Providing workflow function is one of the most important research issues in the next generation Internet services such as Web Service and Grid Computing. Scalability for Internet scale services, reliability for unstable Internet resources, and management functions of workflow systems are the essential requirements in these environments. However, existing workflow enactment models for enterprises could not meet these requirements. This paper proposes the PeerFlow that is a P2P based workflow enactment model, to provide workflow functions for the next generation Internet services. To apply P2P model to the workflow enactment model, we introduce the concept of the instance buddy and the index data of workflow instances, then propose the principle architecture of the PeerFlow. The instance buddy enables the autonomous processing of peers, and it is used for recovery and monitoring functions. This paper also presents the recovery capabilities of PeerFlow with formal proofs for the reliability issues and a performance evaluation with SimPy, the Python simulation package.

The number of customers of a service for Internet access from cellular phones in Japan has been explosively increasing for some time. We analyze the relation between the number of customers and the volume of traffic, with a view to finding clues to the structure of human relations among the very large set of potential customers of the service. The traffic data reveals that this structure is a scale-free network, and we calculate the exponent that governs the distribution of node degree in this network. The data also indicates that people who have many friends tend to subscribe to the service at an earlier stage. These results are useful for investigating various fields, including marketing strategies, the propagation of rumors, the spread of computer viruses, and so on.

This paper describes a new speech enhancement system that employs a microphone array with post-processing based on minimum mean-square error short-time spectral amplitude (MMSE-STSA) estimator. To get more accurate MMSE-STSA estimator in a microphone array, modification and refinement procedure are carried out from each microphone output. Performance of the proposed system is compared with that of other methods using a microphone array. Noise removal experiments for white and pink noises demonstrate the superiority of the proposed speech enhancement system to others with a microphone array in average output SNRs and cepstral distance measures.

This paper presents an efficient LSP quantizer implementation for low bit-rate coders. The major feature of the quantizer is that it uses a truncated cepstral distance criterion for the code selection procedure. This approach has generally been considered too computationally costly. We utilized the quantizer with a moving-average predictor, two-stage-split vector quantizer and delayed decision. We have investigated the optimal parameter settings in this case and incorporated the quantizer thus obtained into an ITU-T 4-kbit/s speech coding candidate algorithm with a bit budget of 21 bits. The objective performance is better than that with a conventional weighted mean-square criterion, while the complexity is still kept to a reasonable level. The paper also describes the codebook design and techniques that were employed to achieve robustness in noisy channel conditions.

In this paper, the effects of visual information on associated auditory information were investigated when presented simultaneously under dynamic conditions on a wide screen. Experiments of an auditory-visual stimulus presentation using a computer graphics movie of a moving patrol car and its siren sound, which were combined in various locations, were performed in 19 subjects. The experimental results showed the following: the visual stimulus at the beginning of the presentation captured the sound image stronger than that at the end (i.e., beginning effect), the sound image separated from the visual image even when both stimulus locations were exactly at the same place and then when both stimuli moved in opposite directions from each other, the visual stimulus tended to capture the sound image stronger in the peripheral visual field than in the central visual field, and the visual stimulus moving toward the sound source captured the sound image stronger than that moving away from the sound source.

Recently, the small-world network model has been popular to describe a wide range of networks such as human social relations and networks formed by biological entities. The network model achieves a small diameter with relatively few links as measured by the ratio of clustering coefficient and the number of links. It is quite natural to consider email communication similar to social network patterns. Quite surprisingly, we find from our empirical study that local email networks follow a different type of network model that falls into the category of scale-free network. We propose new network models to describe such communication structure.

The goal of source localization in the brain is to estimate a set of parameters for representing source characteristics; one of such parameters is the source number. We here propose a method combining the Powell algorithm with the information criterion method for determining the optimal dipole number. The potential errors can be calculated by the Powell algorithm with the concentric 4-sphere head model and 32 electrodes, then the number of dipoles is determined by the information criterion method with the potential errors mentioned above. This method has the advantages of a high identification accuracy of dipole number and a small number of EEG data because in this method: (1) only one EEG topography is used in the computation, (2) 32 electrodes are used to obtain the EEG data, (3) the optimal dipole number can be obtained by this method. In order to prove our method to be efficient, precise and robust to noise, 10% white noise is introduced to test this method theoretically. Some investigations are presented here to show our method is an advanced approach for determining the optimal dipole number.

This paper investigates the relations between IP network performances and the speech quality of the Voice over IP (VoIP) service through extensive experiments on a test bed network. The aim is to establish an effective and practical methodology for telecommunications operators to manage the quality of VoIP service via the management of IP network performances under their control. As IP network performances, utilization of the bottleneck link in the test bed and the following statistical factors of VoIP packets are examined: the standard deviation of delay variations (jitters), the standard deviation of packet interarrival times, and the packet loss ratio. On the other hand, VoIP speech quality is monitored as the Perceptual Evaluation of Speech Quality (PESQ). To investigate the relations under various network conditions, the experiments are performed by varying the following network related parameters of the test bed: the bandwidth of the bottleneck link, the size of the bottleneck buffer, the propagation delay, and the average of the data sizes transmitted as background data traffic. Statistical analyses of the experimental results suggest that managing the standard deviation of jitters in a network serves as a promising methodology, because its close relation to VoIP speech quality possesses robustness to changes in the network conditions. The robustness makes it practically useful since telecommunications operators can apply it to their networks, which are subject to change. The findings in this paper have opened up new visions for telecommunications operators to manage the Quality of Service (QoS) of VoIP service.

