This paper proposes a new method of phoneme power control for speech synthesis by rule. The innovation of this method lies in its use of the phoneme environment and the relationship between speech power and pitch frequency. First, the permissible threshold (PT) for power modification is measured by subjective experiments using power manipulated speech material. As a result, it is concluded that the PT of power modification is 4.1 dB. This experimental result is significant when discussing power control and gives a criterion for power control accuracy. Next, the relationship between speech power and pitch frequency is analyzed using a very large speech data base. The results show that the relationship between phoneme power and pitch frequency is affected by the kind of phoneme, the adjoining phonemes, rising or falling pitch, and initial or final position in the sentence. Finally, we propose that the phoneme power should be controlled by pitch frequency and phoneme environment. This proposal is implemented in a waveform concatenation type text-to-speech synthesizer. This new method yields an averaged root mean square error between real and estimated speech power of 2.17 dB. This value indicates that 94% of the estimated power values are within the permissible threshold of human perception.

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.

In the future, it will be necessary that robot technology or environmental technology has an auditory function of recognizing sound expect for speech. In this letter, we propose a recognition system for the species of birds receiving birdcalls, based on network technology. We show the first step of a recognition system for the species of birds, as an application of a recognition system for environmental sound.

Reactive systems respond to internal or external stimuli and act in an event-driven manner. It is generally difficult to specify a complex reactive systems' behavior using conventional state machine formalism. One reason is that actual reactive systems are usually formed by combining plural state-machince that behave concurretly. This paper presents the State Diagram Matrix (SDM) which is a visual and hierarchical formalism of such a reactive system's behavior. SDM has two concepts. The first is matrix plane description on which 3-dimensional state space is projected. The second is state abstraction for hierarchical state-machine definition. Understandability and reliability of control software was improved as a consequence of adopting SDM for specifying disk-subsystem control requirements. The development support functions of SDM using a workstation are also described.

The paper presents the improvement of out new approach to optimize the process parameter variation, device heat and wire parasitics for analog LSI design by explicitly incorporating various performance estimations into objective functions for placement and routing. To minimize these objective functions, the placement by the simulated annealing method, and maze routing are effectively modified with the perfomance estimation. The improvement results in the excellent performance driven layout for the large size of analog LSIs.

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.

The fast and highly reliable method reported here uses two techniques to detect all types of defects, such as unsoldered leads, solder bridges, and misalignes leads in the minute solder joints of high density mounted devices. One technique uses external force applied by an air jet that vibrates or shifts unsoldered leads. The vibration and shift is detected as a change in the speckle pattern produced by laser illumination of the solder joints. The other technique uses fluorescence generated by short-wavelength laser illumination. The fluorescence from a printed circuit board produces a silhouette of the solder joint and this image is processed to detect defects. Experimental results show that this inspection method detects all kinds of defects accurately and with a very low false alarm rate.

The characteristics of an open-boundary Cherenkov laser for an electromagnetic wave with a continuous frequency spectrum are numerically analyzed. A given power spectral density for the input wave is found to get concentrated around the frequency where the spatial growth rate is maximum, as it grows along the electron beam. In addition, the frequency for the maximum growth rate is found to shift gradually to higher values. Furthermore, by gradually increasing the permittivity of the dielectric waveguide along it, we can always get the maximum power spectral density at the frequency where the spatial growth rate initially becomes maximum at the input.

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.

The Cooper pairing interaction in high Tc oxide superconductor is discussed in terms of an empirical expression; TcDexp[1/g], gcωo which was derived in our previous investigation. The dual character of this expression consisting of the phonon Debye temperature D and electronic excitation ωo in the mid-infrared region can be interpreted on the basis of the phonon-assisted mechanism on carrier conduction and the electronic excitation. A tunneling spectrum here presented shows certain evidence of the phonon contribution. The characteristics of the long range superconductive proximity phenomena recently reported are also may be interpreted by this mechanism.

The well known optimum approach to detect spread spectrum signals transmitted in bursts over frequency selective radio channels is matched filtering, which performs despreading, and subsequent Viterbi equalization (VE) to cope with intersymbol interference (ISI). With respect to complexity, VE is feasible only if data modulation schemes with a few symbol levels as e.g. 2PSK are used and if the delay spread of the channel is not too large. The paper gives a survey of suboptimum data detectors based on linear block estimation. Such data detectors are less expensive than VE especially in the case of multilevel data modulation schemes as 4PSK or 16QAM. Special emphasis is laid on data detectors based on Gauss-Markoff estimation because these detectors combine the advantages of unbiasedness and minimum variance of the estimate. In computer simulations, the Gauss-Markoff estimation algorithm is applied to spread spectrum burst transmission over radio channels specified by COST 207. It is shown that the SNR degradation which is a measure of the suboptimality of the detector does not exceed a few dB, and that even moderate spectrum spreading considerably reduces the detrimental effect of channel frequency selectivity.

It is concluded from numerical examples for the well-known linear PN sequence families of a large range of periods that the mean-square cross-correlation value between sequences is the dominating parameter to the average signal-to-noise power ratio performance of an asynchronous direct-sequence (DS) code-division multiple-access (CDMA) system. The performance parameters derived by Pursley and Sarwate are used for numerical evaluation and the validity of conclusion is supported by reviewing the other related works. The mean-square periodic cross-correlation takes the equal value p (code period) for the known CDMA code families. The equal mean-square cross-correlation performance results from the basic results of coding theory.

