Most research on the estimation of direction of arrival (DOA) has been performed based on the assumption that the signal sources are point sources. In some real surroundings, signal source localization can more adequately be accomplished with distributed source models. When the signal sources are distributed over an area, we cannot directly use well-known DOA estimation methods, because these methods are established based on the point source assumption. In this paper, we propose a 3-dimensional distributed signal source model, in which a source is represented by two parameters, the center angle and degree of dispersion. Then, we address the estimation of the elevation and azimuth angles of distributed sources based on the parametric distributed source modeling in the 3-dimensional space.

In this paper, we first analyze the resolution performance of the Gerschgorin radii based source number estimator (GDE, Gerschgorin Disk Estimator) proposed in [1] for independent closely-spaced plane waves. Based upon this analysis, we verify the resolution threshold of the Gerschgorin radii based method for two sources. New close-form expressions of the Gerschgorin radii are formulated and examined. For improvement of detection performance, we then further propose a enhanced GDE method (EGDE). Examples and comparisons with methods based on Gerschgorin radii and weighted Gerschgorin radii, as well as conventional methods are included. Finally, multi-source and/or closely spaced source problems are discussed.

We have fabricated a new analog memory for integrated artificial neural networks. Several attempts have been made to develop a linear characteristics of floating-gate analog memorys with feedback circuits. The learning chip has to have a large number of learning control circuit. In this paper, we propose a new analog memory SDAM with three cascaded TFTs. The new analog memory has a simple design, a small area occupancy, a fast switching speed and an accurate linearity. To improve accurate linearity, we propose a new chargetransfer process. The device has a tunnel junction (poly-Si/poly-Si oxide/poly-Si sandwich structure), a thin-film transistor, two capacitors, and a floating-gate MOSFET. The diffusion of the charges injected through the tunnel junction are controlled by a source follower operation of a thin film transistor (TFT). The proposed operation is possible that the amounts of transferred charges are constant independent of the charges in storage capacitor.

A new source line routing architecture features a blanket-like source line made of double aluminum layers by utilizing a pure tungsten metal layer as the local interconnection layer in the peripheral region. The relaxed pitch of the signal lines improves the RC time delay constant of the signal lines and gives stable Vcc and Vss levels throughout the chip. Furthermore, this architecture brings about an 8% area reduction of the peripheral region in 256 Mb DRAMs with high performance,when used in collaboration with hierarchical bit-line architecture.

It is well recognized that the electromagnetic interference due to indirect electrostatic discharge (ESD) is not always proportional to the ESD voltage and also that the lower voltage ESD sometimes causes the more serious failure to high-tech information equipment. In order to theoretically examine the peculiar phenomenon, we propose an analytical approach to model the indirect ESD effect. A source ESD model is given here using the spark resistance presented by Rompe and Weizel. Transient electromagnetic fields due to the ESD event are analyzed, which are compared with the experimental data carefully given by Wilson and Ma. A model experiment for indirect ESD is also conducted to confirm the validity of the ESD model presented here.

A new line simulator, SEMALIS has been developed. This simulator can handle complicated lot processings to maintain processing quality and efficient line operations to improve line performance. The current manufacturing line consists of five resource models: lot, process sequence, equipment, lot processing, and line operations. The parameters of these models are defined so as to accurately reflect the state of the line operations. From our simulation results, we confirmed that SEMALIS accurately identifies bottlenecks or starvations where equipment can be added or reduced to optimize equipment utilization through resource planning, and that SEMALIS can also be used to evaluate the long-term effects of line operating methods on the line performance of ASIC manufacturing lines.

This paper proposes a new emphasizing three-dimensional pointing device considering user friendliness and lack of cable clutter. The proposed method utilizes five degrees of freedom via the medium of non-verbal voice of human. That is, the spatial direction of the sound source, the type of the voice phoneme and the tone of the voice phoneme are utilized. The input voice is analyzed regarding the above factors and then taking proper effects as previously defined for human interface. In this paper the estimated spatial direction is used for three-dimensional movement for the virtual object as three degrees of freedom. Both of the type and the tone of the voice phoneme are used for remaining two degrees of freedom. Since vocalization of nonverbal human voice is an everyday task, and the intonation of the voice can be quite easily and intentionally controlled by human vocal ability, the proposed scheme is a new three-dimensional spatial interaction medium. In this sense, this paper realizes a cost-effective and handy nonverbal interface scheme without any artificial wearing materials which might give a physical and psychological fatigue. By using the prototype the authors evaluate the performance of the scheme from both of static and dynamic points of view and show some advantages of look and feel, and then prospect possibilities of the application for the proposed scheme.

This paper presents some tighter bounds on universal noiseless coding, in particular, the lowerbound tighter than Davisson et al.'s for finite sequence and the upperbound for some typical universal data compression. We find that Davisson et al.'s bound satisfies some optimization in the case of using the Jeffreys prior and also that the derived upperbound in this paper is within O(1/n) from the Clarke and Barron asymptotics in the case of some restricted typical universal data compression defined in the paper.

Slepian, Wolf and Wyner proved famous source coding theorems for correlated i.i.d. sources. On the other hand recently Han and Verdú have shown the source and channel coding theorems on general sources and channels whose statistics can be arbitrary, that is, no assumption such as stationarity or ergodicity is imposed. We prove source coding theorems on correlated general sources by using the method which Han and Verdú developed to prove their theorems. Also, through an example, we show some new results which are essentially different from those already obtained for the i.i.d. source cases.

