This paper proposes a technique for estimating the harmonic frequencies based on instantaneous frequency (IF) of speech signals. The main problem is how to decompose the speech signal into the harmonic components. For this purpose, we use a set of bandpass-filters, each of whose center frequencies changes with time in order to track the instantaneous freuency of its output. As a result, the outputs of the band-pass filters become the harmonic components, and the instantaneous frequencies of the harmonics are accurately estimated. To evaluate the effectiveness of the approach, we apply it to pitch determination of speech. Pitch determination is simply accomplished by selecting the correct fundamental frequency out of the harmonic components. It is confirmed that the pitch extraction using the proposed pitch determination algorithm (PDA) is stable and accurate. The most significant feature of the PDA is that the extracted pitch contour is smooth and it requires no post-processing such as nonlinear filtering or any smoothing processes. Several examples are presented to demonstrate the capability of the harmonics estimation technique and the PDA.

This paper describes, from a system architectural viewpoint, how knowledge-based technologies have been utilized in developing EXLOG (an LSI circuit synthesis system) and SOFTEX (a software synthesis system) inside the authors' projects. Although the system architectures for EXLOG and SOFTEX started from the same production systems, consisting of transformation rules in the middle of the 1980's, both branched off in different directions in the 1990's. Based on experiences with EXLOG and SOFTEX, the differences between LSI and software design models are discussed, and the future directions are indicated for the knowledge-based design system architectures.

This paper presents a simple and efficient method for estimation of parameters useful for textured image analysis. On the basia of a 2-D Wold-like decomposition of homogenenous random fields, the texture field can be decomposed into a sum of two mutually orthogonal components: a deterministic component and an indeterministic component. The spectral density function (SDF) of the former is a sum of 1-D or 2-D delta functions. The 2-D autocorrelation function (ACF) of the latter is fitted to the assumed anisotropic ACF that has an elliptical contour. The parameters representing the ellipse and those representing the delta functions can be used to detect rotation angles and scaling factors of test textures. Specially, rotation and scaling invariant parameters, which are applicable to the classification of rotated and scaled textured images, can be estimated by combining these parameters. That is, a test texture can be correctly classified even if it is rotated and scaled. Several computer experiments on natural textures show the effectiveness of this method.

In case of object recognition using 3-D configuration data, the scale and poses of the object are important factors. If they are not known, we can not compare the object with the models in the database. Hence we propose a strategy for object recognition independently of its scale and poses, which is based on Hopfield neural network. And we also propose a strategy for estimation of the camera motion to reconstruct 3-D configuration of the object. In this strategy, the camera motion is estimated only with the sequential images taken by a moving camera. Consequently, the 3-D configuration of the object is reconstructed only with the sequential images. And we adopt the multiple regression analysis for estimation of the camera motion parameters so as to reduce the errors of them.

In this paper, a novel noise parameters extraction technique for microwave packaged BJT and FET is proposed. The noise parameters of packaged BJT and FET for the entire operating frequency band can be obtained from the four noise parameters measured at a single frequency or a few frequencies. The predicated results obtained with this method agree well with the measured data. As a result, the novel noise parameters extraction technique can be used to predict the noise with a minimum effort.

In this paper performance of M-stage detection for DS/CDMA is considered in terms of near-far resistance. Asymptotic multiuser efficiency (AME) of M stage detection over fading multipath channel is calculated and it is shown that even in the fading case the detector is near-far resistant i.e., insensitive to the relative energies of the users. The idea is extended to cellular environment. The effect of power control imperfection is investigated. It is shown that capacity can be increased if near-far resistant technique is employed in conjunction with limited power control.

Ever increasing demand for higher bandwidth memories, which is fueled by multimedia and 3D graphics, seems to be somewhat satisfied with various emerging memory solutions. This paper gives a review of these emerging DRAM architectures and a performance comparison based on a condition to let the readers have some perspectives of the future and optimized graphics systems.

