In this paper, we focus on the problem of secure multicast in dynamic groups. In this problem, a group of users communicate using a shared group key. Due to the dynamic nature of these groups, to preserve secrecy, it is necessary to change the group key whenever the group membership changes. While the group key is being changed, the group communication needs to be interrupted until the rekeying is complete. This interruption is especially necessary if the rekeying is done because a user has left (or is removed). We split the rekeying cost into two parts: the cost of the critical path--where each user receives the new group key, and the cost of the non-critical path--where each user receives any other keys that it needs to obtain. We present a family of algorithms that show the tradeoff between the cost of the critical path and the cost of the non-critical path. Our solutions allow the group controller to choose the appropriate algorithm for key distribution by considering the requirements on critical and non-critical cost. In our solutions, the group controller can dynamically change the algorithm for key distribution to adapt to changing application requirements. Moreover, we argue that our solutions allow the group controller to effectively manage heterogeneous groups where users have different requirements/capabilities.

For radar tracking, the α-β filter and the Kalman filter, both of which do not require large computational requirements, have been widely utilized. However these filters cannot track a maneuvering target accurately. In recent years, the IMM (Interactive Multiple Model) algorithm has been proposed. The IMM is expected to reduce tracking errors for both non-maneuvering and maneuvering target. However, the IMM requires heavy computational burden, because it utilizes multiple Kalman filters in parallel. On the other hand, the α-β filter with an acceleration term which can estimate maneuver acceleration from the past target estimated positions using the kinetic model, has been proposed. This filter is not available for tracking targets under clutter environment, since it does not calculate the covariance matrix which is needed for gate setting. In this paper, we apply the acceleration estimate to the Kalman filter, and propose the hybrid Kalman filter with a constant-velocity filter and an acceleration estimation filter, and it integrates the outputs of two filters using the normalized distance of the prediction error of each filter. The computational requirement of the proposed filter is smaller than that of the IMM since the proposed filter consists of only two Kalman based filters. The proposed method can prevent deteriorating tracking accuracy by reducing the risk of maneuver misdetection when a target maneuvers. We evaluate the performance of the proposed filter by computer simulation, and show the effectiveness of the proposed filter, comparing with the conventional Kalman filter and the two-stage Kalman filter.

Beam-space time coding methods are being extensively investigated, since they provide levels of performance appropriate for the next and future generations of wireless communication systems. In this paper, we focus on beam-domain space-time coding, especially considering the case when transmit beams have inter-beam interference (IBI). A new beam-space time coding scheme that takes into account the overlap amount among beams is proposed. We observe that the overlap of beams introduces an amount of correlation to the channels in a similar way to the well-known Partial Response (PR) channel in magnetic recording. Based on that observation, the proposed system can make use of IBI to encode and decode the signals. We evaluate the proposed system both via theoretical upper bound and via computer simulations. The Bit Error Rate (BER) performance of the proposed system using IBI is better than that of the system with no-IBI because the proposed system delivers more coding gain. However, the overlap of beams decreases the diversity gain. The tradeoff relationship between diversity gain and coding gain is investigated.

Determining indexing strings is an important factor in information retrieval. Ideally, the strings should be words that represent documents or queries. Although any single word may be the first candidate for indexing strings for an English corpus, it may not be ideal due to the existence of compound nouns, which are often good indexing strings, and which often depend on the genre of the corpus used. The situation is even worse in Japanese or Chinese where the words are not separated by spaces. In this paper, we propose a method of determining indexing strings based on statistical analysis. The novel features of our method are to make the most of the statistical measure called "adaptation" and not to use language-dependent resources such as dictionaries and stop word lists. In evaluating our method using a Japanese test collection, we found that it actually improves the precision of information retrieval systems.

A low complexity multi-user receiver with blind channel estimation and multiple access interference (MAI) suppression is proposed for a CDMA system under multipath fading and frequency offset. The design of the receiver involves the following procedure. First, a method of joint MAI suppression and channel estimation is developed based on the generalized sidelobe canceller (GSC) technique. In particular, channel estimates are obtained blindly in the form of the effective composite signature vectors (CSV) of the users. Second, a low-complexity partially adaptive (PA) realization of the receiver is proposed which incorporates reduced-rank processing based on the information of multi-user CSV's. By a judiciously designed decorrelating procedure, a new PA receiver is obtained with a much lower complexity. Finally, pilot symbols assisted frequency offset estimation and channel gain compensation give the estimate of users' symbols. Further performance enhancement is achieved by a decision aided scheme in which the signal is reconstructed and subtracted from the receiver input data, leading to significantly faster convergence. The proposed receiver is shown to be robust to multipath fading and frequency offset, and achieves nearly the same performance of the optimal maximum SINR and MMSE receivers with a much lower overhead for pilot symbols.

