The present study investigated the human ability to selectively process pictures and words in free recall. We explored whether successful bias towards a subset of priority items occurs at the expense of the remaining items-i.e., whether successful priority item bias necessitates the dumping of information related to non-priority items. It has been shown that an increase in the percentage of correct recalls to items given priority in the pre-test instructions induces a decrease in the percentage of correct recalls for non-priority items. Even in a free recall experimental paradigm, the information dumping phenomenon was observed. However, there were no effects of stimulus presentation time and stimulus modality (picture vs. word) on the percentage of correct recalls detected.

This paper proposes a protocol to support mobile hosts in IPv6 by introducing a new addressing architecture and a new hop-by-hop option. This protocol also allows a mobile host to communicate with another host via a firewall machine which drops packets from untrustworthy hosts. The new addressing scheme is based on the separation of the identifier and the location of a mobile host. This is a straightforward implementation of the basic concept of VIP, a protocol providing seamless mobility in IPv4. The new hop-by-hop option of IPv6 allows a firewall machine to authenticate the source host of the forwarded packet with negligible overhead. The author plans to implement this protocol on several operating systems in the near future.

In this paper we propose two methods, named the time smoothing and the scale smoothing respectively, to reduce the Gibbs overshoot in continuous wavelet transform. In is shown that for a large kind of wavelets the scale smoothing cannot remove the Gibbs overshoot completely as in the case of Fourier analysis, but it is possible to reduce the overshoot for any wavelets by choosing the smoothing window functions properly. The frequency behavior of scale smoothing is similar to that of the time smoothing. According to its frequency behavior we give the empirical conditions for selecting the smoothing window functions. Numerical examples are given for illustrations.

This paper presents a generating method of LSP parameter sequences for speech synthesis by rule. In our method, neural networks are schemed to generate LSP parameter sequences of Vowel-Consonant-Vowel (VCV) units. The quality of synthesized speech by concatenation way of VCV units through table-look-up technique can not be improved so much owing to the distortion appearing on VCV units junction. In our method, the neural networks concatenate VCV units step by step with less distortion on VCV units junction, which synthesizes good quality speech.

This paper presents a new method for constructing orthonormal wavelet bases with vanishing moments based on general IIR filters. It is well-known that orthonormal wavelet bases can be generated by paraunitary filter banks. Then, synthesis of orthonormal wavelet bases can be reduced to design of paraunitary filter banks. From the orthonormality and regularity of wavelets, we derive some constraints to IIR filter banks, and investigate relations between the constrained filter coefficients and its zeros and poles. According to these relations, we can apply Remez exchange algorithm in stopband directly, and formulate the design problem in the form of an eigenvalue problem. Therefore, a set of filter coefficients can be easily computed by solving the eigenvalue problem, and the optimal filter coefficients with an equiripple response can be obtained after applying an iteration procedure. The proposed procedure is computationally efficient, and the number of vanishing moments can be arbitrarily specified.

A new Neural Network Equalizer (NNE), employing multilayer feedforward neural network, is proposed as a compensation method for nonlinear and multipath distortion that arises from FTTA (Fiber To The Air) system. If a signal in a channel is affected by nonlinear distortion, the conventional Decision Feedback Equalizer (DFE) finds difficulty in perfect compensation of it. To compensate for nonlinear distortion as well as multipath distortion, an equalizer, employing neural network, is investigated. A new neural network equalizer, yielding a cubic function as unit output function, is proposed in order to compensate the nonlinear distortion effectively. We also propose an initial weights of neural network for preventing from local minimum. Computer simulation results show that the compensation performance of the new NNE is superior to the conventional DFE and the conventional NNE.

In this paper, we prove that for a class of nonsymmetric neural networks with connection matrices T having nonnegative off-diagonal entries, -T is an M-matrix is a necessary and sufficient condition for absolute exponential stability of the network belonging to this class. While this result extends the existing one of absolute stability in Forti, et al., its proof given in this paper is simpler, which is completed by an approach different from one used in Forti, et al. The most significant consequence is that the class of nonsymmetric neural networks with connection matrices T satisfying -T is an M-matrix is the largest class of nonsymmetric neural networks that can be employed for embedding and solving optimization problem with global exponential rate of convergence to the optimal solution and without the risk of spurious responses. An illustrating numerical example is also given.

