The Direct-Sequence Code Division Multiple Access (DS-CDMA) is one of the most likely candidates for the 3rd generation mobile communication systems. On the other hand, a network supporting integrated data transfer and providing predictable qualities of service (QoS) to such application is an attractive issue. This paper proposes several schemes based on the majority rule concerned with resource reservation to investigate the performance of a CDMA cellular system. The admission control is based on the residue capacity and required signal-to-noise ratio (SNR) for multimedia traffic including voice, data and video. The system performance is explored with variable spreading gain and multi-code CDMA systems under the proposed schemes. Moreover, several system parameters including the handoff probability (lifetime in a cell) and the traffic load are investigated to study the impact to system performance. The results show the trade-off between the blocking probability of new calls and dropping probability of handoff calls under diversified situations. However, it is inefficient under resource reservation due to unused capacity. In this paper, we also propose a resource-borrowing scheme to improve system performance. Simulation results show that our proposed algorithm really improves the system performance in a multimedia CDMA cellular system with resource reservation.

In this paper, a medium access control protocol is proposed for the integrated voice and data services in the local CDMA communication systems. Based on WB-TD/CDMA, uplink channels for the proposed protocol are composed of time slots with multiple spreading codes for each slot. During a talkspurt, a voice terminal transmits its entire packets over a reserved code. On the other hand, a data terminal transmits its packet after sensing the spreading code status. The base station broadcasts the status of spreading codes for each slot. In this protocol, voice packets never contend with data packets. The numerical results show that this protocol increases the system capacity for voice service by applying the reservation scheme. Despite the low access priority of data terminal, the data traffic performance can be increased in proportion to the number of spreading codes.

In this paper, we first investigate the problems of the existing branch point algorithms for available bit rate (ABR) multicast connections in ATM networks, and then propose various solutions for resolving those problems. By combining these solutions, we also propose a new efficient and scalable branch point algorithm. In the proposed algorithm, each branch point stores the feedback information on a per-branch basis for each virtual connection and only passes BRM cells returning from the farthest destination. Simulation results show that the proposed algorithm can provide a good fairness, a higher efficiency and an excellent scalability, compared with the existing algorithms.

The third generation cellular system will provide high bandwidth data service and multimedia applications are proposed on the high-speed data link. On the other hand, cellular handsets are getting equipped with a short-range radio communication device that is mainly targeted to connect cellular handsets to computers. In this paper, we propose a percolating data delivery mechanism for a cellular-ad hoc integrated network utilizing multicast/broadcast communication, which is endorsed by data synchronization, and single-hop ad hoc networks for information shower/exchange services. The mechanism lessens data traffic in both the cellular and the ad hoc network for data delivery and tolerates unexpected disconnection in the ad hoc network. The mechanism also compensates data delivery in areas out of the cellular service. Our implementation of the data delivery mechanism utilizes Bluetooth for an ad hoc network configuration, and delivers data in the cellular network to the Bluetooth ad hoc network via the data synchronization mechanism. We evaluate communication traffic and delivery time on the prototype system and discuss about the efficiency of the mechanism.

In any ill-conditioned information-transfer system, as in long-distance communication, we often must construct feedback confirmation channels, in order to confirm that informations received at destinations are correct. Unfortunately, for such systems, undesirable uncertain fluctuations may be induced not only into forward communication channels but also into feedback confirmation channels, and it is such difficult that transmitters always confirm correct communications. In this paper, two fuzzy-set-valued mappings are introduced into both the forward communication channel and the feedback confirmation channel, separately, and overall system-behaviors are discussed from the standpoint of functional analysis, by means of fixed point theorem for a system of generalized equations on fuzzy-set-valued mappings. As a result, a good mathematical condition is successfully obtained, for such information-transfer systems, and fine-textured estimations of solutions are obtained, at arbitrary levels of values of membership functions.

If there exist any two vertices in G whose distance becomes longer when a vertex u is removed, then u is defined as a hinge vertex. Finding the set of hinge vertices in a graph can be used to identify critical nodes in an actual network. A number of studies concerning hinge vertices have been made in recent years. In general, it is known that more efficient sequential or parallel algorithms can be developed by restricting classes of graphs. For instance, Chang et al. presented an O(n+m) time algorithm for finding all hinge vertices of a strongly chordal graph. Ho et al. presented a linear time algorithm for all hinge vertices of a permutation graph. In this paper, we shall propose a parallel algorithm which runs in O(log n) time with O(n) processors on CREW PRAM for finding all hinge vertices of an interval graph.