Applying the formerly proposed classification framework for first order line search optimization techniques we introduce novel superlinear first order line search methods. Novelty of the methods lies in the line search subproblem. The presented line search subproblem features automatic step length and momentum adjustments at every iteration of the algorithms realizable in a single step calculation. This keeps the computational complexity of the algorithms linear and does not harm the stability and convergence of the methods. The algorithms have none or linear memory requirements and are shown to be convergent and capable of reaching the superlinear convergence rates. They were practically applied to artificial neural network training and compared to the relevant training methods within the same class. The simulation results show satisfactory performance of the introduced algorithms over the standard and previously proposed methods.

This paper presents a new submesh allocation scheme for mesh connected multicomputer systems, called CFSL-TR (Classified Free Submesh List-Task Relocation), which reduces task waiting time in two aspects, shortening submesh search time and reducing the submesh allocation delay caused by external fragmentation. This scheme classifies independent free submeshes by their types: square, horizontal rectangle, or vertical rectangle. Then it searches for the best-fit submesh only from one list depending on the type of the given task, thus saving submesh searching time. If no suitable submeshes are found, it is most likely caused by external fragmentation. In such a case, our scheme relocates the tasks being executed to free submeshes and combines the newly available submesh with other fragmented ones to form a larger submesh. This allows allocation of the task, otherwise to be put on the queue, hence reducing the submesh allocation delay. Through simulation, we show that our scheme helps reduce task waiting time and that it is by far more effective to reduce the submesh allocation delay caused by external fragmentation rather than to reduce submesh search time for reduction of the task waiting time.

A high-speed bipolar ECL comparator circuit with a latch is described. The spike noise generated by charging the base-to-emitter diffusion capacitor on the transition of differential transistors' switching in a sample-and-latch circuit is reduced by inserting the emitter degeneration resistors so that neither of them becomes completely cut off. The frequency bandwidth of a pre-amplifier is increased by using coupled inductors as differential loads. As a result, -3 dB frequency bandwidth of a pre-amplifier becomes 10 GHz, and 4 GS/s operation with 6-bit equivalent precision from a 3.3 V power supply is confirmed by the circuit simulation using device parameters from the 25 GHz silicon bipolar process.

The purpose of this study is to design a humanoid robotic hand system that is capable of conveying feelings and sensitivities by finger movement for the non-verbal communication between men and robots in the near future. In this paper, studies have been made in four steps. First, a small-sized and light-weight robotic hand was developed to be used as the humanoid according to the concept of extracting required minimum motor functions and implementing them to the robot. Second, basic characteristics of the movement were checked by experiments, simple feedforward control mechanism was designed based on velocity control, and a system capable of tracking joint time-series change command with arbitrary pattern input was realized. Third, tracking performances with regard to sinusoidal input with different frequencies were studied for evaluation of the system thus realized, and space- and time-related accuracy were investigated. Fourth, the sign language motions were generated as examples of information transmission by finger movement. A series of results thus obtained indicated that this robotic hand is capable of transmitting information promptly with comparatively high accuracy through the movement.

In this paper, we propose a new haptic interface for two-handed manipulation. The system, named the SPIDAR-G&G system, consists of a pair of string-based 6DOF haptic devices called SPIDAR-G for both hands. By grasping the grip of each SPIDAR-G in each of the user's hands, the user can manipulate one virtual object with their right hand and the other one with their left hand cooperatively, while the user senses interaction force. We evaluated the system by measuring the completion time of a 3D pointing task, and demonstrated enhanced interactivity with virtual objects.

The unpredictable behavior of cache memory makes it difficult to statically analyze the worst-case performance of real-time systems. This problem is further exacerbated in the case of preemptive multitask systems because of inter-task cache interference, called Cache-Related Preemption Delay (CRPD). This paper proposes an approach to analyzing the tight upper bound on CRPD which a task might impose on lower-priority tasks. Our method finds the program execution path which requires the maximum number of cache blocks using an integer linear programming technique. Experimental results show that our approach provides up to 69% tighter bounds on CRPD than a conservative approach.

