A clock-feedthrough (CFT) compensation technique using a dummy cell is valid when the CFT current from a switched-current (SI) memory cell is signal-independent. Based on this idea, a SI dummy cell appropriate for the S2I cell is developed. Simulations show that the CFT rejection ratio as high as 60dB is attainable over the temperature range from -30 to 80 with this architecture. The CFT-compensated SI cell proposed here is, therefore, quite usuful for high-accuracy, current-mode signal processing.

Cross talk in mode-division multiplexing system is studied. The mode patterns when the fiber is bent are observed in the experiment.

This paper reviews recent developments in small-sized broadband antennas for EMI measurements, especially in the microwave frequency region. Transient EMI measurements are also discussed by introducing complex antenna factors and conversion of frequency-domain data into time-domain data. This paper also focuses on considerable improvements achieved in calibration techniques for conventional EMI antennas in VHF/UHF bands.

In this paper, we give a new lower bound for the dimension of subfield subcodes. This bound improves the lower bound given by Stichtenoth. A BCH code and a subfield subcode of algebraic geometric code on a hyper elliptic curve are discussed as special cases.

The usage of a diagram, which we call a state fence diagram (SFD), for analyzing discrete event systems such as reactive systems, is presented. This diagram is useful for event concurrent response and scenario analysis by using its three description styles.

This paper compares signal classification performance of multilayer neural networks (MLNNs) and linear filters (LFs). The MLNNs are useful for arbitrary waveform signal classification. On the other hand, LFS are useful for the signals, which are specified with frequency components. In this paper, both methods are compared based on frequency selective performance. The signals to be classified contain several frequency components. Furthermore, effects of the number of the signal samples are investigated. In this case, the frequency information may be lost to some extent. This makes the classification problems difficult. From practical viewpoint, computational complexity is also limited to the same level in both methods.IIR and FIR filters are compared. FIR filters with a direct form can save computations, which is independent of the filter order. IIR filters, on the other hand, cannot provide good signal classification deu to their phase distortion, and require a large amount of computations due to their recursive structure. When the number of the input samples is strictly limited, the signal vectors are widely distributed in the multi-dimensional signal space. In this case, signal classification by the LF method cannot provide a good performance. Because, they are designed to extract the frequency components. On the other hand, the MLNN method can form class regions in the signal vector space with high degree of freedom.

A new sealing structure for helical heater terminal used in a combination microwave oven is suggested, and its sealing characteristics are investigated. The new structure is simply a conductor tube coaxially covering the heater at the input terminal. With an optimally-designed structure of this type, the transmission coefficient of the heater can be reduced to -41.6 dB at 2450 MHz. When installed in a test set which is running with an output of 500 W at 2450 MHz, the microwave power leaking out from the heater terminal is found to be less than 0.2 mW/cm2.

The filtered-x algorithm, which is widely applied to active noise control system, requires setting a small step gain. Such a small step gain reduces the noise reduction effect when the alogrithm is implemented by fixed point processing. This paper presents an experimental result that the 'polarized-g' individually normalized least mean square (INLMS) algorithm can provide almost the same noise reduction effect even in the fixed point processing of 16 bits as that in floating point processing.

The Multiple input-output INverse/filtering Theorem (MINT) proves that N + 1 inverse filters are necessary to precisely control sound at N points in a space, and gives the minimum orders of such filters. In this paper, we propose the Indefinite MINT Filters (IMFs) for adding one or more control points to the above framework without increasing the number of inverse filters. Although the controllability of the new point is not sufficient, that of the other points is still maintained high enough by the principle of the MINT. In a two point sound control (using two inverse filters), the IMFs could reduce the squared error to the desired sound up to - 10 dB at the second point which is not controlled by the MINT.

This paper describes cellular spaces (or cellular automata) with capabilities of parallel self-reproduction and of parallel universal simulation of other cellular spaces. It is shown that there is a 1-dimensional cellular space U, called a parallel universal simulator, that can simulate any given 1-dimensional cellular space S in the sense that if an initial configuration of U has a coded information of both the local function and an initial configuration of S, then U has the same computation result that S has and the computation time of U is proportional to that of S. Two models of nontrivial parallel self-reproduction are also shown. One model is based on "state-exchange" method, and the other is based on a fixed point program of the parallel universal simulator.

In this paper we describe a method, which allows the likelihood normalization technique, widely used for speaker verification, to be implemented in a text-independent speaker identification system. The essence of this method is to apply likelihood normalization at frame level instead of, as it is usually done, at utterance level. Every frame of the test utterance is inputed to all the reference models in parallel. In this procedure, for each frame, likelihoods from all the models are available, hence they can be normalized at every frame. A special kind of likelihood normalization, called Weighting Models Rank, is also experimented. We have implemented these techniques in speaker identification system based on VQ-distortion codebooks or Gaussian Mixture Models. Evaluation results showed that the frame level likelihood normalization technique gives higher speaker identification rates than the standard accumulated likelihood approach.

This paper presents a theory based on combined differential- and common-mode propagation for crosstalk and transient analysis of pairs of asymmetric coupled interconnects with arbitrary time-invariant linear termination circuits. Time-domain solutions are obtained by an exact numerical inversion of Laplace transform (NILT). Two example circuits (coupled coplanar stripline and microstrip structures) are studied to examine the accuracy and efficiency of the present method.

Some kind of practical problems such as security verification of cryptographic protocols can be described as a problem to accomplish a given purpose by using limited operations and limited materials only. To model such problems in a natural way, unification problems under constrained substitutions have been proposed. This paper is a collection of results on the decidability and the computational complexity of a syntactic unification problem under constrained substitutions. A number of decidable, undecidable, tractable and intractable results of the problem are presented. Since a unification problem under constrained substitutions can be regarded as an order-sorted unification problem with term declarations such that the number of sorts is only one, the results presented in this paper also indicate how the intractability of order-sorted unification problems is reduced by restecting the number of sorts to one.