A microstrip array antenna with 45-degree inclined linear polarization is proposed for automotive radars. The proposed antenna has the advantages of high aperture efficiency, low profile and ease of manufacture. The rectangular radiating elements inclined at 45 degrees to the straight microstrip line are directly connected to it at their corners in the proposed array antenna. The radiating element has a feature that radiation conductance for co-polarization is controlled widely enough to set desired amplitude distribution keeping excited mode for cross-polarization negligibly small. The feed line loss of the linear array antenna having 15 wavelengths is estimated 0.9 dB in the design taking the loss of the microstrip line into account. The performance of two types of developed antennas, for electrical and mechanical scanning radars, is presented. The fan beam subarray antenna for electrical scanning radars has an aperture efficiency of 53% with gain of 22.5 dBi at 76.5 GHz. For mechanical scanning radars, the two-stage series feeding circuit is also proposed for lower feed line loss and setting desired amplitude distribution. The pencil beam array antenna has an aperture efficiency of 39% with gain of 32.2 dBi at 76.5 GHz.

Cochannel interference and multipath propagation reduce the performance of mobile communication systems. Multi-input MLSE with whitening processing can mitigate the influence of the interference and provide path diversity gain. In conventional considerations, however, the required complexity rapidly rises with the number of array elements. In this paper, we propose multi-input MLSE that whitens error signals in the signal space by using a multibeam adaptive array. This scheme can reduce the computational load of multi-input MLSE than the conventional type when using a large-element array. The results of an analysis show that the proposed type is equivalent to conventional one in the sense of the metric and provides less computational complexity.

ESPAR (Electronically Steerable Passive Array Radiator) antennas were proposed as candidates for low-cost analog adaptive beamforming. The radiation pattern is controlled in an azimuthal plane by variable reactors loaded on each passive element. This paper estimates the frequency bandwidth of an ESPAR antenna in a single beam scanning operation. Bandwidth in terms of gain is predicted statistically as functions of beam direction and dynamic range of variable reactance. The -3 dB bandwidth of 7-element ESPAR antennas can be about 30%, 25% and 15% for the range of reactance of -100Ω Xn 100Ω, -50Ω Xn 50Ω and -100Ω Xn 0Ω, respectively, while the improper choice of reactance sets results in narrow bandwidth less than 5%.

In this paper, a joint adaptive interference canceller (AIC) and maximum likelihood (ML) decoder scheme is proposed for a space-time coded DS-CDMA system with the difference between arrival times from transmit antennas. As the arrival time difference causes not only ISI and MAI, but also inter-antenna interference, performance degradation in the space-time coded DS-CDMA system is more severe than that of a regular DS-CDMA system with single transmit antenna. To mitigate the effect of the interference during space-time decoding, a joint algorithm for the proposed scheme merges adaptation process of the AIC into a ML decoding algorithm. Performance of the proposed scheme is evaluated for QPSK space-time trellis codes with two transmit antennas. It is shown that the proposed scheme achieves better performance than the conventional ML decoding scheme.

A robust adaptive filtering algorithm was established recently (I. Yamada, K. Slavakis, K. Yamada 2002) based on the interactive use of statistical noise information and the ideas developed originally for efficient algorithmic solutions to the convex feasibility problems. The algorithm is computationally efficient and robust to noise because it requires only an iterative parallel projection onto a series of closed half spaces highly expected to contain the unknown system to be identified and is free from the computational load of solving a system of linear equations. In this letter, we show the potential applicability of the adaptive algorithm to the identification problem for the second order Volterra systems. The numerical examples demonstrate that a straightforward application of the algorithm to the problem soundly realizes fast and stable convergence for highly colored excited speech like input signals in possibly noisy environments.

An adaptive backstepping and high order sliding modes control algorithm is proposed for output tracking of mobile robots. The controller can greatly reduce the chattering due to conventional sliding modes technique. The proposed algorithm has certain robustness with respect to the external random disturbances and good adaptability with respect to the parametric uncertainty. The effectiveness of the proposed control strategy is demonstrated by simulations studies.

In this paper, a practical adaptive TuCM scheme is proposed, and its adaptive method is described. With some hardware considerations, a suboptimal optimization algorithm which shows that the number of fading regions is variable is put forward. The proposed adaptive TuCM comes within 3 dB of fading channel capacity, exhibits about 3 dB power gain over conventional adaptive TCM, and is easy to realize by hardware. Considering delay and channel estimation error, the BER performance of adaptive TuCM is analyzed and simulated. In the performance analysis, the method of data fitting is applied to obtain the BER expression for TuCM, and a fitting mathematical model is proposed. Results show that adaptive TuCM is very sensitive to delay and channel estimation error. To alleviate these problems, we proposed an improved power adaptation that can make adaptive TuCM practical.

We present a design and implementation of a QoS control mechanism in an Adaptive Multimedia Communication System (AMCS) using multiagent-based computing technology. In this paper, we first define functional requirements for AMCS. Subsequently we describe the design and implementation of AMCS with a knowledge-based multiagent framework to fulfill the functional requirements. Moreover we evaluate the adaptability of the prototype systems of AMCS with the operational situations observed in its experiments. From the result of the experiments, we conclude that the multiagent-based design and implementation is reasonable for construction of AMCS.

