An adaptive array antenna for the suppression of high-power interference in direct-sequence code-division multiple access (DS-CDMA) systems is presented. Although DS-CDMA has sufficient flexibility to support a variety of services, from voice to moving-pictures, with high levels of quality, multiple access interference (MAI) is a problem. This is particularly so of the high-power interference which accompanies high-speed transmission in DS-CDMA. While the application of adaptive array antennas is an effective way of improving signal-to-interference-plus-noise ratio (SINR), problems with this approach include large levels of power consumption and the high costs of hardware and of implementing the antennas. Therefore, our main purpose is to realize a simple configuration for an adaptive array system. In order to reduce the required amounts of processing, a common beam provides suppression of high-power interference for the low-bit-rate users; this makes per-user preparation of weights unnecessary. This approach also reduces the consumption of power by the system. Interference is cancelled by minimization of the array output power (i.e., the application of a power inversion algorithm) before despreading. The approach also allows us to improve the implementation of the antenna elements by using small auxiliary antennas. The basic performance of the system is confirmed through numerical calculation and computer simulation. Furthermore, a real-time processing unit has been developed and the effectiveness of the approach is confirmed by an experiment in a radio-anechoic chamber.

In this study, we introduce a method for estimating the syntactic structure of Japanese speech from F0 contour and pause duration. We defined a prosodic unit (PU) which is divided by the local minimal point of an F0 contour or pause. Combining PUs repeatedly (a pair of PUs is combined into one PU), a tree structure is gradually generated. Which pair of PUs in a sequence of three PUs should be combined is decided by a discriminant function based on the discriminant analysis of a corpus of speech data. We applied the method to the ATR Phonetically Balanced Sentences read by four Japanese speakers. We found that with this method, the correct rate of judgement for each sequence of three PUs is 79% and the estimation accuracy of the entire syntactic structure for each sentence is 26%. We consider this result to demonstrate a good degree of accuracy for the difficult task of estimating syntactic structure only from prosody.

The present paper deals with the blind deconvolution of a Multiple-Input Multiple-Output Finite Impulse Response (MIMO-FIR) system. To deal with the blind deconvolution problem using the second-order statistics (SOS) of the outputs, Hua and Tugnait considered it under the conditions that a) the FIR system is irreducible and b) the input signals are spatially uncorrelated and have distinct power spectra. In the present paper, the problem is considered under a weaker condition than the condition a). Namely, we assume that c) the FIR system is equalizable by means of the SOS of the outputs. Under b) and c), we show that the system can be blindly identified up to a permutation, a scaling, and a delay using the SOS of the outputs. Moreover, based on this identifiability, we show a novel necessary and sufficiently condition for solving the blind deconvolution problem, and then, based on the condition, we propose a new algorithm for finding an equalizer using the SOS of the outputs, while Hua and Tugnait have not proposed any algorithm for solving the blind deconvolution under the conditions a) and b).

It is well known that KT method proposed by R. Kumaresan and D. W. Tufts is used as a popular parameter estimation method of exponentially damped signal. It is based on linear backward-prediction method and singular value decomposition (SVD). However, it is difficult to estimate parameters correctly by KT method in the case when high noise exists in the signal. In this paper, we propose a parameter (frequency components and damping factors) estimation method to improve the performance of KT method under high noise. In our proposed method, we find the signal zero groups by calculating zeros with different data record lengths according to the combination of forward-prediction and backward-prediction, the mean value of the zeros in the signal zero groups are calculated to estimate the parameters of the signal. The proposed method can estimate parameters correctly and accurately even when high noise exists in the signal. Simulation results are shown to confirm the effectiveness of the proposed method.