A problem of synthesizing an optimal file transfer on a file transmission net N is to consider how to distribute, with a minimum total cost, copies of a file J with some information from source vertex set S to all vertices of N by the respective vertices' copy demand numbers. The case of |S| =1 has been studied so far. This paper deals with N such that |S|1, where a forest-type file transfer is defined. This paper proposes a polynomial time algorithm to synthesize an optimal forest-type file transfer on such N satisfying SM U, where M and U are mother vertex set and positive demand vertex set of N, respectively.

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.

In this paper, two models are proposed for the simulation of MPEG video sources in ATM networks. The projected autoregressive (PAR) model is based on the autoregressive (AR) model compensated by a projection function. The projection function is capable of adjusting the histogram generated by the AR model so that it better fits the histogram obtained from real data. The state transition (ST) model is developed on die basis of recording the variation of frame size in a video sequence. Each state denotes the size of a frame and the number of state depends on the degree of correlation between frames. Our results show that the histogram generated by the ST model is almost identical with that of the real data and the PAR model performs better in capturing the property of autocorrelation of real data. When compared with other models, both of the two models demonstrate an excellent property of fitting the complex histogram curve, which was not achieved by the AR model, and preserving the correlation characteristics. A heuristic search algorithm is also proposed to make our modelling processes more efficient.

This paper describes a SQUID magnetometer and the measurement of small signals. It also describes the current state of SQUID technology developed in the SSL project.

Data communication by using TCP/IP is one of important services on ATM networks. At one approach in traffic control of this service, the dedicated bandwidth for data transfer is not guaranteed and the feedback congestion control to prevent cell loss is performed in the congestion case. However, when a large quantity of data is transferred within a short period, this traffic control cannot be expected to achieve high efficiency. In this case, it is suitable that the dedicated bandwidth is guaranteed by FRP (Fast Reservation Protocol) before the data is transferred. This paper describes that FRP is superior to the feedback congestion control for large size data transmission. Next, it proposes a selectable traffic control which selects adaptively one of the feedback congestion control and FRP.

Telecommunications management activities have mostly been supported by operators; however, machines are gradually playing more important roles in the management arena by utilizing computing technology. Additionally, management systems can now be networked by using standard interface specifications. The study of human and machine integration is thus essential for achieving the sophisticated management objectives of telecommunications. This paper proposes the principles for a telecommunications management integration network (TMIN), which integrates human and machine management networks, and proposes a source text description method for transferring management communication knowledge from human to machine. First, reference models are proposed for the management process and management communication. These models cover network management activities of both humans and machines. Second, the contents of the source text are clarified. Source text presents human management knowledge in a form suitable for machine-machine communication. Third, an efficient source text description method is proposed that reduces redundancy and proliferation. Finally, a means of harmonizing management information definitions with TMIN is suggested to facilitate human-machine cooperation.

A novel effective channel length extraction method has been developed, which utilizes the difference between the local threshold voltage of channel region and that of external region. In this method, the dependence of external resistance on Vg is taken into account, and it is not necessary to extract Vth. It is found that the external resistance can be approximated as the linear function of Vg with Vg around Vth. For a 0.4 µm gate length LDD MOSFET, the accuracy and resolution are estimated to be less than 0.02 µm and 0.003 µm, respectively.

In this paper, we developed the analytical method for the radiation field from a line current source placed in a stratified slab waveguide. This method is applicable to the analysis of excitation problem of inhomogeneous slab waveguide by increasing the number of layers. The numerical results are given for the cases of five layers, such as W and M type waveguides, and the inhomogeneous slab waveguide. The influence of guided and leaky modes on the radiation field are studied.

A new method for the accurate extraction of glottal source parameters is proposed. This method, called Heuristic Analysis-by-Synthesis (HAbS), has been developed specifically to overcome the weaknesses of other methods of glottal source parameter extraction. The specific features of this method are the use of the AbS method for extraction of glottal source and vocal tract parameters, the use of a parametric glottal source model during vocal tract analysis, the use of alternating glottal source and vocal tract analyses, and simultaneous, time-domain analysis of the glottal source parameters and the first formant. This method has been implemented in such a way that user interaction is not required. The performance of the HAbS method is evaluated using both synthetic-speech and natural-speech data. Error is measured in both the time domain and the spectral domain, and the standard deviation of extracted parameter values is computed. In addition, the error in analysis of each glottal-source parameter is computed using synthetic-speech data. In order to assess the accuracy of the HAbS method as compared to other methods, three other methods (LPC, AIF, and AbS) are evaluated using the same data methods of error measurement. From these evaluations, it is clear that the HAbS method yields results that are more accurate than these other methods.

This article shows construction of an asymptotically optimal source code for transmitting sentences together with truth values on a [0,1]-valued logic system.

Recent achievements in low-voltage and low-power circuit techniques are reported in this paper. DC current in low-voltage CMOS circuits stemming from the subthreshold current in MOS transistors, is effectively reduced by applying switched-power-line schemes. The AC current charging the capacitance in DRAM memory arrays is reduced by a partial activation of array blocks during the active mode and by a charge recycle during the refresh mode. A very-low-power reference-voltage generator is also reported to control the internal chip voltage precisely. These techniques will open the way to using giga-scale LSIs in battery-operated portable equipment.