A pilot needs operational information about wind over sea as well as wave height to provide safety of hydroplane landing on water. Near-surface wind speed and direction can be obtained with an airborne microwave scatterometer, a radar designed for measuring the scatter characteristics of a surface. Mostly narrow-beam antennas are applied for such wind measurement. Unfortunately, a microwave narrow-beam antenna has considerable size that hampers its placement on flying apparatus. In this connection, a possibility to apply a conventional airborne radar altimeter as a scatterometer with a nadir-looking wide-beam antenna in conjunction with simultaneous range Doppler discrimination techniques for recovering the wind vector over sea at low speed of flight is discussed, and measuring algorithms of sea surface wind speed and direction are proposed. The principle considered and algorithms proposed in the paper can be used for creation an airborne radar system for operational measurement of the sea roughness characteristics and for safe landing of a hydroplane on water.

Adaptive arrays have been recognized as an attractive mean for overcoming multipath fading and interference rejection in the field of mobile communications. In, an adaptive array applicable to single-user spread spectrum (SS) systems has been proposed. In this system, any a priori information concerning incoming signal, even the spreading code and synchronization, is not necessary while it achieves equalizing, beamforming and despreading of a received signal, simultaneously. In this paper, we propose a multistage blind adaptive array antenna based on the above-mentioned adaptive algorithm in order to realize blind signal processing that is applicable to multi-user SS systems. Behavior and performance of the proposed multistage system are examined through computer simulations.

Weight divided adaptive control method for a microwave FeedForward Power Amplifier (FFPA) is presented. In this adaptive controller, an output signal of a power amplifier is used as reference signal. Additionally, reference signal is divided by the weight of adaptive filter, so that characteristics of the power amplifier, such as temperature dependence, do not have influence on the convergence performances. The proposed adaptive algorithm and the convergence condition are derived analytically and we clarify that the proposed weight divided adaptive algorithm is more stable than the conventional Normalized Least Mean Square (NLMS) algorithm. Then, the convergence condition considering phase calibration error is discussed. The effectiveness of the proposed algorithm are also verified by the nonlinear simulations of the FFPA having AM-AM and AM-PM nonlinearity of GaAsFET.

This paper describes a new training method of average voice model for speech synthesis in which arbitrary speaker's voice is generated based on speaker adaptation. When the amount of training data is limited, the distributions of average voice model often have bias depending on speaker and/or gender and this will degrade the quality of synthetic speech. In the proposed method, to reduce the influence of speaker dependence, we incorporate a context clustering technique called shared decision tree context clustering and speaker adaptive training into the training procedure of average voice model. From the results of subjective tests, we show that the average voice model trained using the proposed method generates more natural sounding speech than the conventional average voice model. Moreover, it is shown that voice characteristics and prosodic features of synthetic speech generated from the adapted model using the proposed method are closer to the target speaker than the conventional method.

Adaptive arrays with signal blocking have the advantages of fast convergence and robustness to pointing errors as well as of rejecting a coherent interference in addition to incoherent ones. In this paper, we propose a novel method for performance improvement in such arrays with no increase in complexity. The proposed method utilizes all of the array elements to obtain the adaptive output so that its performance is superior to that of the conventional method which does not utilize one of the elements. Their performances are compared analytically and by computer simulation.

The system uses spectral shaping of a supercontinuum source followed by wavelength-to-time mapping to generate ultra wideband RF waveforms with arbitrary modulation. It employs an adaptive computer control to mitigate the non-ideal features inherent in the optical source and in the spectrum modulation process. As proof of concept, ultra-wideband frequency hopped CDMA waveforms are demonstrated.

In order to increase the capacity of a DS-CDMA system, several kinds of interference suppression techniques have been studied, such as multiple access interference (MAI) cancellers and adaptive array antennas. However, their performance tends to degrade in high traffic-load situations. To compensate for the degradation, a receiver cascading an adaptive array antenna and a multistage parallel interference canceller (PIC) is studied in this paper. This receiver first uses an adaptive array antenna to suppress interference signals spatially, and uses a multistage PIC to suppress in-beam interference effectively. The performance of the cascaded receiver is evaluated with two schemes for antenna weight generation by computer simulations assuming a Rayleigh-distributed L-path channel. When antenna weights are generated for each user by an LMS algorithm, the cascaded receiver has shown better performance at the cost of a large number of pilot symbols and symbol by symbol weight update. Its performance degradation is 2.8 dB at the BER of 10-4 even when the number of users increases from one to 24. On the other hand, when antenna weights are generated for each path by a DMI algorithm, its performance is degraded due to the inaccurate weight generation which occurs when the SINR of the desired signal is small. This degradation can be mitigated by using all signals of the desired user received by all antenna patterns of desired user for RAKE combining when the difference among arrival angles of the paths of the desired user is small.