Packet Reservation Multiple Access (PRMA) is emerging as a possible multiple access scheme for the forth-coming Personal Communication systems, due to its inherent flexibility and to its capability to exploit silence periods to perform a statistical multiplexing of traffic sources, often characterised by a high burstiness. On the other hand, the current trend in reducing cell sizes and the more complex traffic scenarios pose major planning problems, which are best coped with by adaptive allocation schemes. The identification of adaptive schemes suitable to operate on a shorter time scale, which is typical of packetised information, disclose a number of problems which are addressed in this paper. A viable solution is provided by a well-known self-adaptive assignment method (Channel Segregation), originally developed for FDMA systems, provided it is conveniently adapted for PRMA operation. Simulations show good performance, provided that values of some system variables are correctly chosen. These results encourage further studies in order to refine adaptive methods suitable for cellular, packet switched personal communications systems.

The word error probability of linear block codes is computed for diversity systems with maximal ratio combining in mobile communications with three decoding algorithms: error correction (EC), error/erasure correction (EEC), and maximum likelihood (ML) soft decoding algorithm. Ideal interleaving is assumed. EEC gives 0.1-1.5dB gain over EC. The gain of EEC over EC decreases as the number of diversity channels increases. ML soft gives 1.8-5.5dB gain over EC.

The goal of this paper is to propose a new symbolic model checking approach named time-space modal model checking, which could be applicable to verification of bit-slice microprocessor of infinite bit width and one dimensional systolic array of infinite length. A simple benchmark result shows the effectiveness of the proposed approach.

In the course of the development of the "Hyper Hospital," a novel medical care system constructed in the computerized information network using virtual reality as its human interface, we devised a virtual reality world creating system which allows users to con figure the world interactively. The re-configuration of the virtual world was designed to be carried out without interruptions of activity and the world can continue to exist during the reconfiguration process. This facility comprises an important part of our Hyper Hospital system because one of our major goals of this proposal of the Hyper Hospital is to restore maximum freedom for patients in the medical care system. Discussion was given in the present study with respect to the basic requirements of the system to be realized, including discussions on the permission given to the participants of different levels, and means by which to modify the structure of the virtual world. A preliminary implementation was described following this general consideration. The developed prototype was shown to be practically suitable to the test of our virtual environment applied to realistic medical scenes.

We consider slotted ALOHA systems with a controlled output power level. The systems were proposed to improve the throughput performance by the capture effect. However widely used linear modulation systems have no capture effect, and a power level distribution dominates the performance in those systems. In this paper we consider linear modulation systems employing PSK. We introduce an average error probability of the highest power signal as a performance measure, and a uniform distribution is applied to the error probability analysis. Numerical results show the superiority of the systems with uniform distribution to a conventional slotted ALOHA in a heavy traffic condition. On the other hand, in a light traffic condition, the optimal power distribution which minimizes the error probability is obtained for 2-level ALOHA. We also propose the power level selection method to search the optimal power level. The validity of analytical results are confirmed by simulations.

The subject is the horizontal coil's temperature rise in DY for high frequency by being unavoidable for the tendency of more information on display monitor equipments. Writers made the temperature-balance model from a point of view that this temperature rise is coming from the heat rise and the conductivity, and we expressed the temperature rise of DY by using amount of the heat rise and conductivity characteristics of each element. Also, we indicated the method to decide about the selection of the wire size of coils, the section area and deflection sensitivity, with regard to reducing the temperature rise. We confirmed the effect of the temperature rise reduction by about 9 on products, under the condition of 64 kHz horizontal frequency.

In the final stages of VLSI testing, improved quality VLSI testing is an important subject for ensuring reliability in the forwarded VLSI market. On the other hand, developments in high integration technology have resulted in an increased number of functional blocks in VLSI devices and an increased number of gates for each terminal. Consequently, it has become more difficult to improve the quality of VLSI tests. We have developed a new test method in addition to conventional testing methods intended for improving the test coverage in VLSI tests. This new test method analyzes the relationship between IDDq (Quiescent Power Supply Current) of DUT and DUT failure by applying the concept of the toggle rate. Accordingly, in this paper we report that the results of IDDq testing confirm a correlation with defect level.

