An acoustic echo canceller that also cancels room noise is proposed. This system has an additive (noise reference) input port, and a noise canceller (NC) precedes the echo canceller (EC) in a cascade configuration. The adaptation control problem for the cascaded echo and noise canceller is solved by controlling the adaptation process to match the occurrence of intermittent speech/echo; the room noise is a stationary signal. A simulation shows that adaptation using the NLMS algorithm is very effective for the echo and noise cancellation. Sub-band cancelling techniques are utilized. Noise cancellation is realized with a lower band EC. Hardware is implemented and its performance evaluated through experiments under a real acoustic field. The combination of the EC with NC maintains excellent performance at all echo to room noise power ratios. It is shown that the proposed canceller overcomes the disadvantages traditionally associated with ECs and NSc.

Existing solid models often contain redundant primitives and null blocks, which both slows down the rendering process and makes the process complex. There has been recent progress toward solving this problem, but existing modeling schemes cannot support eliminating all the redundancies, especially the null blocks, from the solid models. This paper proposed a technique that can eliminate redundancies. By dividing a primitive into some surface dispersed points, a new primitive representation is obtained. The sample segments of the primitive or the object are used to locate composition position to prevent the null primitives from being generated. By drawing out the geometric shape points set corresponding to a common acting area, the volume boundary of a primitive or an object is evaluated by only the Boolean set operations. The null blocks can be picked out in terms of the volume boundary. The resulting solid model generated in this way has no redundancies and is suitable for fast rendering of the image.

A newly developed planar inductor and its application to dc-dc converters are described. The planar inductor consists of a planar spiral coil and soft magnetic sheets, it has a small size (11110.8mm), 33µH inductance and a maximum quality factor of 14. The step down chopper dc-dc converter has been developed by using planar inductor, which has small size (20154mm), 5V-2W typical output and output power/volume ratio of 1.7W/cc. The switching converter can be miniaturized by using the planar inductor.

In complex self-checking systems several blocks (i.e. functional blocks and checkers) are embedded. In order to check the self-checking properties of such blocks we need to know the set of vectors they receive from the blocks feeding their inputs (i.e. the code word output spaces of the source blocks). In a complex system the computation of the output spaces by means of exhaustive simulation of the system is intractable. In this paper we present a tool which performs this computation with low CPU time. Some other tools allowing to verify the self-checking properties of embedded blocks (like the strongly fault secure property of embedded PLAs and the self-testing property of embedded checkers), have also been developed and experimented.

It has been reported that the efficiency of a low voltage power supply is improved by replacing diodes in an output-stage with synchronous rectifiers (SR). A SR consists of a bipolar junction transistor with a low-saturation voltage and a current transformer. Although the SR has low offset-voltage, its reverse recovery characteristic is usually poor. In this paper, an RCD circuit which improves the reverse recovery characteristic of the SR is proposed. This circuit is simple, and it is composed of a diode, a capacitor and a resistor. The analysis and the experimental results of the SR with the proposed RCD circuit are presented. The optimum design of the RCD to improve the reverse recovery characteristic of SR is discussed.

A new indexing technique for rapid evaluation of nested query on composite object is propoced, reducing the overall cost for retrieval and update. An extended B+ tree is introduced in which object identifier (OID) to be searched and path information usud for update of index record are stored in leaf node and subleaf node, respectively. In this method, the retrieval oeration is applied only for OIDs in the leaf node. The index records of both leaf and subleaf nodes are updated in such a way that the path information in the subleaf node and OIDs in the leaf node are reorganized by deleting and inserting the OIDs. The techniaue presented offers advantages over currently related indexing techniques in data reorganization and index allocation. In the proposed index record, the OIDs to be reorganized are always consecutively provided, and thus only the record directory is updated when an entire page should be removed. In addition, the proposed index can be allocate to a path with the length greater than 3 without splitting the path. Comparisons under a variety of conditions are given with current indexing techniques, showing improved performance in cost, i.e., the total number of pages accessed for retrieval and update.

This paper describes the derivation of a parallel program from a nondeterministic sequential program using a bottom-up parser as an example. The derivation procedure consists of two stages: exploitation of AND-parallelism and exploitation of OR-parallelism. An interpreter of the sequential parser BUP is first transformed so that processes for the nodes in a parsing tree can run in parallel. Then, the resultant program is transformed so that a nondeterministic search of a parsing tree can be done in parallel. The former stage is performed by hand-simulation, and the latter is accomplished by the compiler of ANDOR-, which is an AND/OR parallel logic programming language. The program finally derived, written in KL1 (Kernel Language of the FGCS Project), achieves an all-solution search without side effects. The program generated corresponds to an interpreter of PAX, a revised parallel version of BUP. This correspondence shows that the derivation method proposed in this paper is effective for creating efficient parallel programs.

