Joint frequency-domain equalization (FDE) and antenna diversity combining is applied to the reception of multi-rate DS-CDMA signals to achieve the frequency diversity effect while suppressing inter-path interference (IPI) resulting from the asynchronism of different propagation paths. At a receiver, fast Fourier transform (FFT) is applied for FDE and then inverse FFT (IFFT) is used to obtain a frequency-domain equalized DS-CDMA chip sequence for the succeeding despreading operation. An arbitrary spreading factor SF can be used for the given value of FFT window size; an extreme case is the nonspread SC system with SF=1. This property allows a flexible design of multi-rate DS-CDMA systems. Three types of FDE are considered; minimum mean square error (MMSE) equalization, maximal-ratio combining (MRC) equalization and zero-forcing (ZF) equalization. Matched filter bound analysis for achievable BER performance is presented. The improvement in the BER performance in a frequency-selective Rayleigh fading channel is evaluated by computer simulation. First, we consider the single-user case and compare the BER performances achievable with MMSE, MRC and ZF equalizations. How the fading rate and the spreading factor affect the BER performance is also evaluated. Furthermore, the BER performance comparison between FDE and rake combining is presented for various values of SF and also performance comparison between DS-CDMA and SC signal transmissions, both using FDE, is presented. Finally, we extend our evaluation to the multi-user case. Both downlink and uplink are considered and how the BER performances of downlink and uplink differ is discussed.

A waveguide crossed-slot linear array with a matching element is accurately analyzed and designed by the method of moments using numerical-eigenmode basis functions developed by the authors. The rounded ends of crossed-slots are accurately modeled in the analysis. The initial values of the slot parameters determined by a model with assumption of periodicity of field are modified and refined by the full-wave finite-array analysis for uniform excitation and small axial ratio. As an example, an 8-element linear array is designed at 11.85 GHz, which radiates a circularly polarized wave at a beam-tilting angle of 50 degrees. The radiation pattern, the frequency characteristics of the reflection and the axial ratio are compared between the analysis and the measurement and they agree very well. The calculated and measured axial ratio at the beam direction are 0.1 dB and 1.7 dB, respectively. This method provides a basic and powerful design tool for slotted waveguide arrays.

In this paper, conventional validity indexes are reviewed and the shortcomings of the fuzzy cluster validation index based on inter-cluster proximity are examined. Based on these considerations, a new cluster validity index is proposed for fuzzy partitions obtained from the fuzzy c-means algorithm. The proposed validity index is defined as the average value of the relative intersections of all possible pairs of fuzzy clusters in the system. It computes the overlap between two fuzzy clusters by considering the intersection of each data point in the overlap. The optimal number of clusters is obtained by minimizing the validity index with respect to c. Experiments in which the proposed validity index and several conventional validity indexes were applied to well known data sets highlight the superior qualities of the proposed index.

This paper analytically formulates both the optimal quantization noise allocation ratio and the coding gain of the two-dimensional morphological Haar wavelet transform. The two-dimensional morphological Haar wavelet transform has been proposed as a nonlinear wavelet transform. It has been anticipated for application to nonlinear transform coding. To utilize a transformation to transform coding, both the optimal quantization noise allocation ratio and the coding gain of the transformation should be derived beforehand regardless of whether the transformation is linear or nonlinear. The derivation is crucial for progress of nonlinear transform image coding with nonlinear wavelet because the two-dimensional morphological Haar wavelet is the most basic nonlinear wavelet. We derive both the optimal quantization noise allocation ratio and the coding gain of the two-dimensional morphological Haar wavelet transform by introducing appropriate approximations to handle the cumbersome nonlinear operator included in the transformation. Numerical experiments confirmed the validity of formulations.

For a kernel-based topographic map formation, kMER (kernel-based maximum entropy learning rule) was proposed by Van Hulle, and some effective learning rules related to kMER have been proposed so far with many applications. However, no discusions have been made concerning the determination of the number of units in kMER. This letter describes a unit-pruning rule, which permits automatic contruction of an appropriate-sized map to acquire the global topographic features underlying the input data. The effectiveness and the validity of the present rule have been confirmed by some preliminary computer simulations.

This paper investigates the parameter tying structures of a mixture of factor analyzers (MFA) and discriminative training of MFA for speaker identification. The parameters of factor loading matrices or diagonal matrices are shared in different mixtures of MFA. Then, minimum classification error (MCE) training is applied to the MFA parameters to enhance the discrimination ability. The result of a text-independent speaker identification experiment shows that MFA outperforms the conventional Gaussian mixture model (GMM) with diagonal or full covariance matrices and achieves the best performance when sharing the diagonal matrices, resulting in a relative gain of 26% over the GMM with diagonal covariance matrices. The improvement is more significant especially in sparse training data condition. The recognition performance is further improved by MCE training with an additional gain of 3% error reduction.