Appropriate test signals defined by formula or generated by algorithm are used for measuring objective QoS (Quality of Services) for voice operated telecommunication devices such as telephone and speech codec (coder-decoder). However, that for measuring residual echo characteristics in hands-free telecommunications equipped with acoustic echo canceller is under study in ITU-T Recommendation G.167. This paper describes comparative assessment of test signals for measurement of residual echo characteristics. In hands-free telecommunications, acoustical echo canceller has been developed to remove a room echo signal through the loudspeaker to the microphone in the receiving end. Performance of the echo canceller system is evaluated by residual echo characteristics expressed in echo return loss enhancement (ERLE). The ERLE can be conventionally measured by putting white noise into the echo canceller system. However, white noise is not adequate as the test signal for measuring the performance of the echo canceller, since the performance may depend on the characteristics of input test signal, and the characteristics of the white noise differ from those of real voice. Therefore, this paper discusses appropriate characteristics of real voice required for objective quality evaluation of echo canceller system. The test signals used for this verification tests were real voice (RV), white noise (WN), frequency weighted noise (FWN), artificial voice (AV), and composite source signal (CSS) depending on the approximation of real voice characteristics. As the comparative assessment results, the ERLE characteristics measured by artificial voice conforming to ITU-T Recommendation P.50 having average characteristics of real voices in time and frequency domains are almost equivalent to those of real voice and best among those test signals. It is concluded that artificial voice P.50 is satisfied with measurement of residual echo characteristics.

In this letter, we study the performance of the iterative receiver as applied to the space division multiplexing/orthogonal frequency division multiplexing (SDM/OFDM) systems. The iterative receiver under consideration employs the soft in/soft out (SISO) decoding process, which operates iteratively in conjunction with channel estimation for performing data detection and channel estimation at the same time. As opposed to the previous studies in which the perfect channel state information is assumed, the effects of channel estimation are taken into account for evaluating the performance of the iterative receiver and it is shown that the channel estimation applied in every iteration step of the iterative receiver plays a crucial role to warrant the performance, especially at a low signal-to-noise power ratio (SNR).

Invention and development of the Yagi-Uda antenna, and the self-complementary antenna are described. Analysis methods of large loop antennas and the improved circuit theory (ICT) for design of linear antennas are presented. Recent developments of axial mode helical antennas and spiral antennas for radiating circularly polarized waves are also described.

This paper reviews the historical development of adaptive antennas in Japan. First of all, we watch basic adaptive algorithms. In 1980s, particularly, the following issues were a matter of considerable concern to us; (a) behavior to the coherent interference like multipath waves or radar clutters, (b) signal degradation in case that the direction of arrival (DOA) of desired signal is different from the DOA specified beforehand in the adaptive antennas with the DOA of the desired signal as a prior knowledge, and (c) performance of adaptive antennas when the desired signal and interference are broadband. Although there are a lot of development and modification of adaptive algorithms in Japan, we refer in this paper only to the above-mentioned topics. Secondly, our attention is paid to implementation of adaptive antennas and advanced technologies. A large number of researches on the subjects have been carried out in Japan. Particularly, we focus on the initiative studies in Japan toward mobile communication application. They include researches of mobile radio propagation for adaptive antennas, calibration methods, and adaptive antenna for mobile terminals. As a matter of course, we also refer to adaptive antenna technologies for advanced communication schemes such as CDMA, SDMA, OFDM and so on. Finally, we take notice of some pilot products which were developed to verify the effect of the adaptive antenna in the practical environments. As the initiative ones, a couple of equipments are introduced in this paper.

In this paper, we propose a new technique of filter bank subtraction for robust speech recognition under various acoustic conditions. Spectral subtraction is a simple and useful technique for reducing the influence of additive noise. Conventional spectral subtraction assumes accurate estimation of the noise spectrum and no correlation between speech and noise. Those assumptions, however, are rarely satisfied in reality, leading to the degradation of speech recognition accuracy. Moreover, the recognition improvement attained by conventional methods is slight when the input SNR changes sharply. We propose a new method in which the output values of filter banks are used for noise estimation and subtraction. By estimating noise at each filter bank, instead of at each frequency point, the method alleviates the necessity for precise estimation of noise. We also take into consideration expected phase differences between the spectra of speech and noise in the subtraction and control a subtraction coefficient theoretically. Recognition experiments on test sets at several SNRs showed that the filter bank subtraction technique improved the word accuracy significantly and got better results than conventional spectral subtraction on all the test sets. In other experiments, on recognizing speech from TV news field reports with environmental noise, the proposed subtraction method yielded better results than the conventional method.

