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.

We investigate the multipacket message transmissions and variable length message transmissions in slotted ALOHA systems with capture effect. First, we propose an approach that the transmission power level is controlled probabilistically depending on message length for multipacket messages. We consider the multipacket messages model with capture. We derive explicit equations of the effective channel utilization of the model. It is demonstrated that if we increase the numbar of power levels, we can get more effective channel utilization of the system. Secondly, we propose how to assign the slot size and show that the effective utilization of the channel is improved for variable length messages using the approach proposed for multipacket messages. Channel design issue about length of the slot depending on the number of power levels used for transmission is discussed. Thirdly, we propose the multiple messages per slot model with capture. The analytical results show that the multiple messages per slot model can achieve the highest channel utilization among the models discussed in this paper.

This paper presents a new structure for noise and echo cancelers based on a combined fast abaptive algorithm. The main purpose of the new structure is to detect both the double-talk and the unknown path change. This goal is accomplished by using two adaptive filters. A main adaptive filter Fn, adjusted only in the non-double-talk period by the normalized LMS algorithm, is used for providing the canceler output. An auxiliary adaptive filter Ff, adjusted by the fast RLS algorithm, is used for detecting the double-talk and obtaining a near optimum tap-weight vector for Fn in the initialization period and whenever the unknown path has a sudden or fast change. The proposed structure is examined through computer simulation on a noise cancellation problem. Good cancellation performance and stable operation are obtained when signal is a speech corrupted by a white noise, a colored noise and another speech signal. Simulation results also show that the proposed structure is capable of distinguishing the near-end signal from the noise path change and quickly tracking this change.

The global interpolation method we propose evaluates segment pattern continuity and connectedness to produce characters with smooth edges while interpreting blank or missing segments based on global label connectivities, e.g, in extracting a handwritten character overlapping a border, correctly. Conventional character segmentation involving overlapping a border concentrates on removing the thin border based on known format information rather than extracting the character. This generates discontinuous segments which produce distortion due to thinning and errors in direction codes, and is the problem to recognize the extracted character. In our method, characters contacting a border are extracted after the border itself is labeled and removed automatically by devising how to extract wavy and oblique borders involved in fax communication. The absence of character segments is then interpolated based on segment continuity. Interpolated segments are relabeled and checked for matching against the original labeled pattern. If a match cannot be made, segments are reinterpolated until they can be identified. Experimental results show that global interpolation interprets the absence of character segments correctly and generates with smooth edges.

In the correspondence discrete Wigner higher order spectra (WHOS) of harmonizable random signals are addressed and their relations with polyspectra (HOS) are illustrated. It is shown, that discrete WHOS of a random stationary signal do not reduce to the aliased polyspectra in a similar way as Wigner distribution (WD) reduces to the power spectrum of a random signal. Wigner 2nd-order time-frequency distribution of deterministic signals and the 3rd-order spectrum of stationary signals are presented in their modified forms to be used to estimate time-varying third-order spectrum of discrete nonstationary random harmonizable processes.

The applicability of Dynamic Channel Assignment methods to a Reuse Partitioning system in cellular radio systems is investigated in this paper. The investigations indicate that such a system has a tendency to increase the difference between blocking probability for the partitioning two coverage areas in comparison with the conventional Reuse Partitioning system employing Fixed Channel Assignment method. Two schemes using new Channel Rearrangement algorithms are also proposed in order to alleviate the difference as a disadvantage which gives unequal service to the system. The simulation results show that the proposed schemes are able to reduce the difference significantly while increasing the carried traffic by 10% as compared with the conventional system.

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.

Some undetectable stuck-at faults called the redundant faults are included in practical combinational circuits. The redundant fault does not affect the functional behavior of the circuit even if it exists. The redundant fault, however, causes undesirable effects to the circuit such as increase of delay time and decrease of testability of the circuit. It is considered that some redundant faults may cause the logical defects in the future. In this paper, firstly, we study the testability of the redundant fault in the combinational circuit by using delay effects. Secondly, we propose a method for generating a test-pair of a redundant fault by using an extended seven-valued calculus, called TGRF (Test-pair Generation for Redundant Fault). TGRF generates a dynamically sensitizable path for the target line which propagates the change in the value on the target line to a primary output. Finally, we show experimental results on the benchmark circuits under the assumptions of the unit delay and the fanout weighted delay models. It shows that test-pairs for some redundant faults are generated theoretically.

A 3.3 V single power-supply 64 Mb flash memory with a DBL programming scheme has been developed and fabricated with 0.4 µm CMOS technology. 50 ns access time and 256 b erase/programming unit-capacity have been achieved by using hierarchical word- and bit-line structures and DBL programming scheme. Furthermore in order to lower operating voltage the HiCR cell is used. The chip size is 19.3 mm13.3 mm.

This paper proposes two algorithms for defining a routing domain in multiclass-of-service networks. One an off-line-based method, whose objective is to optimize dynamic routing performance by using precise knowledge on the traffic levels. The algorithm of the proposed method takes into account the random nature of the traffic flow, which is not considered in the network flow approach. The proposed method inherits the conceptual simplicity of the network flow approach and remains applicable to large and complex networks. In simulation experiments, the proposed off-line-based method performs better than the method based on the network flow approach, but has a similar the computation time requirement. The other method proposed here is an on-line-based method for application to B-ISDNs, where precise traffic data is not expected to be available. In this method, the routing domain is defined adaptively according to the network performance (call-blocking probability) measured in real-time. In simulation experiments, the performance of this method is comparable to that of the off-line-based method--especially when highly efficient dynamic routing is used. This paper also derives and describes methods for approximating the implied costs for multiclass-of-service networks. The approximations are very useful not only for off-line-based routing domain definition (RDD) methods but also for other kinds of network controls or optimal network dimensioning based on the concept of revenue optimization.