This paper presents a new framework for effectively creating VoiceFonts for speech synthesis. A VoiceFont in this paper represents a voice inventory aimed at generating personalized voices. Creating well-formed voice inventories is a time-consuming and laborious task. This has become a critical issue for speech synthesis systems that make an attempt to synthesize many high quality voice personalities. The framework we propose here aims to drastically reduce the burden with a twofold approach. First, in order to substantially enhance the accuracy and robustness of automatic speech segmentation, we introduce a multi-layered speech segmentation algorithm with a new measure of segmental reliability. Secondly, to minimize the amount of human intervention in the process of VoiceFont creation, we provide easy-to-use functions in a data viewer and compiler to facilitate checking and validation of the automatically extracted data. We conducted experiments to investigate the accuracy of the automatic speech segmentation, and its robustness to speaker and style variations. The results of the experiments on six speech corpora with a fairly large variation of speaking styles show that the speech segmentation algorithm is quite accurate and robust in extracting segments of both phonemes and accentual phrases. In addition, to subjectively evaluate VoiceFonts created by using the framework, we conducted a listening test for speaker recognizability. The results show that the voice personalities of synthesized speech generated by the VoiceFont-based speech synthesizer are fairly close to those of the donor speakers.

Behavioral synchronization between speech and finger tapping provides a novel approach to improving speech recognition accuracy. We combine a sequence of finger tapping timings recorded alongside an utterance using two distinct methods: in the first method, HMM state transition probabilities at the word boundaries are controlled by the timing of the finger tapping; in the second, the probability (relative frequency) of the finger tapping is used as a 'feature' and combined with MFCC in a HMM recognition system. We evaluate these methods through connected digit recognition under different noise conditions (AURORA-2J). Leveraging the synchrony between speech and finger tapping provides a 46% relative improvement in connected digit recognition experiments.

A wavelet feature selection derived by using fuzzy evaluation index for speaker identification is described. The concept of a flexible membership function incorporating weighed distance is introduced in the evaluation index to make the modeling of clusters more appropriate. Our results have shown that this feature selection introduced better performance than the wavelet features with respect to the percentages of recognition.

A game-theoretic analysis is applied to the evaluation of capacity and stability of a wireless ad hoc network in which each source node independently chooses a route to the destination node so as to enhance throughput. First, the throughput of individual multihop transmission with rate adaptation is evaluated. Observations from this evaluation indicate that the optimal number of hops in terms of the achievable end-to-end throughput depends on the received signal-to-noise ratio. Next, the decentralized adaptive route selection problem in which each source node competes for resources over arbitrary topologies is defined as a game. Numerical results reveal that in some cases this game has no Nash equilibria; i.e., each rational source node cannot determine a unique route. The occurrence of such cases depends on both the transmit power and spatial arrangement of the nodes. Then, the obtained network throughput under the equilibrium conditions is compared to the capacity under centralized scheduling. Numerical results reveal that when the transmit power is low, decentralized adaptive route selection may attain throughput near the capacity.

We present a world model for location-aware and user-aware services in ubiquitous computing environments. It can be dynamically organized like a tree based on geographical containment, such as in a user-room-floor-building hierarchy and each node in the tree can be constructed as an executable software component. The model is unique to existing approaches because it enables location-aware services to be managed without databases, can be managed by multiple computers, and provides a unified view of the locations of not only physical entities and spaces, including users and objects, but also computing devices and services. A prototype implementation of this approach was constructed on a Java-based mobile agent system. This paper presents the rationale, design, implementation, and applications of the prototype system.

Process-centered software engineering environments (PSEEs) facilitate controlling complicated software processes. Traditional PSEEs are generally centrally controlled, which may result in the following drawbacks: (1) the server may become a bottleneck and (2) when the server is down, processes need to be suspended. To overcome the drawbacks, we developed a decentralized process engine ADPE (agent-based decentralized process engine). ADPE can be embedded in any PSEE to decentralize the PSEE. This paper presents ADPE.

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.

160 Gbit/s optical-time-division-multiplexing (OTDM) transmitter/receiver employing electroabsorption (EA) modulators are described. In the 160 Gbit/s OTDM transmitter, the optical multiplexer, which implemented four EA modulators, is used and the generation of authentic 160 Gbit/s OTDM signal is realized. The optical multiplexer also enables to generate the phase-coded OTDM signal such as carrier-suppressed return-to-zero (CS-RZ) signal at 160 Gbit/s by changing driving temperatures of the EA modulators. In the 160 Gbit/s receiver, the EA modulator is also used in an optical demultiplexer and a phase-locked-loop (PLL) for clock extraction. As both optical demultiplexer and PLL are insensitive to polarization state of incoming signal, highly stable operation is achieved. We also show some results of transmission experiment using the developed OTDM transmitter/receiver and discuss the advantage of a switching capability of modulation format in the 160 Gbit/s signal transmission.