MISRs are widely used as signature circuits for VLSI built-in self tests. To improve the aliasing probability of MISRs, multiple MISRs and M-stage MISRs with m inputs are available, where M is grater than m. The aliasing probability as a function of the test length is analyzed for the compaction circuits for a binary symmetric channel. It is observed that the peak aliasing probability of the double MISRs is less than that of M-stage MISRs with m inputs. It is also shown that the final aliasing probability for a multiple MISR with d MISRs is 2dm and that for an M-stage MISR with m imputs is 2M if it is characterized by a primitive polynomial.

Strongly Fault-Secure (SFS) circuits are known to achieve the TSC goal of producing a non-codeword as the first erroneous output due to a fault. Strongly Code-Disjoint (SCD) circuits always map non-codeword inputs to non-codeword outputs even in the presence of faults so long as the faults are undetectable. This paper presents a new generalized design method for the SFS and SCD realization of combinational circuits. The proposed design is simple, and always gives an SFS and SCD combinational circuit which implements any given logic function. The resulting SFS/SCD circuits can be connected in cascade with each other to construct a larger SFS/SCD circuit if each interface is fully exercised.

In this paper we present and analyze multi-level quorum schemes for maintaining the consistency of replicated data in the presence of concurrency and failures in a large distributed environment. The multi-level quorum method operates on a logical hierarchy of the nodes in the network and applies well known flat voting algorithms for replicated data concurrency control in a layered fashion. We show how the number of hierarchy levels, the number of logical entities per level and the voting algorithms used on each level affect the costs and the degree of availability associated with a wide range of multi-level quorum schemes. The results of the analysis are used to provide guidelines for designing the most suitable multi-level quorum strategy for a given application scenario. Comparative performance measurements in a simulated network are presented to illustrate the properties of multi-level approaches when some of the assumptions of the analytical investigation do not hold.

This paper proposes a method to estimate the waveform of a specified sound source in a noisy and reverberant environment using a sensor array. Previously, we proposed an iterative method to estimate the waveform. However, in this method the effect of reflection sound reduces to 1/M, where M is the number of microphones. Therefore, to solve the reverberation problem, we propose a new method using inverse filters of the transfer functions from the sound sources to each microphone. First, the transfer function from each sound source to each microphone is measured by the cross-spectrum technique and each inverse filter is calculated by the QR method. Then the initially estimated waveform of a sound source is the averaged signal of the inverse filter outputs. Since this waveform still contains the effects of the other sound sources, the iterative technique is adopted to estimate the waveform more precisely, reducing the effects of the other sound and the reflection sound. Some computer simulations and experiments were carried out. The results show the effectiveness of our method.

This paper describes a new trial of dynamical parameter estimation for the actual room acoustic system, in a practical case when the input excitation is polluted by a background noise in contrast with the usual case when the output observation is polluted. The room acoustic system is first formulated as a discrete time model, by taking into consideration the original standpoint defining the system parameter and the existence of the background noise polluting the input excitation. Then, the recurrence estimation algorithm on a reverberation time of room is dynamically derived from Bayesian viewpoint (based on the statistical information of background noise and instantaneously observed data), which is applicable to the actual situation with the non-Gaussian type sound fluctuation, the non-linear observation, and the input background noise. Finally, the theoretical result is experimentally confirmed by applying it to the actual estimation problem of a reverberation time.

This paper presents a new type of zero-voltage-switched (ZVS) push-pull dc-dc converter with two synchronous rectifiers in the secondary circuit. ZVS is realized using the magnetizing current of the transformer as a constant current source during the commutation. The output voltage is controlled by PWM with a constant switching frequency. The circuit operation is described using equivalent circuits. The steady-state and dynamic characteristics are analyzed and confirmed experimentally.

Inverse filters can be designed in order to enhance the accuracy with which signals recorded in a given space can be reproduced in a given listening space. The problem is considered here of the design of an inverse filter matrix which enables K recorded signals to be accurately reproduced at K points in the listening space when transmitted via M loudspeaker channels. The analysis is sufficiently general to incorporate the case when the best (least squares) approximation is sought to the reproduction of K signals at L points in the space when LK. An analysis is presented which demonstrates that the approach suggested by the Multiple-Input/Output Inverse Filtering theorem of Miyoshi and Kaneda can be realised adaptively by using the Multiple Error LMS algorithm of Elliott et al.