In cellular networks, the system performance can be degraded because of the inefficient use of channels with the two types of traffic, that is, the new call traffic and handover call traffic. This problem can be alleviated if the channels are used efficiently for new call traffic and handover call traffic. In this paper, we consider the proper ratio of the number of channels for new calls to the number of total channels denoted as α. We introduce the new call channel limiting technique incorporating velocity of mobile user. We model one-dimensional cellular networks and evaluate the performance in terms of the blocking probabilities, the probability of call dropping and the probability of incomplete call. We show with numerical examples that the system performance can be improved by selecting the appropriate α for various velocities of mobile users.

Conventional spatial transform based motion estimation algorithms are not practical because of their heavy computational loads. In this paper, we proposed motion estimation method with variable grid size, which is more efficient than conventional spatial transform based methods and gives better PSNR performance than conventional BMA.

Equivalent-time sampling is a well-known technique to capture repetitive signals at finer time intervals than a sampling clock cycle time and it is widely used to implement waveform measurement with high time resolution. There are three techniques for implementing its time base (i.e., sequential sampling, random sampling and coherent sampling), and they have their respective advantages and disadvantages. In this paper we propose a new coherent sampling system which incorporates a pretrigger and time jitter reduction function for a fluctuating input signal which a random sampling system has, while maintaining the waveform recording efficiency of a conventional coherent sampling system. We also report on a technique for measuring a reference trigger time period accurately which is necessary to implement the proposed sampling system, and show its effectiveness through numerical calculations of its data recording time.

For the purpose of equalisation of rapidly time variant multipath channels, we derive a novel adaptive algorithm, the amplitude banded LMS (ABLMS), which implements a non-linear adaptation based on a coefficient matrix. Then we develop the ABLMS algorithm as the adaptation procedure for a linear transversal equaliser (LTE) and a decision feedback equaliser (DFE) where a parallel adaptation scheme is deployed. Computer simulations demonstrate that with a small increase of computational complexity, the ABLMS based parallel equalisers provide a significant improvement related to the conventional LMS DFE and the LMS LTE in the case of a second order Markov communication channel model.

The variabilities of device characteristics are usually regarded as a normal distribution. If we consider the variabilities over the whole wafer, however, they cannot be expressed as a normal distribution due to the existence of global systematic component. We propose a statistical model, characterizing the global systematic component according to the distance from the center of the wafer, which can express the variabilities over the whole wafer statistically.

A direct conversion receiver IC including an on-chip balun, an I/Q mixer, a variable gain amplifier and a 90 phase-shifter is fabricated in a Bi-CMOS technology with 15 GHz transition frequency (fT). This paper demonstrates that cascaded connection of an on-chip balun and a double balanced mixer as the I/Q mixer is effective to achieve a low DC offset and a low second-order distortion, on the basis of both careful examination of the mixer behavior and measurement. Input-referred DC offset voltage of less than 300 µV and spurious free dynamic range (SFDR) of over 67 dB are obtained by measurement. The IC consumes 52 mA from 2.7 V power supply voltage. The die size is 3 mm 3 mm.

This paper describes a 1-GHz portable digital delay-locked loop (DLL) with 0.15-µm CMOS technology. There are three factors contributing to jitter in digital DLLs. One is supply-noise induced jitter, another is jitter caused by delay time resolution and phase step in the delay line, and the third is jitter caused by the sensitivity of the phase detector. In order to achieve a low jitter digital DLL, we have developed a master-slave architecture that achieves infinite phase capture ranges and low latency, a delay line that improves the delay time resolution, a phase step suppression technique and a dynamic phase detector with increased sensitivity. These techniques were used to fabricate a digital DLL with improved jitter performance. Measured results showed that the DLL successfully achieves 29-ps peak-to-peak jitter with a quiet supply and 0.2-ps/ mV supply sensitivity.