In the field of speech recognition, many researchers have proposed speech recognition methods using auditory information like acoustic signal or visual information like shape and motion of lips. Auditory information has valid features for speech recognition, but it is difficult to accomplish speech recognition in noisy environment. On the other side, visual information has advantage to accomplish speech recognition in noisy environment, but it is difficult to extract effective features for speech recognition. Thus, in case of using either auditory information or visual information, it is difficult to accomplish speech recognition perfectly. In this paper, we propose a method to fuse auditory information and visual information in order to realize more accurate speech recognition. The proposed method consists of two processes: (1) two probabilities for auditory information and visual information are calculated by HMM, (2) these probabilities are fused by using linear combination. We have performed speech recognition experiments of isolated words, whose auditory information (22.05kHz sampling, 8-bit quantization) and visual information (30-frame/s sampling, 24-bit quantization) are captured with multi-media personal computer, and have confirmed the validity of the proposed method.

This paper describes a new multiple DmB1C (Differential m Binary 1 Complement insertion) /DmB1M (Differential m Binary with 1 Mark insertion) coding scheme for high-speed optical multiplex transmission. The coding scheme has the characteristics of small consecutive identical digits and a good balance between marks and spaces. Furthermore, it has also good synchronization characteristics and higher flexibility for extension to high capacity transmission than the conventional mB1C or DmB1M coding schemes. We describe a design methodology for a multiplex transmission system using the proposed coding scheme, and verify the characteristics of the proposed coding scheme using an experimental setup of a 2.8 Gbit/s serial optical interconnection circuit, which has 16 parallel 156 Mbit/s inputs. The coding scheme realizes transmission systems with simple analog circuit configuration, and small digital circuit complexity with wide dynamic range and good mark ratio tolerance.

The finite reconfigurability and local reconfigurability of graphs were proposed by Sha and Steiglitz [1], [2] in connection with a problem of on-line reconfiguraion of WSI networks for run-time faults. It is shown in [2] that a t-locally-reconfigurable graph for a 2-dimensional N-vertex array AN can be constructed from AN by adding O(N) vertices and edges. We show that Ω(N) vertices must be added to an N-vertex graph GN in order to construct a t-locally-reconfigurable graph for GN, which means that the number of added vertices for the above mentioned t-locally-reconfigurable graph for AN is optimal to within a constant factor. We also show that a t-finitely-reconfigurable graph for an N-vertex graph GN can be constructed from GN by adding t vertices and tN + t (t+1)/2 edges.

When moving images are displayed on color PDPs, motional artifacts such as disturbances of gray scales and colors are often observed. Reduction of the disturbances is essential in achieving PDPs with acceptable picture quality for TV use. The moving picture quality has been improved by using a modified-binary-coded light-emission-period scheme and a 3dimensional (2D in space and 1D in time) scattering technique. In the 10-sub-field modified-binary-code scheme for 256 gray level expression, sub-field B (of period equivalent to 64) and C (128) of conventional 8-sub-field binary-coded scheme are added and then re-distributed into four sub-fields D (48). The modifiedbinary-coded scheme therefore has the light-emitting-period ratio 1:2:4:8:16:32:48:48:48:48. The maximum period, 128 of the conventional, is reduced to 48. By using the modified-binary-coded scheme, the motional artifacts are reduced significantly, but still perceptible because they appear in forms of continuous lines. In order to make the disturbance less conspicuous, a 3D scattering technique is introduced. The technique has been made possible because of the redundancies of the modified-binary-coded scheme: namely, (1) the position of sub-field-block A (63) can be placed at one of the five positions among four sub-fields D (48), (2) there are various choices when newly assigning one of the four sub-fields D, (3) one can arbitrarily choose whether or not to assign a new sub-field D between the gray levels 48-63, 96-111, 144-160, and 192-207. By randomly selecting one of these emission patterns, the disturbances change their forms from continuous lines to scattered dots. The randomization can be performed at each horizontal line of the display, at each vertical line, at each pixel, of at each TV field. An appreciable improvement of moving picture quality has been realized without influencing the still image.

A new cost function based on multi-band decomposition of the estimation error and application of a different step-size for each band is used in connection with the least-mean-square criterion to improve the fidelity of estimates as compared to those obtained with conventional least-mean-square adaptive algorithms. The basic new idea is to trade off time and frequency resolutions of the adaptive algorithm along the frequency domain by using different step-sizes in the analysis of distinct frequencies in accordance with the frequency-localized statistical behavior of the imput signal. The mathematical background for a stochatic approach to the multi-band decomposition-based scheme is presented and algorithms with fixed and variable step-sizes are derived. Computer experiments compare the performance of multiband and conventional least-mean-square methods when applied to system identification.

In this letter, the absolute exponential stability result of neural networks with asymmetric connection matrices is obtained, which generalizes the existing one about absolute stability of neural networks, by a new proof approach. It is demonstrated that the network time constant is inversely proportional to the global exponential convergence rate of the network trajectories to the unique equilibrium. A numerical simulation example is also given to illustrate the obtained analysis results.