When a group of developers are involved in the distributed development of some software product, they must communicate with one another frequently to exchange information about the product. To reduce the penalty of communication, the support environment should provide developers with their necessary information and update the information automatically while the product is modified by developers. Furthermore, the environment must meet the following requirements despite of workstation failures: whether a specific information is correct or not should always be decidable; as much information as possible should be updated correctly and efficiently. This paper presents a framework to construct such a fault-tolerant environment based on attribute grammars. In the framework, a product is represented by an attributed tree, which is partitioned into several subtrees {T1,,Tm}. Attribute values in each subtree Ti(1im) express the information about the product required by a developer. We introduce a set of redundant data and algorithms to meet the fault-tolerance requirements mentioned above. The correctness of an attribute value in Ti can then be decided in O(mn0log n) time, where n0n, and n is the number of attribute instances in Ti. All available attribute values can be updated with time complexity O(m2n1 log n) and communication complexity O(m2), where n1 is the number of attribute instances that must be reevaluated.

Fahlman and Lebiere's (FL) learning algorithm begins with a two-layer network and in course of training, can construct various network architectures. We applied FL algorithm to the same three-layer network architecture as a back propagation (BP) network and compared their generalization properties. Simulation results show that FL algorithm yields excellent saturation of hidden units which can not be achieved by BP algorithm and furthermore, has more desirable generalization ability than that of BP algorithm.

This paper describes a nonstationary spectral analysis method and its application to prognosis and diagnosis of automobiles. An instantaneous frequency spectrum is considered first at a single point of time based on the instantaneous representation of autocorrelation. The spectral distortion is then considered on two-dimensional spectrum, and the filtering is introduced into the instantaneous autocorrelations. By the above procedure, the Instantaneous Covariance method (ICOV), the Instantaneous Maximum Entropy Method (IMEM), and the Wigner method are shown and they are unified. The IMEM is used for the time-dependent spectral estimation of vibration and acoustic sound signals of automobiles. A multi-dimensional (M-D) space is composed based on the variables which are obtained by the IMEM. The M-D space is transformed into a simple two-dimensional (2-D) plane by a projection matrix chosen by the experiments. The proposed method is confirmed useful to analyze nonstationary signals, and it is expected to implement automatic supervising, prognosis and diagnosis for a traffic system.

This paper describes a new timing calibration method for IC testers that uses a Timing Calibration Device (TCD). The TCD is a chip fabricated using the same process the device to be tested. Since the TCD has the same assignment pins as the LSI memory device under test (called the "MUT"), it enables an IC tester to evaluate the timing accuracy at the input/output terminal of MUT. The block-select-access time of a 1 K ECL RAM, which is less than 3.0 nanoseconds, has been accurately measured using this device. A timing-calibration subsystem is proposed for IC testers as an application of the TCD. Such a device would achieve precise measurement of high-speed LSI memory devices.

A new design method is proposed for realizing a hypercube network (HC) structured multicomputer system on a wafer using wafer-scale integration (WSI). The probability that an HC can be constructed on a wafer is higher in this method than in the conventional method; this probavility is called a construction probability. We adopt the FUSS method for the processor (PE) address allocation in our desing because it has a high success probability in the allocation. Even if the design renders the address allocation success probalility hegher, it is of no use if it makes either the maximum wiring length between PEs or the array size (wiring area) larger. A new wiring channel structure capable of connecting PEs on a wafer is proposed in this paper, where a channel, called a basic channel, is used. A one-dimensional-array sub-HC row network (RN) or column networks (CN) can be constructed using the basic channel. The sub-HC construction method, which embeds wirings into the basic channel, is also proposed. It requires almost the same wiring width as conventional method. However, it has an advantage in that maximum wiring length between PEs can be about half that of the conventional method. If PEs must be shifted in the case of PE defects, they can be shifted and connected to the basic channel using other PE shifting channels, and an RN or CN can be constructed. The maximum wiring length between PEs, array size, and construction probability will also be derived, and it will be shown that the proposed design is superior to the conventional one.

This paper presents a new method for estimating both the parameters of a nonminimum phase system and its unknown input signal. An approximate inverse system method is used to estimate the unknown input signal, and then, by using a Kalman filter, approximately consistent parameter estimates of the nonminimum phase system can be obtained effectively. This method can be used to estimate the parameters of a nonminimum phase system and a minimum phase one in the case when the input signal is a white noise or an impulse sequence.