A speaker adaptation technique that maximizes the observation probability of an input speech is proposed. It is applied to semi-continuous hidden Markov model (SCHMM) speech recognizers. The proposed algorithm adapts the mean µ and the covariance Σ iteratively by the gradient search technique so that the features of the adaptation speech data could achieve maximum observation probabilities. The mixture coefficients and the state transition probabilities are adapted by the model interpolation scheme. The main advantage of this scheme is that the means and the variances, which are common to all states in SCHMM, are adapted independently from the other parameters of SCHMM. It allows fast and precise adaptation especially when there is a large acoustic mismatch between the reference model and a new speaker. Also, it is possible that this scheme could be adopted to other areas which use codebook. The proposed adaptation algorithm was evaluated by a male speaker-dependent, a female speaker-dependent, and a speaker-independent recognizers. The experimental results on the isolated word recognition showed that the proposed adaptation algorithm achieved 46.03% average enhancement in the male speaker-dependent recognizer, 52.18% in the female speaker-dependent recognizer, and 9.84% in the speaker-independent recognizer.

The transmission S-parameter, S21, between dipole elements on a rectangular finite ground plane is calculated by the MoM with planar-segments in the horizontally and vertically polarized configurations. Supposed a 1/10 scaling, the frequency range is selected 0.15-0.8 GHz. The size of the finite ground plane is 40 cm 100 cm. The dipole-element length is 18.8 cm (half-wavelength at 0.8 GHz). The distance between dipole elements is 30 cm. The results are compared to the calculated results with the conventional MoM-GTD hybrid method and also the measured results with a TRL-calibrated network analyzer. It makes clear that the MoM-GTD hybrid method is not applicable to a small ground plane in the vertically polarized configuration. The results calculated by the MoM with planar-segments agree well to the measured results both in the horizontal and vertical polarizations. The results show that the size of the finite ground plane for the vertical polarization should be much larger than for the horizontal polarization.

In this paper, we investigate the performance of Fast Reservation Protocols (FRP) for burst-level bandwidth allocation in ATM networks. FRP schemes can be classified into delayed transmission (DT) and immediate transmission (IT) methods according to reservation procedure. Moreover, according to the responsibility for negative acknowledgment (NAK) cell generations when burst blocking occurs, FRP schemes can be further classified into blocking node NAK (BNAK) and destination node NAK (DNAK) schemes. We analize the FRP schemes with different reservation and NAK methods for single node and multihop network models, respectively. We then discuss the dependence of performance for each FRP scheme on propagation delay, peak rate, and the number of hops.

The structural characteristics of clusters are investigated in the partitioning process. Two partition functions, which show opposite properties around the optimal cluster number, are found and a new cluster validity index is presented based on the combination of these functions. Some properties of the index function are discussed and numerical examples are presented.

This paper, for the first time, presents a provably secure signature scheme with message recovery based on the elliptic-curve discrete logarithm. The proposed scheme is proven to be secure in the strongest sense (i.e., existentially unforgeable against adaptively chosen message attacks) in the random oracle model under the discrete logarithm assumption. We give a concrete analysis of the security reduction. When practical hash functions are used in place of truly random functions, the proposed scheme is almost as efficient as the elliptic-curve version of the Schnorr signature scheme and existing schemes with message recovery such as the elliptic-curve version of the Nyberg-Rueppel and Miyaji schemes.

This paper is, in half part, written in review nature, and presents recent theoretical results on linear-filtering and -prediction problems of nonstationary Gaussian processes. First, the basic concepts, signal and noise, are mathematically characterized, and information sources are defined by linear stochastic differential equations. Then, it is shown that the solution to a conventional problem of filtering or prediction of a nonstationary time series is, in principle, reducible to a problem, of which solution is given by Kalman-Bucy's theory, if one can solve a problem of finding the canonical representation of a Gaussian process such that it has the same covariance functions as those of the time series under consideration. However, the problem mentioned above is left open. Further, the problem of time-frequency analysis is discussed, and physical realizability of the evolutionary, i.e., the online, spectral analyzer is shown. Methods for dealing with differential operators are presented and their basic properties are clarified. Finally, some of related open problems are proposed.

This paper describes an application of a speech translation system to another task/domain in the real-world by using developmental data collected from real-world interactions. The total cost for this task-alteration was calculated to be 9 Person-Month. The newly applied system was also evaluated by using speech data collected from real-world interactions. For real-world speech having a machine-friendly speaking style, the newly applied system could recognize typical sentences with a word accuracy of 90% or better. We also found that, concerning the overall speech translation performance, the system could translate about 80% of the input Japanese speech into acceptable English sentences.