This paper presents a parallel sorting algorithm which sorts n elements on O(n/w+n log n/p) time using p(n) processors arranged in a 1-dimensional grid with w(n1-ε) buses for every fixed ε>0. Furthermore, it is shown that np elements can be sorted in O(n/w+n log n/p) time on pp (pn) processors arranged in a 2-dimensional grid with w(n1-ε) buses in each column and in each row. These algorithms are optimal because their time complexities are equal to the lower bounds.

This paper describes automatic generation of speech synthesis rules which predict a stress level for each bunsetsu in long noun phrases. The rules are inductively inferred from a lot of speech data by using two kinds of tree-based methods, the conventional decision tree and the SBR-tree methods. The rule sets automatically generated by two methods have almost the same performance and decrease the prediction error to about 14 Hz from 23 Hz of the accent component value. The rate of the correct reproduction of the change for adjacent bunsetsu pairs is also used as a measure for evaluating the generated rule sets and they correctly reproduce the change of about 80%. The effectiveness of the rule sets is verified through the listening test. And, with regard to the comprehensiveness of the generated rules, the rules by the SBR-tree methods are very compact and easy to human experts to interpret and matches the former studies.

Multiplication-accumulation is the basic computation required for image filtering operations. For real-time image filtering, very high throughput computation is essential. This work proposes a hardware algorithm for an application-specific VLSI architecture which realizes an area-efficient high throughput multiplier-accumulator. The proposed algorithm utilizes a priori knowledge of filter mask coefficients and optimizes number of basic hardware components (e.g., full adders, pipeline latches, etc.). This results in the minimum area VLSI architecture under certain input/output constraints.

This paper proposes a reconfigurable parallel processor based on a two-dimensional linear celular automaton model. The processor based on the model can be reconfigured quickly by utilizing the characteristics of the automaton used for its model. Moreover, the processor has short data path length between processing elements compared with the length of the processor based on one-dimensional linear cellular automaton model which has been already discussed. The processing elements of the processor based on the two-dimensional linear cellular automaton model are regarded as cells and the operational states of the processor are treated as the states of the automaton. When faults are detected, the processor can be reconfigured by changing its state under the state transition function of the processor determined by the weighting function of the automaton model. The processor can be reconfigured within a clock period required for making a state transition. This processor is extremely effective for real-time data processing systems required high reliability.

An approximate but sufficiently accurate high frequency solution is developed in this paper for analyzing the problem of electromagnetic plane wave scattering by an open-ended, perfectly-conducting, semi-infinite parallel-plate waveguide with a thin layer of lossy or absorbing material on its inner wall, and with a planar termination inside. The high frequency solution combines uniform geometrical theory of diffraction (UTD) and aperture integration (AI) methods. The present method has several advantages in comparison with other methods.

This paper focuses on two areas in our effort to synthesize speech from neuromotor input using neural network models that effect transforms between cognitive intentions to speak, their physiological effects on vocal tract structures, and subsequent realization as acoustic signals. The first area concerns the biomechanical transform between motor commands to muscles and the ensuing articulator behavior. Using physiological data of muscle EMG (electromyography) and articulator movements during natural English speech utterances, three articulator-specific neural networks learn the forward dynamics that relate motor commands to the muscles and motion of the tongue, jaw, ant lips. Compared to a fully-connected network, mapping muscle EMG and motion for all three sets of articulators at once, this modular approach has improved performance by reducing network complexity and has eliminated some of the confounding influence of functional coupling among articulators. Network independence has also allowed us to identify and assess the effects of technical and empirical limitations on an articulator-by-articulator basis. This is particularly important for modeling the tongue whose complex structure is very difficult to examine empirically. The second area of progress concerns the transform between articulator motion and the speech acoustics. From the articulatory movement trajectories, a second neural network generates PARCOR (partial correlation) coefficients which are then used to synthesize the speech acoustics. In the current implementation, articulator velocities have been added as the inputs to the network. As a result, the model now follows the fast changes of the coefficients for consonants generated by relatively slow articulatory movements during natural English utterances. Although much work still needs to be done, progress in these areas brings us closer to our goal of emulating speech production processes computationally.

For analyzing the transient electromagnetic fields caused by electrostatic discharge (ESD), a new ESD model is presented here. Numerical calculation is also given to explain the distinctive phenomenon being well-recognized in the ESD event.

This paper proposes a new scheme for concatenative speech synthesis to improve the speech segment selection procedure. The proposed scheme selects a segment sequence for concatenation by minimizing acoustic distortions between the selected segment and the desired spectrum for the target without the use of heuristics. Four types of distortion, a) the spectral prototypicality of a segment, b) the spectral difference between the source and target contexts, c) the degradation resulting from concatenation of phonemes, and d) the acoustic discontinuity between the concatenated segments, are formulated as acoustic quantities, and used as measures for minimization. A search method for selecting segments from a large speech database is also descrided. In this method, a three-step optimization using dynamic programming is used to minimize the four types of distortion. A perceptual test shows that this proposed segment selection method with minimum distortion criteria produces high quality synthesized speech, and that contextual spectral difference and acoustic discontinuity at the segment boundary are important measures for improving the quality.

