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.

A new system for speech synthesis by concatenating waveforms selected from a dictionary is described. The dictionary is constructed from a two-hour speech that includes isolated words and sentences uttered by one male speaker, and contains over 45,000 entries which are identified by their average pitch, dynamic pitch parameter which represents micro pitch structure in a segment, duration and average amplitude. Phoneme duration is set according to phoneme environment, and phoneme power is controlled, by both pitch frequency and phoneme environment. Tests show the average errors in vowel duration and consonant duration are 28.8 ms and 16.8 ms respectively, and the vowel power average error is 2.9 dB. The pitch frequency patterns are calculated according to a conventional model in which the accent component is abbed to a gross phrase component. Set a phoneme string and prosody information, the optimum waveforms are selected from the dictionary by matching their attributes with the given phonetic and prosodic information. A waveform selection function, which has two terms corresponding to prosody and phonological coincidence between rule-set values and waveform values from the dictionary, is proposed. The weight coefficients used in the selection function are determined through subjective hearing tests. The selected waveform segments are then modified in waveform domain to further adjust for the desired prosody. A pitch frequency modification method based on pitch synchronous overlap-add technique is introduced into the system. Lastly, the waveforms are interpolated between voiced waveforms to avoid abrupt changes in voice spectrum and waveform shape. An absolute evaluation test of five grades is performed to the synthesized voice and the mean of the score is 3.1, which is over "good," and while the original speaker quality is retained.

Text-to-speech synthesis (TTS) is currently one of the most important media conversion techniques. In this paper, we describe a Japanese TTS card developed for constructing a personal-computer-based multimedia platform, and a TTS software package developed for a workstation-based multimedia platform. Some applications of this hardware and software are also discussed. The TTS consists of a linguistic processing stage for converting text into phonetic and prosodic information, and a speech processing stage for producing speech from the phonetic and prosodic symbols. The linguistic processing stage uses morphological analysis, rewriting rules for accent movement and pause insertion, and other techniques to impart correct accentuation and a natural-sounding intonation to the synthesized speech. The speech processing stage employs the cepstrum method with consonant-vowel (CV) syllables as the synthesis unit to achieve clear and smooth synthesized speech. All of the processing for converting Japanese text (consisting of mixed Japanese Kanji and Kana characters) to synthesized speech is done internally on the TTS card. This allows the card to be used widely in various applications, including electronic mail and telephone service systems without placing any processing burden on the personal computer. The TTS software was used for an E-mail reading tool on a workstation.

A module for rule-based Japanese speech synthesis has been developed. The synthesizer was constructed using the Multiple-Cascade Terminal Analog (MCTA) structure, and this sturcture has been improved in three respects: the voicing-source model has an increased number of variable parameters which allows for voicing-source waveforms that better approximate natural speech; the spectral characteristics of the fricative source have been improved; and the path used for nasal consonants has an increased number of resonators to better conform to theory. The current synthesis system uses a modified stored-pattern data structure which allows better transitions between syllables; however, time-invariant values are used in certain cases in order to decrease the amount of required memory. This system also has a new consolidated method for generating geminate obstruents and syllabic nasals. This synthesizer and synthesis system have been implemented in a re-developed rule-based speech-synthesis module. This module has been constructed using ASIC technology and has both small size (56368 mm) and light weight (19g); it is therefore possible to embed it in various types of portable or moving machinery. The module can be connected directly to a mocroprocessor bus and accepts as input sentences which are generated by the host computer. The input sentences are written with the Japanese katakana or romaji syllabaries and other symbols which describe the sentence structure. The syllable articulation rate for one hundred Japanese syllables (including palatalized sounds) is 65% and for sixty-seven syllables (not including palatalized sounds) is 74%. The word intelligibility, measured using phonetically-balanced words, it 88%.

This article discusses on PDM (Petri net based Development Methodology) which integrates approaches, modeling methods, design methods and analysis methods in a coherent manner. Although various development techniques based on Petri nets have demonstrated advantages over conventional techniques, those techniques are rather ad hoc and lack an overall picture on entire development process. PDM anticipates to provide a refernce process model to develop distributed systems with various Petri net based development methods. Behavioral properties of distrbuted systems can be an appropriate application domain of PDM.

This paper presents a new method to derive the phase difference between n-tuples of an m-sequence over GF(p) of period pn-1. For the binary m-sequence of the characteristic polynomial f(x)=xn+xd+1 with d=1,2c or n-2c, the explicit formulas of the phase difference from the initial n-tuple are efficiently derived by our method for specific n-tuples such as that consisting of all 1's and that cosisting of one 1 and n-1 0's, although the previously known formula exists only for that consisting of all 1's.