We have developed a fiber physical contact (FPC) connector for the high-density connection of optical fibers. This connector individually aligns multiple bare fibers in micro-holes without ferrules and realizes physical contact by using the buckling force of the fibers themselves. The fiber endfaces must be tapered to allow the fibers to be inserted into the micro-holes. The endfaces must also be polished so that they realize physical contact (PC) with excellent optical performance. For each process, we examined the required shape and processing condition of the fiber endface for the FPC connector. As regards tapering, we determined the processing condition for achieving a target tapering angle and developed a non-breaking process with the optical fibers bent. In terms of polishing, we revealed that it is important for the fiber endface angle error to be less than 0.7 degrees if we are to achieve excellent optical performance. These results allowed us to fabricate an FPC connector that exhibited excellent levels of optical performance.

There is a growing demand that mobile networks should provide quality-of-service (QoS) to mobile users since portable devices become popular and more and more applications require real-time services. Providing QoS to mobile hosts is very difficult due to mobility of hosts. The resource ReSerVation Protocol (RSVP) establishes and maintains a reservation state to ensure a given QoS level between the sender and receiver. However, RSVP is designed for fixed networks and thus it is inadequate in wireless mobile networking environments. In this paper, we propose a resource reservation protocol for mobile hosts in mobile networks. The proposed protocol extends the RSVP by introducing RSVP agents in local networks to manage the reservations. The proposed protocol reduces packet delay, bandwidth overhead, and the number of RSVP messages to maintain reservation states. We examined the performance of the proposed protocol by simulation and we got an improved performance over the existing protocols.

The goal of the maximum cut problem is to partition the vertex set of an undirected graph into two parts in order to maximize the cardinality of the set of edges cut by the partition. The maximum cut problem has many important applications including the design of VLSI circuits and communication networks. Moreover, many optimization problems can be formulated in terms of finding the maximum cut in a network or a graph. In this paper, we propose an improved Hopfield neural network algorithm for efficiently solving the maximum cut problem. A large number of instances have been simulated. The simulation results show that the proposed algorithm is much better than previous works for solving the maximum cut problem in terms of the computation time and the solution quality.

This paper discusses a nonlinear function for independent component analysis to process complex-valued signals in frequency-domain blind source separation. Conventionally, nonlinear functions based on the Cartesian coordinates are widely used. However, such functions have a convergence problem. In this paper, we propose a more appropriate nonlinear function that is based on the polar coordinates of a complex number. In addition, we show that the difference between the two types of functions arises from the assumed densities of independent components. Our discussion is supported by several experimental results for separating speech signals, which show that the polar type nonlinear functions behave better than the Cartesian type.

This paper considers a link of two problems; multichannel blind deconvolution and multichannel blind identification of linear time-invariant dynamic systems. To solve these problems, cumulant maximization has been proposed for blind deconvolution, while cumulant matching has been utilized for blind identification. They have been independently developed. In this paper, a cumulant maximization criterion for multichannel blind deconvolution is shown to be equivalent to a least-squares cumulant matching criterion after multichannel prewhitening of channel outputs. This equivalence provides us with a new link between a cumulant maximization criterion for blind deconvolution and a cumulant matching criterion for blind identification.

This paper proposes two techniques for improving the accuracy of gate-level power analysis for system-on-a-chip (SoC). (1) Creation of custom wire load models for clock nets. (2) Use of layout information (actual net capacitance and input signal transition time). The analysis time is reduced to less than one three-hundredth of the transistor-level power analysis time. Error is within 5% against a real chip, (the same level as that of the transistor-level power analysis), if technique (2) is used, and within 15% if technique (1) is used.