This paper describes a flexible point-to-multipoint access protocol for the fiber-optic passive double star (PDS) system. To provide various types of services, and permit flexibility in changing transport capacity, a time division multiple access (TDMA) scheme for the PDS system is considered. Dynamic time slot multiplexing based on TDMA is proposed to provide required time slots efficiently according to service changes. The effectiveness of dynamic time slot multiplexing is calculated and compared to fixed time slot multiplexing for telephony services. A TCM/TDMA frame structure and an access protocol enabling dynamic time slot multiplexing are proposed. ONU bandwidth is dynamically assigned by using a set of pointers. The ONU access protocol causes no interruption to operating ONUs on the same PDS system during the configuration or reconfiguration of an ONU. The access time is analyzed to estimate the performance of the access protocol. The probability density of access time is calculated for the number of ONUs connected. The calculation results indicate that a PDS system can accommodate up to around 60 ONUs within the maximum access time specified by ITU-T. The experimental results also agree fairly well with the theoretical values.

Ever increasing demand for higher bandwidth memories, which is fueled by multimedia and 3D graphics, seems to be somewhat satisfied with various emerging memory solutions. This paper gives a review of these emerging DRAM architectures and a performance comparison based on a condition to let the readers have some perspectives of the future and optimized graphics systems.

This paper presents techniques used in combinational test generation for multiple stuck-at faults using the parallel vector pair analysis. The techniques accelerate a test generation procedure previously proposed and reduce the number of test vectors generated, while higher fault coverage is derived. The first technique proposed in this paper, which is applied at the first phase of test generation, is rules of ordering vector pairs to be analyzed, to derive high fault coverage without repeating the analysis for the same vector pairs. The second one is to generate new vector pairs for undetected faults, instead of random vector pairs. Both techniques are based on the idea that faults close to primary inputs should be detected earlier than close to primary outputs. The third technique proposed here is how to construct vector pairs from one input vector in order to accelerate test generation especially for circuits with many primary inputs and scan flip-flops. Experimental results for bench-mark circuits show the effectiveness of the techniques.

This paper establishes a relationship among the accepting powers of deterministic, nondeterministic, and alternating one-way auxiliary pushdown automata, for any tape bound below n. Some other related results are also presented.

A novel adaptive pitch-synchronous analysis method is proposed to estimate simultaneously vocal tract (formant/antiformant) and voice source parameters from speech waveforms. We use the parametric Rosenberg-Klatt (RK) model to generate a glottal waveform and an autoregressive-exogenous (ARX) model to represent voiced speech production process. The Kalman filter algorithm is used to estimate the formant/antiformant parameters from the coefficient of the ARX model, and the simulated annealing method is employed as a nonlinear optimization approach to estimate the voice source parameters. The two approaches work together in a system identification procedure to find the best set of the parameters of both the models. The new method has been compared using synthetic speech with some other approaches in terms of accuracy of estimated parameter values and has been proved to be superior. We also show that the proposed method can estimate accurately the parameters from natural speech sounds. A major application of the analysis method lies in a concatenative formant synthesizer which allows us to make flexible control of voice quality of synthetic speech.

This paper describes a method for predicting the user's next utterances in spoken dialog based on the topic transition model, named TPN. Some templates are prepared for each utterance pair pattern modeled by SR-plan. They are represented in terms of five kinds of topic-independent constituents in sentences. The topic of an utterance is predicted based on the TPN model and it instantiates the templates. The language processing unit analyzes the speech recognition result using the templates. An experiment shows that the introduction of the TPN model improves the performance of utterance recognition and it drastically reduces the search space of candidates in the input bunsetsu lattice.

We discuss how to determine automatically the number of mixture components in continuous mixture density HMMs (CHMMs). A notable trend has been the use of CHMMs in recent years. One of the major problems with a CHMM is how to determine its structure, that is, how many mixture components and states it has and its optimal topology. The number of mixture components has been determined heuristically so far. To solve this problem, we first investigate the influence of the number of mixture components on model parameters and the output log likelihood value. As a result, in contrast to the mixture number uniformity" which is applied in conventional approaches to determine the number of mixture components, we propose the principle of distribution size uniformity". An algorithm is introduced for automatically determining the number of mixture components. The performance of this algorithm is shown through recognition experiments involving all Japanese phonemes. Two types of experiments are carried out. One assumes that the number of mixture components for each state is the same within a phonetic model but may vary between states belonging to different phonemes. The other assumes that each state has a variable number of mixture components. These two experiments give better results than the conventional method.