A method of tone recognition has been developed for dissyllabic speech of Standard Chinese based on discrete hidden Markov modeling. As for the feature parameters of recognition, combination of macroscopic and microscopic parameters of fundamental frequency contours was shown to give a better result as compared to the isolated use of each parameter. Speaker normalization was realized by introducing an offset to the fundamental frequency. In order to avoid recognition errors due to syllable segmentation, a scheme of concatenated learning was adopted for training hidden Markov models. Based on the observations of fundamental frequency contours of dissyllables, a scheme was introduced to the method, where a contour was represented with a series of three syllabic tone models, two for the first and the second syllables and one for the transition part around the syllabic boundary. Corresponding to the voiceless consonant of the second syllable, fundamental frequency contour of a dissyllable may include a part without fundamental frequencies. This part was linearly interpolated in the current method. To prove the validity of the proposed method, it was compared with other methods, such as representing all of the dissyllabic contours as the concatenation of two models, assigning a special code to the voiceless part, and so on. Tone sandhi was also taken into account by introducing two additional models for the half-third tone and for the first 4th tone of the combination of two 4th tones. With the proposed method, average recognition rate of 96% was achieved for 5 male and 5 female speakers.

This paper presents a new Adaptive Convergence Factor (ACF) algorithm without the damping parameter adjustment acoording to the input signal and/or the composition of the filter system. The damping parameter in the ACF algorithms has great influence on the convergence characteristics. In order to examine the relation between the damping parameter and the convergence characteristics, the normalization which is realized by the related signal terms divided by each maximum value is introduced into the ACF algorithm. The normalized algorithm is applied to the modeling of unknown time-variable systems which makes it possible to examine the relation between the parameters and the misadjustment in the adaptive algorithms. Considering the experimental and theoretical results, the optimum value of the damping parameter can be defined as the minimum value where the total misadjustment becomes minimum. To keep the damping parameter optimum in any conditions, the new ACF algorithm is proposed by improving the invariability of the damping parameter in the normalized algorithm. The algorithm is investigated by the computer simulations in the modeling of unknown time-variable systems and the system indentification. The results of simulations show that the proposed algorithm needs no adjustment of the optimum damping parameter and brings the stable convergence characteristics even if the filter system is changed.

This letter presents new estimation algorithm of ARMAX systems which do not always satisfy the strictly positive real (SPR) condition. We show how estimated parameters can converge to their true values based on the overparameterized system. Finally, the results of computer simulation are presented to illustrate the effectiveness of the proposed method.

This paper describes a parameter extraction system that can easily accommodate many MOSFET models. The model-adaptability is contributed by tow factors; a model-adaptable initial value estimation technique and an environment which stores and reuses extraction procedures. A designer can easily develop an extraction procedure for a new MOSFET model by modifying a procedure for another MOSFET model developed previously. We have verified that the system is suitable for major SPICE models.

A number of flexible control mechanisms used in buffer management, congestion control and bandwidth allocation have been proposed to improve the performance of ATM networks by introducing parameters, such as threshold, push-out probability and incremental bandwidth size of a virtual path, which are adjustable by network providers. However, it is difficult to adaptively adjust these parameters, since the traffic in ATM networks is further complicated by accommodating various kinds of services. To overcome the problem, we propose in this paper a control scheme based on the genetic algorithms and the neural estimator. The neural estimator forecasts the future QOS values for each candidate parameter set, and the genetic algorithms select the best one to control the real network. An example of buffer management in an ATM switch is examined in this paper. Simulation results show the effectiveness of the proposed control scheme in adaptively adjusting the parameter set even when the traffic environment and the QOS requirements are dynamically changing.

Some CMOS gates are topologically asymmetric in inputs, even though they are logically symmetric. It implies a possibility to reduce power consumption by optimizing signal assignment to the inputs. In this study we theoretically derive power consumption of 2-input NAND gate based on transition probability of input signals, with taking into account charging current due to an internal node. We also propose a signal assignment method to input terminals for reducing power consumption reduction by extending our method for large circuits, and demonstrate the effect of power consumption reduction by the present method.

B-ISDN telecommunication systems will require signal processing speeds up to 600 Mbps or more. We must therefore consider the affects of signal reflection, signal attenuation, time dalay, and so on when designing these systems. The higher the signal speed, the larger the electrical noise induced around the connector, especially in the plated through holes (PTHs) area. This paper presents the results of our investigation focused on connector mounting configurations in the signal transmission line, especially whether or not signals transmit through the PTH in a printed circuit board (PCB). How the signal reflection characteristics depend upon transmission line configurations are discussed and experimental results and simulation analyses for a transmission line system using a small miniature A-type (SMA) connector as an example are performed. It is suggested that designs for future high-speed signal transmission circuits take into account the PTH diameter and/or the PTH pitch conditions, values for which can be determined from simulation analysis.

