrete-time systems with uncertain observation. A new approximate adaptive filter with a parallel structure is herein proposed. It is based on the optimal mean square combination of arbitrary number of correlated estimates which is also derived. The equation for error covariance characterizing the mean-square accuracy of the new filter is derived. In consequence of parallel structure of the filtering equations the parallel computers can be used for their design. It is shown that this filter is very effective for multisensor systems containing different types of sensors. A practical implementation issue to consider this filter is also addressed. Example demonstrates the accuracy and efficiency of the proposed filter.

This paper proposes a novel multi-layer approach to fundamental frequency modeling for concatenative speech synthesis based on a statistical learning technique called additive models. We define an additive F0 contour model consisting of long-term, intonational phrase-level, component and short-term, accentual phrase-level, component, along with a least-squares error criterion that includes a regularization term. A backfitting algorithm, that is derived from this error criterion, estimates both components simultaneously by iteratively applying cubic spline smoothers. When this method is applied to a 7,000 utterance Japanese speech corpus, it achieves F0 RMS errors of 28.9 and 29.8 Hz on the training and test data, respectively, with corresponding correlation coefficients of 0.806 and 0.777. The automatically determined intonational and accentual phrase components turn out to behave smoothly, systematically, and intuitively under a variety of prosodic conditions.

In this paper, we propose a multiple-input multiple-output (MIMO) beamforming scheme for a multiuser system in frequency-selective fading channels. The maximum signal-to-noise and interference ratio (MSINR) is adopted as a criterion to determine the transmit and receive weight vectors. In order to maximize the output SINR over all users, two algorithms for base station are considered: the first algorithm is based on the receive weight vector optimization and the second algorithm is based on an iterative update of both transmit and receive weight vectors. Based on the result of single user MIMO beamforming, we analyze the interference channels cancellation ability of multiuser MIMO system. The first algorithm is a simple method and the second algorithm is a performative solution. Through computer simulations, it is shown that multiuser communication system is achievable using the proposed methods in frequency-selective fading condition.

In this paper, we propose an advanced location-based service that we call a direction-based service, which utilizes both the position and direction of a user. The direction-based service enables a user to point to an object of interest for command or investigation. We also describe the design, implementation and evaluations of a direction-based service system named Azim. With this system, the direction of the user can be obtained by a magnetic-based direction sensor. The sensor is also used for azimuth-based position estimation, in which a user's position is estimated by having the user point to and measure azimuths of several markers or objects whose positions are already known. Because this approach does not require any other accurate position sensors or positive beacons, it can be deployed cost-effectively. Also, because the measurements are naturally associated with some degree of error, the position is calculated as a probability distribution. The calculation considers the error of direction measurement and the pre-obtained field information such as obstacles and magnetic field disturbance, which enables robust position measurements even in geomagnetically disturbed environments. For wide-area use, the system also utilizes a wireless LAN to obtain rough position information by identifying base stations. We have implemented a prototype system for the proposed method and some applications for the direction-based services. Furthermore, we have conducted experiments both indoors and outdoors, and exemplified that positioning accuracy by the proposed method is precise enough for a direction-based service.

A flooding algorithm is an indispensable and fundamental network control mechanism for achieving some tasks, such notifying all nodes of some information, transferring data with high reliability, getting some information from all nodes, or to reserve a route by flooding the messages in the network. In particular, the flooding algorithm is greatly effective in the heterogeneous and dynamic network environment such as so-called ubiquitous networks, whose topology is indefinite or changes dynamically and whose nodal function may be simple and less intelligent. Actually, it is applied to grasp the network topology in a sensor network or an ad-hoc network, or to retrieve content information by mobile agent systems. A flooding algorithm has the advantages of robustness and optimality by parallel processing of messages. However, the flooding mechanism has a fundamental disadvantages: it causes the message congestion in the network, and eventually increases the processing time until the flooding control is finished. In this paper, we propose and evaluate methods for producing a more efficient flooding algorithm by adopting the growth processes of primitive creatures, such as molds or microbes.

This paper presents a multiple constrained subspace based multiuser detector for synchronous long-code downlink multirate DS-CDMA systems. The novel receiver adapts its fractionally-spaced equaliser tap-weights based upon two modes, namely training and decision-directed modes. Switching between two modes is achieved by changing the code constraint in the associated subspace algorithm. Moreover, detection of the desired user requires the knowledge of the desired user's spreading code only. Simulation results show that the proposed receiver is capable of multiple access interference (MAI) suppression and multipath mitigation. Besides, the results reveal the improvement in terms of convergence speed and mean square error (MSE) of the proposed receiver over the existing receiver in both static and dynamic environments.

This paper describes recent progress in research on wavelength converters that employ quasi-phase-matched LiNbO3 (QPM-LN) waveguides. The basic structure and operating principle of these devices are presented. The conversion efficiency in difference frequency generation (DFG), second harmonic generation (SHG) and an SHG/DFG cascade scheme are explained. Device fabrication technologies such as periodic poling, and those used for annealed proton-exchanged (APE) waveguides, and direct bonded waveguides are introduced. An APE waveguide is used to demonstrate the wavelength conversion of broadband (> 1 Tbit/s) WDM signals. The low penalty conversion of high-speed (40 Gbit/s) based WDM signals is also reported. Excellent resistance to photorefractive damage in a direct bonded waveguide is presented. This high level of resistance enabled highly efficient wavelength conversion. A new design concept is introduced for a multiple QPM device based on the continuous phase modulation of a periodically poled structure. This multiple QPM device enables the variable wavelength conversion of WDM signals. High-speed wavelength switching between ITU-T grid wavelengths using a finely tuned multiple QPM device is also reported. QPM-LN based wavelength converters have several advantages, including the ability to convert high-speed signals of 1 THz or greater, no signal-to-noise (S/N) ratio degradation, no modulation format dependence, and they are capable of the simultaneous conversion of broadband WDM channels. They will therefore be key devices in future photonic networks.

A numerical technique for improving the accuracy of a rectangular loop antenna based on the Finite Difference Time Domain (FDTD) method is proposed. In this technique, a quasi-static field behaviour is incorporated into the FDTD update equations, and the more accurate numerical technique can be obtained without the need of using fine cells. The simulation results of this proposed technique are compared with the Method of Moment to confirm the effectiveness of the technique.

We have fabricated a multilayer structure for single flux quantum (SFQ) circuit application using a high-temperature superconductor (HTS). La0.2-Y0.9Ba1.9Cu3Ox (La-YBCO) base electrode layers were prepared by a dc or rf magnetron sputtering method. The reproducibility of film quality for dc-sputtered La-YBCO films was better than that for rf-sputtered films, and the dc sputtered films exhibited the average surface roughness Ra less than 1.0 nm and a Tc zero value of 88 K. By using the dc-sputtered La-YBCO films, a multilayer structure of SrSnO3/La-YBCO/SrSnO3/La-YBCO on MgO substrate with Ra below 2.0 nm was obtained. Interface-modified ramp-edge junctions with La0.2-Yb0.9Ba1.9Cu3Ox (La-YbBCO) counter electrodes have been fabricated by using this multilayer structure with dc-sputtered films. The fabricated junctions exhibited RSJ-type I-V characteristics with IcRn products of about 3 mV at 4.2 K. We also obtained a 1-σ Ic spread of 8% for a 1000-junction series-array. The sheet inductance values at 4.2 K for the base and counter electrodes on La-YBCO ground planes were 0.8 pH and 0.7 pH per square, respectively. Operation of several types of elementary SFQ circuits has been successfully demonstrated by using this multilayer structure.