The effectiveness of random-switching control, by which the switching-noise spectrum is spread and its level is reduced, is briefly described through experimental results. The noise spectrum by random switching is analyzed in general approach including a noise-generation model and a switching function with random process. Taking the normal distribution as an instance, the discussion on the amount of random perturbation is made quantitatively. The validity of the analysis is confirmed experimentally by a series of pulse serving as ideal switching-noise.

This paper describes configuration and operation of a high-frequency link resonant inverter using cycloconverter techniques. In this inverter, a resonant link high-frequency voltage generated in a primary resonant inverter is isolated by a high-frequency transformer, then directly converted into a resonant link low-frequency voltage in a cycloconverter. The switching losses and surge voltage levels can be reduced by making all switches in the primary inverter and the cycloconverter operate at zero voltage. The relationship between characteristic impedance of the resonant circuit and the conversion efficiency, and the distortion factor characteristics of the output voltage waveforms are discussed by comparing of analytical and experimental results.

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.

This paper describes a novel Recessed Stacked Capacitor (RSTC) structure for 256 Mbit DRAMs, which can realize the requirements for both fine-pattern delineation with limited depth of focus and high cell capacitance. New technologies involved are the RSTC process, 0.25 µm phase-shift lithography and CVD-tungsten plate technology. An experimental memory array has been fabricated with the above technologies and 25 fF/cell capacitance is obtained for the first time in a 0.61.2 µm2 (0.72 µm2) cell.

This paper describes the delay time on bit lines due to coupling capacitance between adjacent bit lines in megabit SRAM's. The delay time on bit lines in several generations of megabit SRAM's is quantitatively analyzed using device and circuit simulations. It is shown that narrowing the bit-line swing from 200 mV to 30 mV for future 16-Mbit SRAM's will effectively reduce the difference in delay time from 1.0 ns to 0.3 ns, and that a two-block devided bit line will lower the difference in the delay-time ratio to 3% in case of 15-ns access time.

The paper considers the design of two families of binary block codes developed for controlling large numbers of errors which may occur in LSI, optical disks and other devices. The semidistance codes are capable of assuring a required signal-to-noise ratio in information retrieval; the t-symmetric error correcting/all unidirectional error detecting" (t-SyEC/AUED) codes are capable of correcting t or fewer symmetric errors and also detecting any number of unidirectional errors caused by the asymmetric nature of transmission or storage madia. The paper establishes an equivalence between these families of codes, and proposes improved methods for constructing, for any values of t, a class of nonsystematic constant weight codes as well as a class of systematic codes. The constructed codes of both classes are shown to be optimal when t is O, and of asymptotically optimal order" in general cases. The number of redundant bits of the obtained nonsystematic code is of the order of (t+1/2)log2 K bits, where K is the amount of information encoded. The obtained systematic codes have redundancy of the order of (t+1)log2 K bits.

A new circuit and a transformer structure is described for a high-input-voltage converter operating at a high switching frequency. The two-MOSFET forward converter is suitable for a high-input-voltage converter. To increase the switching frequency, the reset period of the transformer core flux must be reduced. There are a few methods for decreasing the reset period. Increasing the transformer flyback voltage and reducing its stray capacitance are effective in decreasing the reset period without increasing power loss. A new two-MOSFET forward converter is proposed which uset the increased flyback voltage and a transformer structure to reduce the stray capacitance. The new converter using this transformer provides the basis for a 48-V, 100-W output, 270-V input converter operating at 200kHz with high efficiency (above 95%).

The characteristics of voltage-resonant dc-dc converters have already been analyzed and described. However, in the conventional analysis, the inductance of the reactor is assumed to be infinity and the loss resistance of the power circuit is not taken into account. Also, in some cases, the averaging method is applied to analyze the resonant dc-dc converters as well as the pwm dc-dc converters. Consequently, the results from conventional analysis are not entirely in agreement with the experimental ones. This paper presents a general design-oriented analysis of the buck-boost type voltage-resonant dc-dc converter in the continuous and discontinuous modes of the reactor current. In this analysis, the loss resistance in each part of the power circuit, the inductance of the reactor, the effective value (not mean value) of the power loss, and the energy-balance among the input, output and internal-loss powers are taken into account. As a result, the behavior and characteristics of the buck-boost type voltage-resonant dc-dc converter are fully explained. It is also revealed that there is a useful mode in the discontinuous reactor current region, in which the output voltage can be regulated sufficiently for the load change from no load to full load and for the relatively large change of the input voltage, and then the change in the switching frequency can be kept relatively small.

