Paralleled converter system with synchronous rectifiers (SRs) causes several problems such as surge voltage, inhalation current and circulating current. Generally, the system stops operation of the SRs in light load to avoid these problems. However, simultaneously, large voltage fluctuations in the output of the modules are occurred due to forward voltage drop of diode. The fluctuations cause serious faults to the semiconductor devices working in very low voltage such as CPU and VLSI. Moreover, the voltage fluctuations generate unstable current fluctuations in the paralleled converter system with current-sharing control. This paper proposes new switching control methods for rectifiers to reduce the voltage and current fluctuations. The effectiveness of the proposed methods is confirmed by computer simulation and experimental results.

We report tunable optical devices based on fiber Bragg gratings (FBGs), whose filtering characteristics are controlled by strain distributions. These devices include a widely wavelength tunable filter, a tunable group-velocity dispersion (GVD) compensator, a tunable dispersion slope (DS) compensator, and a variable-bandwidth optical add/drop multiplexer (OADM), which will play important roles for next-generation reconfigurable optical networks.

A hybrid beamformer composed of a direction-of-arrival (DOA) based scheme and maximal ratio combining (MRC) is proposed to overcome the degradation caused by inaccurate channel estimation due to insufficient pilot power, which happens in conventional single-input, multiple-output (SIMO) Code Division Multiple Access (CDMA) reverse link. The proposed scheme can provide more accurate channel estimation and interference reduction at the expense of diversity gain in the spatially correlated SIMO channel. As a result, the hybrid scheme outperforms conventional MRC beamformers for six or more antennas in the channel environment, in which the angle-of-spread (AOS) is within 30.

Asynchronous optical packet switching (OPS) is a promising solution to support the continuous growth of transmission capacity demand. It has been, however, quite difficult to implement key functions needed at the node of such networks with all-optical approaches. We have proposed a new optoelectronic system composed of a packet-by-packet optical clock-pulse generator (OCG), an all-optical serial-to-parallel converter (SPC), a photonic parallel-to-serial converter (PSC), and CMOS circuitry. The system makes it possible to carry out various required functions such as buffering (random access memory), optical packet compression/decompression, and optical label swapping for high-speed asynchronous optical packets.

We propose an aperture coupling E-bend with step structure as a robust post-wall waveguide interface to an external standard waveguide in which the posts are separated from the aperture. A 61.25 GHz model transformer gives 11.4% bandwidth for the reflection below -25 dB by using a 1.2 mm-height dielectric substrate. We have compared the mechanical tolerances of both the conventional straight structure and the proposed step structure. The step structure is more robust against mechanical errors than the conventional one. Another advantage is that the accuracy of the analysis model for the post-wall to metal-wall replacement is enhanced for the step structure.

For future generation mobile networks, we expect that the mobile devices like PDAs, note PCs or any VoIP-enabled communicators will have the feature of being always switched on, ready for service, constantly reachable by the wireless Internet. In addition to high access speed, attractive real-time contents or other expected spectacular features of the future wireless Internet environment, the mobile terminals has to be very much energy-aware to enable literal untethered movement of the user. Mechanisms for network activities like maintaining location information and wireless system discovery, which require regular network access, should be energy-efficient and resource-efficient in general. Cellular systems employ the notion of passive connectivity to reduce the power consumption of idle mobile hosts. In IP based Multi-service User Terminal (MUT) that may have multiple wireless interfaces for receiving various classes of services from the network, there should be an efficient addressing of the energy consumption issue. To devise an energy-efficient scheme for simultaneous or single operation of the wireless interfaces attached to such terminals we should have comprehensive understanding of the power consumption of the devices/modules in various operational states. This paper investigates the power consumption pattern or behavior of some selected wireless interfaces that are good candidates for being part of the future of the multi-service user terminals. We propose a simple model for predicting energy consumption in a terminal attributed to the wireless network interfaces. We measured the actual consumption pattern to estimate the parameters of the model.

A novel structure of guided-wave electro-optic modulators is proposed and their operation of the optical intensity modulation is successfully confirmed. The modulators use a newly developed modulation electrode consisting of coupled microstrip lines. A high voltage can be induced by the resonance of the odd propagation mode to realize efficient electro-optic modulation in spite of the microstrip-line configuration. The properties of the coupled microstrip lines were analytically evaluated by using the method of conformal transformations and the effectiveness as a modulation electrode was presented. The modulation electrodes of the half-wavelength coupled microstrip line resonator were designed at 10 GHz and 26 GHz, where the electrode lengths were 3 mm and 1.2 mm, respectively. The modulators using these electrodes with the Mach-Zehnder interferometer of LiTaO3 waveguides were fabricated. The measured modulation coefficient of the 26 GHz modulator normalized at 100 mW signal input was 0.13 rad.