In this paper, we perform some experiments to show that the quantization noise caused by low-bit-rate speech coding can be characterized as a white noise process. Then, the signal-to-quantization noise ratio of the decoded speech for a given bit-rate is estimated by observing the perceptual speech quality equivalent to the artificially generated noisy speech obtained by adding a white Gaussian noise source. This information is incorporated into the parameter tuning of a noise-robust compensation algorithm for speech recognition so that the compensation algorithm can be performed better under a range of the estimated SNRs. Finally, we apply the compensation algorithm to a connected digit string recognition system that utilizes speech signals decoded by the GSM adaptive multi-rate (AMR) speech coder. It is shown that the noise-robust compensation algorithm reduces word error rates by 15% or more at low bit-rate modes of the AMR speech coder.

In this paper we propose a parallel and distributed computation of genetic local search with irregular topology in distributed environments. The scheme we propose in this paper is implemented with a tree topology established on an irregular network where each computing element carries out genetic local search on its own chromosome set and communicates with its parent when the best solution of each generation is updated. We evaluate the proposed algorithm by a simulation system implemented on a PC-cluster. We test our algorithm on four types topologies: star, line, balanced binary tree and sided binary tree, and investigate the influence of communication topology and delay on the evolution process.

Internet traffic engineering is much important for Internet Service Providers (ISPs) today, since it can be used to fully utilize already deployed network resources. For ISPs, the requirements for traffic engineering should be simple, easy to configure, cost-effective and efficient. Based on these considerations, we propose an algorithm called Web First Adaptive Traffic Engineering (WFATE). Since World Wide Web (WWW) services dominate most of the total Internet traffic and WWW flows are not long-lived, we only apply load balancing to WWW traffic in the algorithm. It can be shown that the number of coexistent WWW flows at an ingress node is almost certainly below a bound, and thus a forward-per-flow mechanism without keeping track of the state of each flow is feasible. This mechanism can balance traffic load at fine granularity and therefore get better performance. Through simulations and performance comparison, it is shown that WFATE is quite efficient, which can improve the network throughput averagely by 26% under the "dense source" traffic pattern and 9% under the "sparse source" traffic pattern.

In order to perform the registration of virtual objects in vision-based augmented reality systems, the estimation of the relation between the real and virtual worlds is needed. This paper presents a three-point vision-based registration method for video see-through augmented reality systems using binocular cameras. The proposed registration method is based on a combination of monocular and stereoscopic registration methods. A correction method that performs an optimization of the registration by correcting the 2D positions in the images of the marker feature points is proposed. Also, an extraction strategy based on color information is put forward to allow the system to be robust to fast user's motion. In addition, a quantification method is used in order to evaluate the stability of the produced registration. Timing and stability results are presented. The proposed registration method is proven to be more stable than the standard stereoscopic registration method and to be independent of the distance. Even when the user moves quickly, our developed system succeeds in producing stable three-point based registration. Therefore, our proposed methods can be considered as interesting alternatives to produce the registration in binocular augmented reality systems when only three points are available.

VoIP is one of the key technologies for recent telecommunication services. The quality of its services should be discussed in subjective terms. Since subjective quality assessment is time-consuming and expensive, however, objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. This paper discusses the performance of the objective quality measure that was standardized as ITU-T Recommendation P.862 and clarifies the quality factors that can be evaluated with satisfactory accuracy based on it. We found that P.862 can be applied to the evaluation of coding distortion, tandeming of codecs, transmission bit-errors, packet loss, and silence compression in a codec, at least for clean Japanese speech. In addition, we propose a method of estimating the subjective quality evaluation value from objective measurement results and show the validity of this method. We also evaluate the uniqueness of objective quality assessment based on P.862 from the viewpoints of the effect of measurement noise and the variation of test speech samples, and propose how to improve the reproducibility of objective quality assessment.

There is a fatal difference in obtaining information between sighted people and the blind. Screen reading technology assists blind people in accessing digital documents by themselves helping to bridge such gap. However, these days they are becoming much more visual using various types of visual effects for sighted people to explore the information intuitively at a glance. It is very hard to convey visual effects non-visually and intuitively while retaining the original effects. In addition, it takes a long time to explore the information, since blind people use the keyboard for exploration, while sighted people use eye movement. This research aims at improving the non-visual exploration interface and improving the quality of non-visual information. Therefore, TAJODA (tactile jog dial interface) was proposed to solve these problems. It presents verbal information (text information) in the form of speech, while nonverbal information (visual effects) is represented in the form of tactile sensations. It uses a jog dial as an exploration device, which makes it possible to explore forward or backward intuitively in the speech information by spinning the jog dial clockwise or counterclockwise. It also integrates a tactile device to represent visual effects non-visually. Both speech and tactile information can be synchronized with the dial movements. The speed of spinning the dial affects the speech rate. The main part of this paper describes an experimental evaluation of the effectiveness of the proposed TAJODA interface. The experimental system used a preprocessed recorded human voice as test data. The training sessions showed that it was easy to learn how to use TAJODA. The comparison test session clearly showed that the subjects could perform the comparison task using TAJODA significantly faster (2.4 times faster) than with the comparison method that is closest to the existing screen reading function. Through this experiment, our results showed that TAJODA can drastically improve the non-visual exploration interface.