Signature analysis using a Multiple-Input LFSR as the output response compaction circuit is widely adopted in actual BIST schemes. While aliasing probabilities for random errors are usually very small, MI-LFSRs are tend to fail detecting diagonal errors. A spot error, which include the diagonal error as a particular case, is defined as multiple bit crrors adjacent in space and in time domain. Then, shuffling of interconnection between CUT output and MI-LFSR input is studied as a scheme to prevent aliasing for such errors. The condition for preventing aliasing due to a predetermined size of single spot error is shown. Block based shuffling and the shortened one are proposed to realize required distance properties. Effect of shuffling for multiple spot errors is examined by simulation showing that shuffling is effective also for a certain extend of multiple spot errors.

Heat recovery methods and the amount of heat that can be recovered from fuel cell exhaust gas is described. The cooling performance of an absorption refrigerator that uses fuel cell waste heat is also described. Two heat recovery methods from the exhaust gas are considered: one uses heat recovery from mixed exhaust gas from the cathode side of the cells and the reformer (mixed type); the other uses separate heat recovery from these sites (separate type). Simulation shows that the amount of heat recovered between 60 and 75 with the separate type of heat recovery is greater than with the mixed type of heat recovery. The cooling capacity of the refrigerator using the separate type heat recovery and recovering heat between 65 and 85 is about 2.5 times that of one using a generator (heat source) with a constant 85 temperature.

A review is presented of the fundamental principles underlying modern techniques for the active control of acoustic noise. The basic physical principles are first dealt with in the context of the active control of free field radiation and the classical approaches to the problem are briefly discussed. The active control of sound fields in ducts and enclosures is also described and the inherent physical limitations of the technique are emphasised. Modern signal processing methods for realising feedforward control systems are also outlined and least squares formulations are presented which enable performance limits to be established and adaptive algorithms to be derived.

Synchronous clocks are an essential requirement for a variety of distributed system applications. Many of these applications are safety-critical and require fault tolerance. In this paper, a general probabilistic clock synchronization model is presented. This model is uniformly probabilistic, incorporating random message delays, random clock drifts, and random fault occurrences. The model allows faults in any system component and of any type. Also, a new Sliding Window Clock Synchronization Algorithm (SWA) providing increased fault tolerance is proposed. The probabilistic model is used for an evaluation of SWA which shows that SWA is capable of tolerating significantly more faults than other algorithms and that the synchronization tightness is as good or better than that of other algorithms.

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 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 proposes a new adaptive algorithm of the FIR type digital filter for an acoustic echo canceller and similar application fields. Unlike an echo canceller for line, an acoustic echo canceller requires a large number of taps, and it must work appropriately while it is driven by colored input signal. By controlling the filter tap length and updating filter coefficients multiple times during a single sampling interval, the proposed algorithm improves the convergence characteristics of adaptation even if colored input signal is introduced. This algorithm is maned VT-LMS after variable tap length LMS. The results of simulation show the effectiveness of the proposed algorithm not only for white noise but also for colored input signal such as speech. The VT-LMS algorithm has better convergence characteristice with very little extra computational load compared to the conventional algorithm.

Unidirectional/Asymmetric error control codes have extensively been studied, not only from theoretical interest but from application to computer systems or communication systems. Recently, attention has been focused on detecting only d, not all, unidirectional errors, that is, d bits unidirectional error ditecting (d-UED) codes. Borden proposed an optimal nonsystematic d-UED code. This paper shows a new design method for cost-effective self-testing checker for the optimal d-UED code. The checking policy is to check whether condition of the Borden code satisfies or not. The proposed checker includes the parallel weight counter, the comparator and th e modulo adder in which new residue operation is defined and hence this makes the circuit self-testing. These circuits are designed to have all possible input patterns in order to satisfy self-testing property. Finally, the proposed checker has greatly reduced hardware amount compared to the existing one.

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.

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 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.

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.