A method for constructing a compact non-blocking, large matrix-size, optomechanical switch is proposed. It can be switched arbitrarily by disconnecting and reconnecting ferrules on the matrix board. A 250500, 25-mm-ferrule-pitch, 800W855D945H (mm) switch is fabricated. Although the space above the board is densely packed with many ferrule-terminated-fibers, because of the way in which they are arranged and the control of their length, there is no discernible excess loss due to fiber bending.

A text-to-speech conversion system for Japanese has been developed for the purpose of producing high-quality speech output. This system consists of four processing stages: 1) linguistic processing, 2) phonological processing, 3) control parameter generation, and 4) speech waveform generation. Although the processing at the first stage is restricted to the texts on general weather conditions, the other three stages can also cope with texts of news and narrations on other topics. Since the prosodic features of speech are largely related to the linguistic information, such as word accent, syntactic structure and discourse structure, linguistic processing of a wider range than ever, at least a sentence, is indispensable to obtain good quality speech with respect to the prosody. From this point of view, input text was restricted to the weather forecast sentences and a method for linguistic processing was developed to conduct morpheme, syntactic and semantic analyses simultaneously. A quantitative model for generating fundamental frequency contours was adopted to make a good reflection of the linguistic information on the prosody of synthetic speech. A set of prosodic rules was constructed to generate prosodic symbols representing prosodic structures of the text from the linguistic information obtained at the first stage. A new speech synthesizer based on the terminal analog method was also developed to improve the segmental quality of synthetic speech. It consists of four paths of cascade connection of pole/zero filters and three waveform generators. The four paths are respectively used for the synthesis of vowels and vowel-like sounds, nasal murmur and buzz bar, friction, and plosion, while the three generators produce voicing source waveform approximated by polynomials, white Gaussian noise source for fricatives and impulse source for plosives. The validity of the approach above has been confirmed by the listening tests using speech synthesized by the developed system. Improvements both in the quality of prosodic features and in the quality of segmental features were realized for the synthetic speech.

We deal with a new type ceramic emitter which is used in a cleanroom ionizer system and is composed of a needle-shaped silicon and a rod-shaped silicon carbide ceramics. The discharge test was carried out to investigate the particle generation from the emitter and the degradation of the emitter. As a result, it was found that the ceramic emitter had practically higher performance than a conventional tungsten emitter.

Numerical analysis of the electromagnetic radiation from conducting surface structures is concerned. The method of moments is discussed with the surface-patch modeling in which the surface quantities, i.e. the current, charge and impedance are directly introduced and with the wire-grid modeling in which the surface quantities are approximated by the filamentary traces. The crucial element to a numerical advantage of the wire-grid modeling lies in the simplicity of its mathematical involvements that should be traded for the uncertainties in the construction of the model. The surface-patch techniques are generally not only clear and straightforward but also more reliable than the wire-grid modeling for the computation of the surface quantities. In this work, we bring about a comparative discussion of the two approaches while the analysis of a built-in planar antenna is reported. For the purpose of the comparison, the same electric field integral equation and the Galerkin's procedure with the linear expansion/testing functions are used for both the wire-grid and surface-patch modeling.

A stored channel simulator for digital mobile radio enviroments is proposed, which enables the field tests in the laboratory under identical conditions, since it can reproduce the actual multipath radio channels by using the channel impulse responses (CIR's) measured in the field. Linear interpolation of CIR is introduced to simplify the structure of the proposed simulator. The performance of the proposed simulator is confirmed by the laboratory tests.

For the production of high quality synthetic sounds in a text-to-speech system, an excellent synthesizing method of speech signals is indispensable. In this paper, a new speech analysis-synthesis method for the text-to-speech system is proposed. The signals of voiced speech, which have a line spectrum structure at intervals of pitch in the linear frequency domain, can be represented approximately by the superposition of sinusoidal waves. In our system, analysis and synthesis are performed using such a harmonic structure of the signals of voiced speech. In the analysis phase, assuming an exact harmonic structure model at intervals of pitch against the fine structure of the short-time power spectrum, the fundamental frequency f0 is decided so as to minimize the error of the log-power spectrum at each peak position. At the same time, according to the value of the above minimized error, the rate of periodicity of the speech signal is detemined. Then the log-power spectrum envelope is represented by the cosine-series interpolating the data which are sampled at every pitch period. In the synthesis phase, numerical solutions of non-linear differential equations which generate sinusoidal waves are used. For voiced sounds, those equations behave as a group of mutually synchronized oscillators. These sinusoidal waves are superposed so as to reconstruct the line spectrum structure. For voiceless sounds, those non-linear differential equations work as passive filters with input noise sources. Our system has some characteristics as follows. (1) Voiced and voiceless sounds can be treated in a same framowork. (2) Since the phase and the power information of each sinusoidal wave can be easily controlled, if necessary, periodic waveforms in the voiced sounds can be precisely reproduced in the time domain. (3) The fundamental frequency f0 and phoneme duration can be easily changed without much degradation of original sound quality.