We carried out a one year field trial of a voice-activated automatic telephone exchange service at KDD Laboratories which has about 200 branch phones. This system has DSP-based continuous speech recognition hardware which can process incoming calls in real time using a vocabulary of 300 words. The recognition accuracy was found to be 92.5% for speech read from a written text under laboratory conditions independent of the speaker. In this paper, we describe the performance of the system obtained as a result of the field trial. Apart from recognition accuracy, there was about 20% error due to out-of-vocabulary input and incorrect detection of speech endpoints which had not been allowed for in the laboratory experiments. Also, we found that the recognition accuracy for actual speech was about 18% lower than for speech read from text even if there were no out-of-vocabulary words. In this paper, we examine error variations for individual data in order to try and pinpoint the cause of incorrect recognition. It was found from experiments on the collected data that the pause model used, filled pause grammar and differences of channel frequency response seriously affected recognition accuracy. With the help of simple techniques to overcome these problems, we finally obtained a recognition accuracy of 88.7% for real data.

We propose an objective measure from assessing low-rate coded speech. The model for this objective measure, in which several known features of the perceptual processing of speech sounds by the human ear are emulated, is based on the Hertz-to-Bark transformation, critical-band filtering with preemphasis to boost higher frequencies, nonlinear conversion for subjective loudness, and temporal (forward) masking. The effectiveness of the measure, called the Bark spectral distortion rating (BSDR), was validated by second-order polynomial regression analysis between the computed BSDR values and subjective MOS ratings obtained for a large number of utterances coded by several versions of CELP coders and one VSELP coder under three degradation conditions: input speech levels, transmission error rates, and background noise levels. The BSDR values correspond better to MOS ratings than several commonly used measures. Thus, BSDR can be used to accurately predict subjective scores.

With a view toward the improvement of life-saving rate, the advancement of emergency radio system was attempted. The telemetry technique was introduced to the mobile communication from a running ambulance. A system was newly developed which enables us to transmit the information of an emergency patient from an ambulance to an emergency room of a hospital. This system can transmit an audio signal, physiological signals such as an ECG and a blood oxygen level, as well as a color image. In the experiment, the feasibility of this technique was verified. In the test of its practical usefulness, the following points were evaluated using a mobile telephone line and an emergency radio link. With the regular condition of the communication link, the stability of signal transmission was reasonably well. The fidelity of the transmitted signal was satisfactory for the use of an emergency medicine.

This paper presents a performance analysis of the standard non-coherent delay-locked loop in asynchronous direct-sequence code division multiple access (DS-CDMA) environments. In particular, the effects of multiple access interference on the loop performance are addressed. We work out an expression for the steady-state tracking-error variance and provide performance curves in terms of mean time to lose lock as a function of the number of interfering users and Eb/No.

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.

This paper describes a multimodal dialogue system employing speech input. This system uses three input methods (through a speech recognizer, a mouse, and a keyboard) and two output methods (through a display and using sound). For the speech recognizer, an algorithm is employed for large-vocabulary speaker-independent continuous speech recognition based on the HMM-LR technique. This system is implemented for telephone directory assistance to evaluate the speech recognition algorithm and to investigate the variations in speech structure that users utter to computers. Speech input is used in a multimodal environment. The collecting of dialogue data between computers and users is also carried out. Twenty telephone-number retrieval tasks are used to evaluate this system. In the experiments, all the users are equally trained in using the dialogue system with an interactive guidance system implemented on a workstation. Simplified city maps that indicate subscriber names and addresses are used to reduce the implicit restrictions imposed by written sentences, thus allowing each user to develop his own forms of expression. The task completion rate is 99.0% and approximately 75% of the users say that they prefer this system to using a telephone book. Moreover, there is a significant decrease in nonkeyword usage, i.e., the usage of words other than names and addresses, for users who receive more utterance practice.

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.

A new clustering method was proposed to increase the effect of duration modeling on the HMM-based phoneme recognition. A precise observation on the temporal correspondences between a phoneme HMM with output probabilities by single Gaussian modeling and its training data indicated that there were two extreme cases, one with several types of correspondences in a phoneme class completely different from each other, and the other with only one type of correspondence. Although duration modeling was commonly used to incorporate the temporal information in the HMMs, a good modeling could not be obtained for the former case. Further observation for phoneme HMMs with output probabilities by Gaussian mixture modeling also showed that some HMMs still had multiple temporal correspondences, though the number of such phonemes was reduced as compared to the case of single Gaussian modeling. An appropriate duration modeling cannot be obtained for these phoneme HMMs by the conventional methods, where the duration distribution for each HMM state is represented by a distribution function. In order to cope with the problem, a new method was proposed which was based on the clustering of phoneme classes with plural types of temporal correspondences into sub-classes. The clustering was conducted so as to reduce the variations of the temporal correspondences in sub-classes. After the clustering, an HMM was constructed for each sub-class. Using the proposed method, speaker dependent recognition experiments were performed for phonemes segmented from isolated words. A few-percent increase was realized in the recognition rate, which was not obtained by another method based on the duration modeling with a Gaussian mixture.

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.

Three-dimensional vocal tract shapes of a male, a female and a child subjects are measured from magnetic resonance (MR) images during sustained phonation of Japanese vowels /a, i, u, e, o/. Non-uniform dimensional differences in the vocal tract shapes of the subjects are quantitatively measured. Vocal tract area functions of the female and child subjects are normalized to those of the male on the basis of non-uniform and uniform scalings of the vocal tract length and compared with each other. A comparison is also made between the formant frequencies computed from the area functions normalized by the two different scalings. It is suggested by the comparisons that non-uniformity in the vocal tract dimensions is not essential in the normalization of the five Japanese vowels.