UNcorrectable Bit Errors (UNBEs) are important in considering the reliability of Redundant Array of Inexpensive Disks (RAID). They, however, have been ignored or have not been studied in detail in existing reliability analysis of RAID. In this paper, we present an analytic model to study the reliability of declustered-parity RAID by considering UNBEs. By using the analytic model, the optimistic and the pessimistic estimates of the probability that data loss occurs due to an UNBE during the data reconstruction after a disk failed (we call this DB data loss) are obtained. Then, the optimistic and the pessimistic estimates of the Mean Time To Data Loss (MTTDL) that take into account both DB data loss and the data loss caused by double independent disk failures (we call this DD data loss) are obtained. Furthermore, how the MTTDL depends on the number of units in a parity stripe, rebuild time of a failed disk and write fraction of data access are studied by numerical analysis.

Obradovic and Parberry showed that any n-input k-ary function can be computed by a depth 4 unit-weight k-ary threshold circuit of size O(nkn). They also showed that any n-input k-ary symmetric function can be computed by a depth 6 unit-weight k-ary threshold circuit of size O(nk+1). In this paper, we improve upon and expand their results. The k-ary threshold circuits of nonunit weight and unit weight are considered. We show that any n-input k-ary function can be computed by a depth 2 k-ary threshold circuit of size O(kn-1). This means that depth 2 is optimal for computing some k-ary functions (e.g., a PARITY function). We also show that any n-input k-ary function can be computed by a depth 3 unit-weight k-ary threshold circuit of size O(kn). Next, we show that any n-input k-ary symmetric function can be computed by a depth 3 k-ary threshold circuit of size O(nk-1), and can be computed by a depth 3 unit-weight k-ary threshold circuit of size O(knk-1). Finally, we show that if the weights of the circuit are polynomially bounded, some k-ary symmetric functions cannot be computed by any depth 2 k-ary threshold circuit of polynomial-size.

This paper compares three cell transfer quality control schemes, namely HPS, DAS and ORS, which integrate a preventive congestion control and a reactive congestion control in ATM switch. Simulation results showed that ORS achieves the largest network utilization, and HPS provides enough large throughput compared with DAS only when many VBR connections are multiplexed.

Recently, the number of users utilizing mobile communication services has increased greatly in many information and communication fields. In the future, the number of mobile communication system users will increase even faster, until the rate of diffusion ultimately reaches that of telephones. The day that each person has his own portable mobile terminal is not so far off. Moreover, the systems will not only be used as telephones but also as mobile computing for multimedia information. Digitalization technologies of mobile communication systems needed to realize such mobile computing will be introduced in this paper.

Given the tremendous growth in the cellular phone system and the Personal Hadny-phone System (PHS), it is to be expected that demands for mobile computing using those wireless infrastructures, that is mobile computer access, will dramatically increase. This paper describes high-quality and high-speed data transmission technology for PHS mobile computing and current PHS data transmission standardization activities. Furthermore, wireless agent communication and a service example are presented together with the concept of background communication for the coming wireless multimedia services.

Many network protocols for routing messages have been proposed for mobile computing environments. In this paper, we consider the query processing strategy which operates over these network protocols. To begin with, we introduce five fundamental location update methods based on ideas extracted from the representative network protocols. They are the single broadcast notification (SBN), the double broadcast notification (WBN), the single default notification (SDN), the double default notification (WDN), and the no notification (NN). As a network protocol, each method is strong in performance in some system enrivonment, but weak in others. In practical situations, where various kinds of applications are used for various purposes, however, it is required to use a single method. We therefore propose an adaptive query processing strategy where these five location update methods can be dynamically selected. Moreover, we analyze the performance of this adaptive query processing strategy via the Markov chain. We also use the statistical approach to estimate the traffic of individual hosts. Finally, we show the efficiency of our proposed strategy over a wide area of system environments.

This paper discusses a design method for two-dimensional (2-D) periodically time-variant digital filters (PTVDFs) whose filter coefficients vary periodically. First, 2-D periodicities for a variation of filter cefficients are considered, from which two and four-phase variations of coefficients are shown to be suitable for practical applications. Then, the input-output relation (transfer function) for 2-D separable-denominator (SD) PTV DFs is derived, which results in a linear combination of the baseband input signal and its modulated versions. Finally, in order ro approximate given filter specifications, the structure for 2-D SD PTV DFs is given and a design method is proposed. It is shown that, compared with the 2-D SD time-invariant DFs, approximation error can be reduced with the proposed SD PTV DFs.