String grammars (languages) have been extensively studied from 60's. On the other hand, the transformational grammar, proposed by Chomsky, contains the transformation from the set of derivation trees of context-free language to the surface set. And the grammar regarded a tree as an input sentence to some transducer. After that from latter half of 60's, the studies of acceptor, transducer, and so on, whose input is a tree, have been done extensively. In this paper we propose, as a model, a new type of transducer which translates trees into trees and investigate its fundamental properties. The model proposed here is the pushdown tree transducer (for shortly PDTT) that is an extension of the finite state tree transducer discussed by J. W. Thacher, W. C. Rounds, J. Engelfriet, and so on. The main subjects discussed here (we consider only top-down case (t-PDTT)), are as follows: (1) final state t-PDTT translation is equivalent to empty stack t-PDTT translation and vice versa, (2) for any t-PDTT, a single state t-PDTT which is equivalent to it always exists, (3) as a standard form the symmetric stack form t-PDTT is proposed and based on this, it is shown that any single state t-PDTT can be always converted into a linear stack t-PDTT, and so on.

High performances of CMOS/SOI inverter by simulations of analytical model, reducing the poly-Si gate thickness (tm), and experiments are verified and proposed. It is shown that the tm and gate oxide thickness(tox) are correlated to gate fringing capacitance, which largely influences on the Propagation Delay Time(TPD). Contributions of gate fringing capacitance to CMOS/SIMOX inverter time delay in deep submicrometer gate devices are propounded. Measurements of the fifty-one stage ring oscillator's TPDs are completed for comparison with analytical model. Simulation results by the analytical model, including Time-Dependent Gate Capacitance (TDGC) model, agree well with the experimental results at the same conditions. Simulation results are also predicted that SOI technology is promising for speed enhancement by reducing the poly-Si gate thickness, while the tox remains constant. It is concluded that the TPDs by reducing the tm to zero are improved up to about two times faster than typically fabricated ring oscillator at 350 nm of the tm in deep-submicrometer gate CMOS/SIMOX inverters at room temperature.

Let U denote a set comprising elements called "keys." The goal of the nearest point problem is to search quickly for a key among some keys x1 , xn that is the nearest to a reference key x under a partial order relation defined as (x, y) (x, z) for x, y, zU if d(x, y)d(x, z) given a wide-sense distance measure d. This article proposes a method of rearranging x1 , xn into a binary perfectly-balanced tree for solving quickly the nearest point problems. Further, computational performances of the proposed method are evaluated experimentally.

A coherent communication system using squeezed light is one of candidates for a realization of super-reliable systems. In order to design such a system, it is essential to understand and to analyze modulators mathematically. However, quantum noise of squeezed light has a colored spectrum which changes with respect to phase of a local laser. Therefore the optimization of the relationship between signal and quantum noise spectrums is required at a modulator to obtain the ultimate performance of the communication system. In this paper, some ideas of modulators for squeezed light are proposed and their spectrum transformations are given. After the brief summary of squeezed quantum noise, a new concept which originates from the restriction of the local laser phase is applied to it. This concept makes a problem originated from a colored quantum noise spectrum more serious. It results in the optimization problem for the relationship between the quantum noise spectrum and signal power spectrum. The solution of this problem is also given under the restriction of local laser phase. As a result, a general design theory for coherent communication system using the squeezed light is given.

This paper considers the problem of finding a largest common subgraph of graphs, which is an important problem in chemical synthesis. It is known that the problem is NP-hard even if graphs are restricted to planar graphs of vertex degree at most three. By the way, a graph is called an almost tree if E(B)V(B)+ K holds for every block B where K is a constant. In this paper, a polynomial time algorithm for finding a largest common subgraph of two graphs which are connected, almost trees and of bounded vertex degree. The algorithm is an extension of a subtree isomorphism algorithm which is based on dynamic programming. Moreover, it is shown that the degree bound is essential. That is, the problem of finding a largest common subgraph of two connected almost trees is proved to be NP-hard for any K0 if degree is not bounded. The three dimensional matching problem, a well known NP-complete problem, is reduced to the problem.

A linear block code has a finite-length trellis diagram which terminates at the length of the code. Such a trellis diagram is expressed and constructed in terms of sections. The complexity of an L-section trellis diagram for a linear block code is measured by the state and branch complexities, the state connectivity, and the parallel structure. The state complexity is defined as the number of states at the end of each section of the L-section trellis diagram, the branch complexity is defined as the number of parallel branches between two adjacent states, the state connectivity is defined in terms of the number of states which are adjacent to and from a given state and the connections between disjoint subsets of states, and the parallel structure is expressed in terms of the number of parallel sub-trellis diagrams without cross connections between them. This paper investigates the branch complexity, the state connectivity, and the parallel structure of an L-section minimal trellis diagram for a binary linear block code. First these complexities and structure are expressed in terms of the dimensions of specific subcodes of the given code. Then, the complexities of 2i-section minimal trellis diagrams for Reed-Muller codes are given.