A dual-modulus (divide-by-128/129) prescaler has been designed based on 0.25-µm CMOS technology employing new D-flip-flops. The new D-flip-flops are free from glitch problems due to internal charge sharing. Transistor merging technique has been employed to reduce the number of transistors and to secure reliable high-speed operation. At the 2.5-V supply voltage, the prescaler using the proposed dynamic D-flip-flops can operate up to the frequency of 2.95-GHz, and consumes about 10% and about 27% less power than Yuan/Svensson's and Huang's circuits, respectively.

The wavelet transform (WT) has recently emerged as a powerful tool for image compression. In this paper, a new image compression technique combining the genetic algorithm (GA) and grey-based competitive learning network (GCLN) in the wavelet transform domain is proposed. In the GCLN, the grey theory is applied to a two-layer modified competitive learning network in order to generate optimal solution for VQ. In accordance with the degree of similarity measure between training vectors and codevectors, the grey relational analysis is used to measure the relationship degree among them. The GA is used in an attempt to optimize a specified objective function related to vector quantizer design. The physical processes of competition, selection and reproduction operating in populations are adopted in combination with GCLN to produce a superior genetic grey-based competitive learning network (GGCLN) for codebook design in image compression. The experimental results show that a promising codebook can be obtained using the proposed GGCLN and GGCLN with wavelet decomposition.

We propose a new algorithm for blind source separation (BSS), in which independent component analysis (ICA) and beamforming are combined to resolve the low-convergence problem through optimization in ICA. The proposed method consists of the following two parts: frequency-domain ICA with direction-of-arrival (DOA) estimation, and null beamforming based on the estimated DOA. The alternation of learning between ICA and beamforming can realize fast- and high-convergence optimization. The results of the signal separation experiments reveal that the signal separation performance of the proposed algorithm is superior to that of the conventional ICA-based BSS method.

Emerging requirements for higher rate data services and better spectrum efficiency are the main issues of third-generation mobile radio systems. In particular, a new concept of burst switching has been introduced for supporting the packet data services in the CDMA-based wireless system. In the burst switching system, radio resources are allocated to users for the duration of data bursts, which is a series of packets, as opposed to the conventional packet switching scheme. To implement the burst switching scheme, three different states (active, control hold, dormant states) are defined and two transition timers are employed to release the fundamental and supplemental code channels, respectively, at certain instances. Furthermore, the system is subject to burst admission control policy, with which a burst is admitted only when the number of currently available channels is greater than the admission threshold. Since there exists a trade-off between the additional packet access delay during a burst and resource utilization depending on the time-out value of the transition timer and burst admission threshold, it is critical to understand the performance characteristics in terms of the underlying design parameters. In this paper, we develop an analytic model and present a Quasi-Birth-Death (QBD) queueing analysis for evaluating the performance of burst switching schemes. This work focuses on the trade-off studies for optimizing the time-out value of the transition timer so as to minimize the average delay performance. Theoretical performance measures are derived by means of the matrix geometric method and furthermore, some simulation results are presented to validate the proposed analytical approach.

Small-size MOSFETs are becoming core devices in RF applications because of improved high frequency characteristics. For reliable design of RF integrated circuits operating at the GHz range, accurate modeling of small-size MOSFET characteristics is indispensable. In MOSFETs with reduced gate length (Lg), the lateral field along the MOSFET channel is becoming more pronounced, causing short-channel effects. These effects should be included in the device modeling used for circuit simulation. In this work, we investigated the effects of the field gradient in the gate-drain capacitance (Cgd). 2-Dimensional (2D) simulations done with MEDICI show that the field gradient, as it influences the channel condition, induces a capacitance which is visible in the MOSFET saturation operation. Changes in Cgd is incorporated in the modeling by an induced capacitance approach. The new approach has been successfully implemented in the surface-potential based model HiSIM (Hiroshima-university STARC IGFET Model) and is capable of reproducing accurately the measured Cgd-Lg characteristics, which are particularly significant for pocket-implant technology. Results show that pocket-implantation introduces a steep potential increase near the drain region, which results to a shift of the Cgd transition region (from linear to saturation) to lower bias voltages. Cgd at saturation decreases with Lg due to steeper surface potential and increased impurity concentration effects at reduced Lg.