This paper presents a new idea of photometric stereo method which uses 3 point light sources as illumination source. Its intention is to extract the 3-D information of gastric surface. The merit of this method is that it is applicable to the textured and/or specular surfaces, moreover whose environment is too narrow, like gastric surface. The verification of the proposed method was achieved by the theoretical proof and experiment.

A new method of multipath diversity combination is proposed for Direct Sequence Spread Spectrum (DS-SS) mobile communications. In this method, the transmitted signal from the base staion is the sum of a number of the same spread signal, each one delayed and scaled according to the delay and the strength of the multipaths of the transmission channel. As a result the received signal at the mobile unit will already be a Rake combination of the multipath signals. This new method is called Pre-Rake diversity combination because the Rake diversity combination process is performed before transmission By this method the size and complexity of the mobile unit can be minimized, and the unit is made as simple as a non-combining single path receiver. A theoretical examination of the Signal to Noise Ratio (SNR) and the Bit Error Rate (BER) results for the traditional Rake and the Pre-Rake combiners as well as computer simulations show that the performance of the Pre-Rake combiner is equivalent to that of the Rake combiner.

The capacity of a cellular direct sequence code division multiple access system is investigated in situations with and without power control for both the reverse link (from mobile to base station) and the forward link (from base station to mobile). The capacity is defined as the number of simultaneous users per cell with a prespecified performance. A theoretical analysis of the effect of imperfect power control on the reverse link capacity is presented using an analytical model. To investigate the reverse link capacity without any form of power control, a general spatial user distribution is developed which is very suitable for analytical study of any multiple access system with the near-far effect problem. The performance of the reverse link of a CDMA system is also evaluated considering the users located in surrounding cells. Finally, the forward link capacity is studied considering multiple cells. Two possible forward power control schemes, namely carrier-to-interference ratio driven and distance driven systems, are discussed.

In this paper, we propose a spread spectrum pulse position modulation (SS-PPM) system, and describe its basic performances. In direct sequence spread spectrum (DS/SS) systems, pseudo-noise (PN) matched filters are often used as information demodulation devices. In the PN matched filter demodulation systems, for simple structure and low cost of each receiver, it is desired that each demodulator uses only one PN matched filter, and that signals transmitted from each transmitter are binary. In such systems, on-off keying (SS-OOK), binary-phase-shift keying (SS-BPSK) and differential phase-shift keying (SS-DPSK) have been conventionally used. As one of such systems, we propose the SS-PPM system; the SS-PPM system is divided into the following two systems: 1) the SS-PPM system without sequence inversion keying (SIK) of the spreading code (Without SIK for short); 2) the SS-PPM system with SIK of the spreading code (With SIK for short). As a result, we show that under the same bandwidth and the same code length, the data transmission rate of the SS-PPM system is superior to that of the other conventional SS systems, and that under the same band-width, the same code length and the same data transmission rate, the SS-PPM system is superior to the other conventional SS systems on the following points: 1) Single channel bit error rate (BER) (BER characteristics of the SS-PPM system improve with increasing the number of chip slots of the SS-PPM system, and as the number of chip slots increases, it approaches Shannon's limit); 2) Asynchronous CDMA BER; 3) Frequency utilization efficiency. In addition, we also show that With SIK is superior to Without SIK on these points.

In this paper, an optimized wideband channel sounder designed for measuring the time variant impulse response of outdoor radio channels in the frequency range 1800-2000 MHz is presented. Prior to hardware implementation the system was first modelled on a high performance supercomputer to enable the system designer to optimize the digital signal processing algorithms and the parameters of the hardware components by simulation. It is shown that the proposed measuring system offers a significantly larger amplitude resolution, i.e. dynamic range, than conventional systems applying matched filtering. This is achieved by transmitting digitally generated periodic spread spectrum test signals adjusted to amplifier non-linearities and by applying optimum unbiased estimation instead of matched filtering in the receiver. A further advantage of the hardware implementation of the proposed system compared to conventional systems [5]-[7] is its high flexibility with respect to measuring bandwidth, period of the test signal and sounding rate. The main features of the optimized system are described and first measurement results are presented.

The bit error performance of a Direct Sequence Spread Spectrum Communication system in actual land mobile satellite channel is evaluated with experiments. Field test results with the ETS-V satellite in urban and suburban environments at L-band frequency show that this land mobile satellite channel of 3MHz bandwidth can be seen as a non-frequency selective Rician fading channel as well as shadowing channel. The bit error performance can be estimated from signal power measurement as in the case of narrow band modulation signals.

The spread spectrum system (abbreviated as SS system) is known to be an excellent communication system which resists jamming. Recently, its application to a simplified wireless communication system has been considered to be suited for consumer communication. In Japan, SS wireless LAN system has got the approval on 2.4GHz ISM band already. A compact SS transceiver for the SS wireless LAN is realized, whose data ratio is 230kbps. The SS transceiver is based on a direct sequence for the modulation, and the demodulation is carried out by a specially developed SAW device. In the first part of this paper, the technical conditions of the SS wireless LAN are mentioned. Then the SAW device and the principle of the demodulation are discussed. Finally, the configuration of the SS transceiver and the protocol of the SS wireless LAN are presented.

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.