In this paper, we quantitively analyzed speech data in seven different styles to make natural Japanese conversational speech synthesis. Three reading styles were produced at different speeds (slow, normal and fast), and four speaking styles were produced by enacting conversation in different situations (free, hurried, angry and polite). To clarify the differences in prosodic characteristics between conversational speech and read speech, means and standard deviations of vowel duration, vowel amplitude and fundamental frequency (F0) were analyzed. We found large variation in these prosodic parameters. To look more precisely at the segmental duration and segmental amplitude differences between conversational speech and read speech, control rules of prosodic parameters in reading styles were applied to conversational speech. F0 contours of different speaking styles are superposed by normalizing the segmental duration. The differences between estimated values and actual values were analyzed. Large differences were found at sentence final and key (focused) phrases. Sentence final positions showed lengthening of segmental vowel duration and increased segmental vowel amplitude. Key phrase positions featured raising F0.

The authors developed a portable Japanese text-to-speech system using a pocket-sized formant speech synthesizer. It consists of a linguistic processor and an acoustic processor. The linguistic processor runs on an MS-DOS personal computer and has functions to determine readings and prosodic information for input sentences written in kana-kanji-mixed style. New techniques, such as minimization of a cost function for phrases, rare-compound flag, semantic information, information of reading selection and restriction by associated particles, are used to increase the accuracy of readings and accent positions. The accuracy of determining readings and accent positions is 98.6% for sentences in newspaper articles. It is possible to use the linguistic processor through an interface library which has also been developed by the authors. Consequently, it has become possible not only to convert whole texts stored in text files but also to convert parts of sentences sent by the interface library sequentially, and the readings and prosodic information are optimized for the whole sentence at one time. The acoustic processor is custom-made hardware, and it has adopted new techniques, for the improvement of rules for vowel devoicing, control of phoneme durations, control of the phrase components of voice fundamental frequency and the construction of the acoustic parameter database. Due to the above-mentioned modifications, the naturalness of synthetic speech generated by a Klatt-type formant speech synthesizer was improved. On a naturalness test it was rated 3.61 on a scale of 5 points from 0 to 4.

Prosodic features of the spoken Japanese play an important role in the transmission of linguistic information concerning the lexical word accent, the sentence structure and the discourse structure. In order to construct prosodic rules for synthesizing high-quality speech, therefore, prosodic features of speech should be quantitatively analyzed with respect to the linguistic information. With a special focus on the fundamental frequency contour, we first define four prosodic units for the spoken Japanese, viz., prosodic word, prosodic phrase, prosodic clause and prosodic sentence, based on a decomposition of the fundamental frequency contour using a functional model for the generation process. Syntactic units are also introduced which have rough correspondence to these prosodic units. The relationships between the linguistic information and the characteristics of the components of the fundamental frequency contour are then described on the basis of results obtained by the analysis of two sets of speech material. Analysis of weathercast and newscast sentences showed that prosodic boundaries given by the manner of continuation/termination of phrase components fall into three categories, and are primarily related to the syntactic boundaries. On the other hand, analysis of noun phrases with various combinations of word accent types, syntactic structures, and focal conditions, indicated that the magnitude and the shape of the accent components, which of course reflect the information concerning the lexical accent types of constituent words, are largely influenced by the focal structure. The results also indicated that there are cases where prosody fails to meet all the requirements presented by word accent, syntax and discourse.

The finite-difference beam-propagation method is applied to the analysis of hollow slab waveguides (HSWs). The attenuation constants for the TE0 and TE1 modes are evaluated and compared with those obtained by the perturbation theory. The propagating field and differential power loss in the transition from a straight HSW to a bent HSW are revealed and discussed.

We propose a smart design methodology for advanced ULSI memories to reduce the turn around time(TAT) for circuit revisions with no area penalty. This methodology was executed by distributing extra gate-arrays, which were composed of the n-channel and p-channel transistors, under the power line and the signal line. This method was applied to the development of a 16 Mb DRAM with double metal wiring. The design TAT can be reduced to 1/8 using 1500 gates. This design methodology has been confirmed to be very effective.

An ECL-CMOS SRAM technology is proposed which features a combination of ECL word drivers, ECL write circuits and low-voltage CMOS cells. This technology assures both ultra-high-speed and high-density. In the ECL-CMOS SRAM,various kinds of noise generated during the write cycle seriously affect the memory performance, because it has much faster access than conventional SRAMs. To overcome this problem, we propose three noise reduction techniques; a noise reduction clamp circuit, an emitter follower with damping capacitor and a twisted bit line structure with "normally on" equalizer. These techniques allow fast accese and cycle times. To evaluate these techniques, a 64-kb SRAM chip was fabricated using 0.5-µm BiCMOS technology. This SRAM has a short cycle time of 2 ns and a very fast access time of 1.5 ns. Evaluation proves the usefulness of these techniques.