This paper investigates a different method of speaker adaptation for Mandarin syllable recognition. Based on the minimum classification error (MCE) criterion, we use the generalized probabilistic decent (GPD) algorithm to adjust interatively the parameters of the hidden Markov models (HMM). The experiments on the multi-speaker Mandarin syllable database of Telecommunication Laboratories (T.L.) yield the following results: 1) Efficient speaker adaptation can be achieved through discriminative training using the MCE criterion and the GPD algorithm. 2) The computations required can be reduced through the use of the confusion sets in Mandarin base syllables. 3) For the discriminative training, the adjustment on the mean values of the Gaussian mixtures has the most prominent effect on speaker adaptation. 4) The discriminative training approach can be used to enhance the speaker adaptation capability of the maximum a posteriori (MAP) approach.

This paper describes a novel speaker-independent speech recognition method, called speaker-consistent parsing", which is based on an intra-speaker correlation called the speaker-consistency principle. We focus on the fact that a sentence or a string of words is uttered by an individual speaker even in a speaker-independent task. Thus, the proposed method searches through speaker variations in addition to the contents of utterances. As a result of the recognition process, an appropriate standard speaker is selected for speaker adaptation. This new method is experimentally compared with a conventional speaker-independent speech recognition method. Since the speaker-consistency principle best demonstrates its effect with a large number of training and test speakers, a small-scale experiment may not fully exploit this principle. Nevertheless, even the results of our small-scale experiment show that the new method significantly outperforms the conventional method. In addition, this framework's speaker selection mechanism can drastically reduce the likelihood map computation.

One of the most effective methods in speech recognition is the HMM which has been used to model speech statistically. The discrete distribution and the continuos distribution HMMs have been widely used in various applications. However, in recent years, HMMs with various output probability functions have been proposed to further improve recognition performance, e.g. the Gaussian mixture continuous and the semi-continuous distributed HMMs. We recently have also proposed the RBF (radial basis function)-based HMM and the VQ-distortion based HMM which use a RBF function and VQ-distortion measure at each state instead of an output probability density function used by traditional HMMs. In this paper, we describe the RBF-based HMM and the VQ-distortion based HMM and compare their performance with the discrete distributed, the Gaussian mixture distributed and the semi-continuous distributed HMMs based on their speech recognition performance rates through experiments on speaker-independent spoken digit recognition. Our results confirmed that the RBF-based and VQ-distortion based HMMs are more robust and superior to traditional HMMs.

This paper describes new modeling methods combining neural network and hidden Markov model applicable to modeling a time series such as speech signal. The idea assumes that the sequence is nonstationary and is a nonlinear autoregressive process whose parameters are controlled by a hidden Markov chain. One is the model where a non-linear predictor composed of a multi-layered neural network is defined at each state, another is the model where a multi-layered neural network is defined so that the path from the input layer to the output layer is divided into path-groups each of which corresponds to the state of the Markov chain. The latter is an extended model of the former. The parameter estimation methods for these models are shown, and other previously proposed models--one called Neural Prediction Model and another called Linear Predictive HMM--are shown to be special cases of the NPHMM proposed here. The experimental result affirms the justification of these proposed models.

A method of tone recognition has been developed for dissyllabic speech of Standard Chinese based on discrete hidden Markov modeling. As for the feature parameters of recognition, combination of macroscopic and microscopic parameters of fundamental frequency contours was shown to give a better result as compared to the isolated use of each parameter. Speaker normalization was realized by introducing an offset to the fundamental frequency. In order to avoid recognition errors due to syllable segmentation, a scheme of concatenated learning was adopted for training hidden Markov models. Based on the observations of fundamental frequency contours of dissyllables, a scheme was introduced to the method, where a contour was represented with a series of three syllabic tone models, two for the first and the second syllables and one for the transition part around the syllabic boundary. Corresponding to the voiceless consonant of the second syllable, fundamental frequency contour of a dissyllable may include a part without fundamental frequencies. This part was linearly interpolated in the current method. To prove the validity of the proposed method, it was compared with other methods, such as representing all of the dissyllabic contours as the concatenation of two models, assigning a special code to the voiceless part, and so on. Tone sandhi was also taken into account by introducing two additional models for the half-third tone and for the first 4th tone of the combination of two 4th tones. With the proposed method, average recognition rate of 96% was achieved for 5 male and 5 female speakers.

Detection of an unknown word or non-vocabulary word uttered by the user is necessary in realizing a practical spoken language user-interface. This paper describes the evaluation of an unknown word processing method for a subword unit based spoken word recognizer. We have assessed the relationship between the word recognition accuracy of a system and the detection rate of unknown words both by simulation and by experiment of the unknown word processing method. We found that the resultant detection accuracies using the unknown word processing are significantly influenced by the original word recognition accuracy while the degree of such effect depends on the vocabulary size.