Connection Admission Control (CAC) is a key part of traffic control and still leaves several challenging problems peculiar to ATM networks. One of these problems is how to assign sufficient bandwidth for any cell arrival process that satisfies the source traffic descriptor values specified by negotiation between the network and a user at the connection setup. Because the source traffic descriptor cannot describe the actual source traffic characteristics completely, it has already been studied extensively that how to estimate sufficient bandwidth under the assumption that the actual traffic parameter values in the source traffic descriptor are equal to the negotiated values. This paper extends the studies in the literature to how to estimate sufficient bandwidth only assuming that the actual values satisfy the negotiated values, that is the actual values is less than or equal to the negotiated values. We show the sufficient condition for negotiated source traffic descriptors ensuring that the cell-loss ratio calculated from the negotiated values is always the upper-bound of the actual cell-loss ratio. Using this condition, we propose a CAC that can guarantee cell-loss ratio objective so far as a user satisfies the source traffic descriptor values.

This paper proposes a universal structure for STM-N(N=1, 2, 3, ) multiplex line terminals that only utilizes N chips CMOS LSIs for Section OverHead (SOH) processing. The uniquely configured LSIs are applicable to any STM-N line terminal equipment. Reasonable frame alignment performance attributes, such as the maximum average reframe time, false in-frame time, out-of-frame detection time, and misframe time, are calculated for the configuration. A prototype SOH processing LSI built on 0.8m BiCMOS technology successfully realizes the functions needed for multiplex section termination. The STM-64 frame is also demonstrated using the proposed circuit configuration and prototype LSIs.

A beam adaptive frame for finite-element beam-propagation analysis is proposed. The width of the frame can be adapted itself to either the guiding structure or the propagating beam in optical circuits, so the size of the computational window can be reduced.

The warp model of the extended Hough transform (EHT) has been proposed to design the explicit expression of the transform function of EHT. The warp model is a skewed parameter space (R(µ,ξ), φ(µ,ξ)) of the space (µ,ξ), which is homeomorphic to the original (ρ,θ) parameter space. We note that the introduction of the skewness of the parameter space defines the angular and positional sensitivity characteristics required in the detection of lines from the pattern space. With the intent of contributing some solutions to basic computer vision problems, we present theoretically a dynamic and centralfine/peripheral-coarse camera vision architecture by means of this warp model of Hough transform. We call this camera vision architecture askant vision' from an analogy to the human askant glance. In this paper, an outline of the EHT is briefly shown by giving three functional conditions to ensure the homeomorphic relation between (µ,ξ) and (ρ,θ) parameter spaces. After an interpretation of the warp model is presented, a procedure to provide the transform function and a central-coarse/peripheralfine Hough transform function are introduced. Then in order to realize a dynamic control mechanism, it is proposed that shifting of the origin of the pattern space leads to sinusoidal modification of the Hough parameter space.

Synchronization has been one of the problems in M–ary spread spectrum communication systems. In this letter, we propose the frame synchronization method using the Hadamard matrix and a frame synchronization method of PCM communication systems. Moreover, we analyze the probabilities of keeping synchronous state and frame renewal rates, and we evaluate the relationship between these probabilities and the number of stages of counters.

This paper presents a modified disparity measurement to recover the depth and a robust method to estimate motion parameters. First, this paper considers phase correspondence for the computation of disparity. It has less computation for disparity than previous methods that use the disparity from correspondence and from correlation. This modified disparity measurement uses the Gabor filter to analyze the local phase property and the exponential filter to analyze the global phase property. These two phases are added to make quasi-linear phases of the stereo image channels which are used for the stereo disparity finding and the structure recovery of scene. Then, we use feature-based correspondence to find the corresponding feature points in temporal image pair. Finally, we combine the depth map and use disparity motion stereo to estimate 3-D motion parameters.