This paper describes the special demands of conversational speech in the context of corpus-based speech synthesis. The author proposed the CHATR system of prosody-based unit-selection for concatenative waveform synthesis seven years ago, and now extends this work to incorporate the results of an analysis of five-years of recordings of spontaneous conversational speeech in a wide range of actual daily-life situations. The paper proposes that the expresion of affect (often translated as 'kansei' in Japanese) is the main factor differentiating laboratory speech from real-world conversational speech, and presents a framework for the specification of affect through differences in speaking style and voice quality. Having an enormous corpus of speech samples available for concatenation allows the selection of complete phrase-sized utterance segments, and changes the focus of unit selection from segmental or phonetic continuity to one of prosodic and discoursal appropriateness instead. Samples of the resulting large-corpus-based synthesis can be heard at http://feast.his.atr.jp/AESOP.

System-level fault diagnosis deals with the problem of identifying faulty nodes (processors) in a multiprocessor system. Each node is faulty or fault-free, and it can test other nodes in the system, and outputs the test results. The test result from a node is reliable if the node is fault-free, but the result is unreliable if it is faulty. In this paper, we prove that four variants of the hypercube: the crossed cube, the twisted cube, the Mobius cube, and the enhanced cube, are adaptively diagnosed using at most 4 parallel testing rounds, with at most n faulty nodes (for the enhanced cube, with at most n + 1 faulty nodes), where each processor participates in at most one test in each round. Furthermore, we propose another diagnosis algorithm for the n-dimensional enhanced cube with at most n + 1 faulty nodes, and show that it is adaptively diagnosed with at most 5 rounds in the worst case, but with at most 3 rounds if the number of existing faulty nodes is at most n -log(n + 1).

This paper presents an optimal load balancing algorithm based on both of the ANFIS (Adaptive Neuro-Fuzzy Inference System) modeling and the FIS (Fuzzy Inference System) for the local status of real servers. It also shows the substantial benefits such as the removal of load-scheduling overhead, QoS (Quality of Service) provisioning and providing highly available servers, provided by the suggested method.

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.

This paper proposes the hierarchical cloud-router network (HCRN) to overcome the scalability limit in a multi-layer generalized multi-protocol label switching (GMPLS) network. We define a group of nodes as a virtual node, called the cloud-router (CR). A CR consists of several nodes or lower-level CRs. A CR is modeled as a multiple switching capability (SC) node when it includes more than one kind of SC, which is fiber SC, lambda SC, time-division multiplexing (TDM) SC, packet SC, even if there are no actual multiple switching capability nodes in the CR. The CR advertises its abstracted CR internal structure, which is abstracted link state information inside the CR. A large-scale, multi-layer network can then achieve scalability by advertising the CR internal structure throughout the whole network. In this scheme, the ends of a link connecting two CRs are defined as interfaces of the CRs. We adopt the CR internal cost scheme between CR interfaces to abstract the network. This CR internal cost is advertised outside the CR via the interfaces. Our performance evaluation has shown that HCRN can handle a larger number of nodes than a normal GMPLS network. It can also bear more frequent network topology changes than a normal GMPLS network.

Holey Fiber (HF) technology has progressed rapidly in recent years and has resulted in the development of a wide range of optical fibers with unique and highly useful optical properties including endlessly single-mode guidance, and high optical nonlinearity. In this paper the state-of-the-art HF technology for all-optical signal processing devices is reviewed from a perspective of possible application for telecommunications.

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.

In this paper, we investigate packet loss and system dimensioning of feedback (FB) type wavelength division multiplexing (WDM) optical routers under asynchronous and variable packet length self-similar traffic. We first study the packet loss performance for two different types of WDM optical routers under asynchronous and variable packet length self-similar traffic. Based on simulation results, we demonstrate that a 1616 FB type WDM optical router employing more than 4 re-circulated ports without using void filling (VF) algorithm has better performance. We then present the system dimensioning issues of FB type WDM optical routers, by showing the performance of FB type WDM optical routers as a function of the number of re-circulated ports, buffer depth, re-circulation limit, basic delay unit in the fiber delay line optical buffers and traffic characteristics. The sensitivity of the mutual effects of the above parameters on packet loss is investigated in details. Based on our results, we conclude that the FB type WDM optical routers must be dimensioned with the appropriate number of re-circulated ports, re-circulation limits, buffer depth, and optimal basic delay unit in the fiber delay line optical buffers under relevant traffic characteristics to achieve high switching performance.

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.

In this paper we propose an objective method for assessing the capability of a speech translation system. It automates the translation paired comparison method, which gives a simple, easy to understand TOEIC score proposed by Sugaya et al., to succinctly evaluate a speech translation system. To avoid the expensive evaluation cost of the original method where large manual effort is required, the new objective method automates the procedure by employing an objective metric such as BLEU and DP-based measure. The evaluation results obtained by the proposed method are similar to those of the original method. Also, the proposed method is used to evaluate the usefulness of a speech translation system. It is then found that our speech translation system is useful in general, even to users with higher TOEIC score than the system's.

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 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.

We present a new speech enhancement algorithm in a car environment with two microphones. The car audio signals and other background noises are the target noises to be suppressed. Our algorithm is composed of two main parts, i.e., the spatial and the temporal processes. The multi-channel blind deconvolution (MBD) is applied to the spatial process while the Kalman filter with a second-order high pass filter, for the temporal one. For the fast convergence, the MBD is newly expressed in frequency-domain with a normalization matrix. The final performance evaluated with the severely car noise corrupted speech shows that our algorithm produces noticeably enhanced speech.