A new linearization technique of a transconductor is presented. The linearization is realized by using a differential current amplifier with an emitter-coupled pair. A specific value of the linearization parameter gives a maximally flat or an equiripple characteristic. Deviations from the theoretical characteristic can be adjusted by tuning the tail current of the emitter-coupled pair. The proposed technique is demonstrated by PSPICE simulation.

A two-dimensional receiver OEIC array having an address selector for highly parallel interprocessor networks has been realized. The receiver OEIC array consists of two-dimensionally arranged 1616 (256) optical receiver cells with switching transistors, address selectors (decoders), and a comparator. Each optical receiver comprises a pin PD and a transimpedance-type HBT amplifier. The HBT has an InP passivation structure to suppress the emitter-size effect, which results in the improvement of current gains, especially at low collector current densities. The receiver OEIC array was fabricated on a 3-inch diameter InP substrate with pin/HBT integration technology. Due to the function of address selection, only one cell is activated and the other cells are mute, so the receiver OEIC array shows low crosstalk and low power consumption characteristics. The array also shows a 266-Mb/s data transmission capability. This receiver OEIC array is a most complex InP-based OEIC ever reported. The realization of the two-dimensional receiver OEIC array promises the future interprocessor networks with highly parallel optical interconnections.

We present a very fast method for calculating an inverse filter for audio reproduction system. The proposed method of FFT-based inverse filter design, which combines the well-known principles of least squares optimization and regularization, can be used for inverting systems comprising any number of inputs and outputs. The method was developed for the purpose of designing digital filters for multi-channel sound reproduction. It is typically several hundred times faster than a conventional steepest descent algorithm implemented in the time domain. A matrix of causal inverse FIR (finite impulse response) filters is calculated by optimizing the performance of the filters at a large number of discrete frequencies. Consequently, this deconvolution method is useful only when it is feasible in practice to use relatively long inverse filters. The circular convolution effect in the time domain is controlled by zeroth-order regularization of the inversion problem. It is necessary to set the regularization parameter β to an appropriate value, but the exact value of β is usually not critical. For single-channel systems, a reliable numerical method for determining β without the need for subjective assessment is given. The deconvolution method is based on the analysis of a matrix of exact least squares inverse filters. The positions of the poles of those filters are shown to be particularly important.

An algorithm for extracting fundamental frequencies from duet sounds is proposed. The algorithm is based on an acoustical feature that the temporal fluctuation patterns in frequency an power are similar for harmonic components composing a sound for a single musical note played on a single instrument with a single active vibrating source. The algorithm is applied to the sounds of 153 combinations of pair-notes played by a flute duet and a violin duet. Experimental results show that the zone-wize correct identification rate by pitch name are 98% for the flute duet and 95% for the violin duet in the best cases.

The characteristics of high-performance InP-based monolithically integrated single and multiple channel photoreceivers with an InGaAs p-i-n photodiode and InAlAs/InGaAs HBTs, realized by one-step molecular beam epitaxy, are described. The monolithically integrated photoreceiver includes an integrated spiral inductor following the p-i-n diode at the input of the transimpedance amplifier to enhance the circuit response at high frequencies. Crosstalk of the multi-channel photoreceiver arrays is greatly reduced by applying both a metal ground shield and dual bias. The maximum measured -3 dB bandwidth of a single-channel integrated p-i-n/HBT photoreceiver is 19.5 GHz and the minimum crosstalk of the photoreceiver arrays, with an individual channel bandwidth of 11.5 GHz, is 36 dB. At these performance levels, these OEICs represent the state-of-the-art in multichannel integrated photoreceiver arrays.

Theoretical values of site attenuation for broadband receiving antenna or the antenna factor of broadband antenna over the frequency range from 30 MHz to 1 GHz have been calculated or measured by some researchers. For a frequency range over 1 GHz, wire antennas or horn antennas should be used. However, the theoretical site attenuation or antenna factor over 1 GHz have never yet been calculated. A CLS (Conical Log-periodic Spiral) antenna is generally used for EMC/EMI measurements in the microwave band as a broadband wire antenna for the swept frequency method. However, this antenna has the defect that its use results in the loss of polarization information. So we proposed an improved CLS antenna which has linear polarization. This new CLS antenna has another wire wound symmetrically to that of the standard CLS antenna. For this reason, we named it a double-wire CLS antenna. The double-wire CLS antenna loses no polarization information. We calculated the height pattern and the frequency characteristics of CSA (Classical Site Attenuation) for the double-wire CLS antenna when used for receiving, or used for both transmitting and receiving, as well as the antenna factor. Moreover, NSA (Normalized Site Attenuation) when the double-wire CLS antenna is used for receiving or used for both transmitting and receiving in free space were calculated.

The defects in the cold rolled strips have textural characteristics, which are nonuniform due to its irregularities and deformities in geometrical appearance. In order to handle the textural characteristics of images with defects, this paper proposes a surface inspection method based on textural feature extraction using the wavelet transform. The wavelet transform is employed to extract local features from textural images with defects both in the frequency and in the spatial domain. To extract features effectively, an adaptive wavelet packet scheme is developed, in which the optimum number of features are produced automatically through subband coding gain. The energies for all subbands of the optimal quadtree of the adaptive wavelet packet algorithm and four entropy features in the level one LL subband, which correspond to the local features in the spatial domain, are extracted. A neural network is used to classify the defects of these features. Experiments with real image data show good training and generalization performances of the proposed method.