In order to explain the temperature and frequency characteristics of high-Tc superconductors, a new model is proposed, which will be called the improved three-fluid model, where the momentum relaxation time τ is assumed to depend on temperature in the superconducting and normal states, respectively, although τ has been assumed to be independent of temperature for the conventional three-fluid model. According to this model, the complex conductivity σ1jσ2 and the surface impedance ZsRsjXs, where Rs is the surface resistance and Xs is the surface reactance, are expressed as a function of temperature. The temperature dependences of Zs and for a YBCO bulk estimated using this model agree very well with ones measured by the dielectric-loaded cavity method in room to cryogenic temperature. In particular, a peak of σ1 observed just below the critical temperature Tc in experiments, appeared in the calculated results based on this model. This phenomenon has been already known in the BCS theory. Thus, it is verified that this model is useful to explain the microwave characteristics of high-Tc superconductors in room to cryogenic temperature. On the other hand, the residual normal electron density nres4.2541023 m-3 and the total electron density nt7.3081024 m-3 are obtained by calculation. The ratio nres/nt0.058 can be used as figure of merit to evaluate material quality of high-Tc superconductors; thus it means that there is 5.8% nonpairing electron in this YBCO bulk.

This paper proposes a technique for estimating speech parameters in noisy environment. The technique uses a spectral model represented by generalized cepstrum and estimates the generalized cepstral coefficients from the speech which has been degraded by additive background noise. Parameter estimation is based on maximum a posteriori (MAP) estimation procedure. An iterative approach which has been formulated for all-pole modeling is applied to the generalized cepstral modeling. Generalized cepstral coefficients are obtained by an iterative procedure that consists of the unbiased estimation of log spectrum and noncausal Wiener filtering. Since the generalized cepstral model includes the all-pole model as a special case, the technique can be viewed as a generalization of the all-pole modeling based on MAP estimation. The proposed technique is applied to the enhancement of speech and several experimental results are also shown.

The barrier-layer was successfully fabricated for a preparation of tunneling junction using high Tc oxidesuperconductor such as Bi-Sr-Ca-Cu-O system. Bi2Sr2Ca2Cu3Ox films were used for both superconducting electrodes and the barrier was mainly Bi2Sr2CaCu2O and the rest that was formed by effects of de-calcium from the first sputtered (2223) film. The reaction of de-calcium occurred immersing it in carbonated water. The change of (2223) phase of BSCCO was confirmed with a comparison of the intensity of X-ray diffraction. The superconductive transition temperature of the junction is different from that of the single film (2223) which had no treatment with carbonated water. Zero-bias-currents through fabricated barrier are observed and the critical currents depend on temperature so far as measured temperature region of 79 K-72 K.

Experimental results of an N-S junction and analysis of the results using the Arnold theory were reported. Au and Pb were employed as a normal metal and a superconducting material, respectively. The excess current effect due to the Andreev reflection was observed in the current-voltage characteristics of an N-S junction whose normal resistance was 1.603 Ω. The excess current at 4.62 K was about 0.7 mA when the applied voltage was 2 mV. The barrier height and width were estimated to be 1.0169 eV and 0.7 , respectively, by comparing the experimental results and analysis based on the Arnold theory. In the voltage region less than 2 mV, the theory well agreed with the experiment. Moreover, the applied voltage dependence of the supercurrent and quasiparticle current were separately calculated. It was made clear that the supercurrent was larger than the quasiparticle current in the voltage region less than 2Δ/e, where Δ is the superconducting energy gap and e is the absolute value of an electron's charge. The supercurrent began to gradually saturate when the voltage was higher than Δ/e and became constant at the applied voltage greater than 2Δ/e. In our experiment, the excess current larger than expected from the Arnold theory was observed in the voltage region higher than 2Δ/e.

We propose a continuous speech recognition algorithm utilizing island-driven A* search. Conventional left-to-right A* search is probable to lose the optimal solution from a finite stack if some obscurities appear at the start of an input speech. Proposed island-driven A* search proceeds searching forward and backward from the clearest part of an input speech, and thus can avoid to lose the optimal solution from a finite stack.

A speech coding scheme at 3.6 kbit/s has been proposed. The scheme is based on CELP (Code Excited Linear Prediction) with pitch synchronous innovation, which means even random codevectors as well as adaptive codevectors have pitch periodicity. The quality is comparable to 6.7 kbit/s VSELP coder for the Japanese cellular radio standard.

This paper reviews the historical aspect of contributions on the theory of analysis and diagnosis of linear circuits, which have been made by Japanese researchers in these twenty years. On papers of diagnosis, those related to element-value solvability (or determinability) are mainly reviewed. Some important problems are suggested.

This letter proposes an LSP quantizing method which uses interframe correlation of the parameters. The quantized parameters are represented as a moving average of code vectors. Using this method, LSP parameters are quantized efficiently and the degradation of decoded parameters caused by bit errors affects only a few following frames.