Current literature on input buffer management reveals that, in representative ATM networks under highly bursty traffic conditions, the fuzzy thresholding approach yields lower cell loss rate at the cost of lower throughput. Also, under less bursty traffic, the traditional fixed thresholding approach achieves higher throughput at the expense of higher cell loss rate. The integration of these two properties into practice is termed adaptive dynamic buffer management (ADBM) approach for input buffers and its assessment is the objective of this paper. The argument is that, given that the traffic conditions are constantly changing, to achieve efficiency during actual operation, the network control must dynamically switch, at every ATM switch, under the call processor's control, between the two input buffer management techniques, dictated by the nature of the traffic at the inputs of the corresponding switch. The need to involve the call processor marks the first effort in the literature to dynamically configure input buffer management architectures at the switch fabric level under higher level call processor control. It stems from the fact that the switch fabric operates very fast and cannot engage in complex decision making without incurring stiff penalty. To achieve this goal, the network control needs knowledge of the burstiness of the traffic at the inputs of every ATM switch. The difficulties with this need are two-fold. First, it is not always easy to obtain the traffic model and model parameters for a specific user's call. Second, even where the traffic model and the model parameters are known for a specific user's call, this knowledge is valid only at the source switch where the user interfaces with the network. At all other switches in the network, the cells of the traffic in question interact asynchronously with the cells from other traffic sources and are subject to statistical multiplexing. Thus, to obtain the exact nature of the composite traffic at the inputs of any ATM switch, is a challenge. Conceivably, one may determine the burstiness by counting the number of cells incurred at the inputs of an ATM switch over a defined time interval. The challenge posed by this proposition lies in the very definition of burstiness in that the time interval must approach, in the limit, zero or the resolution of time in the network. To address this challenge, first, a 15-node representative ATM network is modeled in an asynchronous, distributed simulator and, second, simulated on a network of workstations under realistic traffic stimuli. Third, burstiness indices are measured for the synthetic, stochastic traffic at the inputs of every ATM switch as a function of the progress of simulation for different choices of time interval values, ranging from 20,000 timesteps down to 1,000 timesteps. A timestep equals 2.73 µs. Results reveal that consistent burstiness indices are obtained for interval choices between 1,000 and 5,000 timesteps and that a burstiness index of 25, measured at 3,000 timestep interval, constitutes a reasonable and practical threshold value that distinguishes highly bursty traffic that warrants the use of the fuzzy thresholding approach from less bursty traffic that can benefit from the fixed thresholding scheme. A comparative performance analysis of ADBM yields the following. For pure fixed and pure fuzzy thresholding schemes, the throughputs are at 73.88% and 71.53% while the cell drop rates are at 4.31% and 2.44%,respectively. For the ADBM approach, where the input buffer management alternates at each individual ATM switch between the fixed and fuzzy schemes, governed by measured burstiness index threshold of 25 for a 3,000 timestep interval, the throughput is 74.77%, which is higher than even the pure fixed scheme while the cell drop rate is 2.21% that is lower than that of the pure fuzzy scheme. In essence, ADBM successfully integrates the best characteristics of the fuzzy and fixed thresholding schemes.

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.

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 letter describes the polarization mode dispersion (PMD) characteristic of optical fiber ribbons inserted tightly into helical slots. We investigate the mechanism of the birefringence induced in the optical fiber ribbons by lateral stress resulting from tension and bending in the helical slots. We discuss methods for the design of low PMD cables with reduced birefringence by considering coating materials and tensile strain control.

In this paper, we examine oscillatory modes generated by the Hopf bifurcations of equilibrium points except for the origin in a system of coupled four oscillators. (The bifurcation analyses of the origin for many coupled oscillators were already done.) The Hopf bifurcations of the equilibrium points with strong symmetrical property and the generated oscillatory modes are classified. We observe four-phase, in-phase and a pair of anti-phase synchronized states. Even in a system of four coupled oscillators, we discover the existence of a stable three-phase oscillation. By the numerical bifurcation analysis of generated periodic oscillations we find out successive period-doubling bifurcations as the route to chaos and show some of them are symmetry-breaking bifurcations. As a result of the symmetry-breaking period-doubling bifurcations, a periodic solution with complete synchronization becomes a chaotic solution with phase synchronization.

This paper overviews recent progress in the development of corpus-based spontaneous speech recognition technology. Although speech is in almost any situation spontaneous, recognition of spontaneous speech is an area which has only recently emerged in the field of automatic speech recognition. Broadening the application of speech recognition depends crucially on raising recognition performance for spontaneous speech. For this purpose, it is necessary to build large spontaneous speech corpora for constructing acoustic and language models. This paper focuses on various achievements of a Japanese 5-year national project "Spontaneous Speech: Corpus and Processing Technology" that has recently been completed. Because of various spontaneous-speech specific phenomena, such as filled pauses, repairs, hesitations, repetitions and disfluencies, recognition of spontaneous speech requires various new techniques. These new techniques include flexible acoustic modeling, sentence boundary detection, pronunciation modeling, acoustic as well as language model adaptation, and automatic summarization. Particularly automatic summarization including indexing, a process which extracts important and reliable parts of the automatic transcription, is expected to play an important role in building various speech archives, speech-based information retrieval systems, and human-computer dialogue systems.