Showing papers in "Electronics and Communications in Japan Part Iii-fundamental Electronic Science in 1999"
TL;DR: A method is described for updating the background image corresponding to the environment changing the illumination conditions significantly and for detecting obstructions and tracking of moving vehicles on the road and a high accuracy of 93% was accomplished for the tracking ofMoving objects in the case of daytime images.
Abstract: In this paper, a method is described for updating the background image corresponding to the environment changing the illumination conditions significantly and for detecting obstructions and tracking of moving vehicles on the road. In the proposed method, the varying region in the monitoring image is derived from the background difference and time difference and is classified into (a) moving objects, (b) stationary objects, and (c) the change due to illumination variation. Based on this classification, region (c) and the region without any change are used for background updating. Also, by forming the a cumulative image in which the binary images composed of regions (a) and (b) are stored for a specified length of time, the detection of the stationary objects is performed and the traffic conditions are determined. For tracking of moving objects, first the DCT is used so that the images consisting only of the moving object and excluding shadow are extracted. Next, these images are used as the template and are tracked by matching. The present method was applied to three types of video images. It was found that all of the 10 stationary vehicles contained in the image were detected. Also, a high accuracy of 93% was accomplished for the tracking of moving objects in the case of daytime images. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(11): 28–37, 1999
31 citations
TL;DR: A new recognition technique is proposed, which uses the 2D DCT of the entire facial image and a neural network to realize a mapping into the facial expression space.
Abstract: The recognition of human facial expressions has been expected to have applications in various fields, such as psychology, engineering, and so forth, and many techniques have been proposed to date. However, most of the techniques are based on the motions of local feature-bearing blocks, such as the eyes and mouth, which are supposed to be closely associated with facial expressions. It is thus required to segment these blocks from the face image and to track their motions in real-time applications, resulting in high complexity. Furthermore, not all information on the motions of the facial expressions has been utilized. In this paper, a new recognition technique is proposed, which uses the 2D DCT of the entire facial image and a neural network. The segmentation of local feature-bearing blocks is not needed. The differences of the DCT coefficients in the lower frequency area between neutral and expression-bearing images are given to a neural network to realize a mapping into the facial expression space. The new technique has been applied to a database of normalized facial images of 60 persons, with images of 40 persons used for network training and the remaining images for testing. The recognition rate is 100% for the training images. A maximum recognition rate of 95% is achieved for the testing images. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(7): 1–11, 1999
26 citations
TL;DR: This paper investigates and defines how dial-a-ride service will optimize many customers' requests with a short response time and applies a genetic algorithm to demonstrate the optimization of their operations and services.
Abstract: In the last decade we have experienced fast-growing traffic congestion and spreading built-up urban areas without adequate transportation systems. We are searching for a new public transit system with dial-a-ride service to maintain our quality of life. Our concept for a public transit management system has three stages of service; the dial-a-ride services operate at the lowest, local-level services, as public taxi or jitney services. This paper will investigate and define how dial-a-ride service will optimize many customers' requests with a short response time. Our previous study showed how one dial-a-ride service vehicle could operate in a service zone. However, realistically there are many more customers in a service zone than one dial-a-ride service vehicle can handle. Therefore, in this paper, we study how multiple dial-a-ride service vehicles can operate in one service zone and apply a genetic algorithm to demonstrate the optimization of their operations and services. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(7): 30–38, 1999
23 citations
TL;DR: In this paper, a noncontact ultrasonic motor with axial acoustic viscous forces and driven in a circumferential direction was proposed and constructed for trial, and the rotor of this motor was analyzed theoretically and experimentally so as to provide guidelines for increasing the levitation force.
Abstract: Noncontact ultrasonic motors, whose rotor is levitated by acoustic pressure and driven by acoustic streaming, have recently been proposed and developed. In this paper, a novel noncontact ultrasonic motor is proposed and constructed for trial. The rotor of this motor is levitated in the direction parallel to the vibrating surface of the stator by axial acoustic viscous forces, and driven in the circumferential direction by circumferential acoustic viscous forces. A higher revolution speed and more stable revolution than other noncontact ultrasonic motors are obtained. Furthermore, the effects of the stator vibration velocity, the vibration distribution, the mode number, the gap width, the driving frequency, and the rotor surface roughness on the levitation force are analyzed theoretically and experimentally so as to provide guidelines for increasing the levitation force. ©1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(4): 56–63, 1999
22 citations
TL;DR: A new description of the block-sorting data compression algorithm in terms of the KarpnMillernRosenberg (KMR) repetition finder is given in order to combine the combinatorial aspect and the reversibility of this reversibility.
Abstract: SUMMARY The block-sorting data compression method of Burrows and Wheeler has received considerable attention in anticipation that it may be comparable, or even superior, to the ZivnLempel codes. This article discusses its characteristic points from the viewpoint of string algorithms. The block-sorting compression algorithm initially sorts all rotations of an input text lexicographically. This transformation is still reversible when we restrict the key length of sorting. We first focus on the combinatorial aspect of this reversibility, then show that the block-sorting algorithm has the capability of finding repetitions in an original string. We give a new description of the algorithm in terms of the KarpnMillernRosenberg (KMR) repetition finder in order to combine the above two aspects. ' 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(6): 18n25, 1999
14 citations
TL;DR: In this paper, the authors describe the unified relationship among waveform distortion, acoustic streaming, and radiation pressure, and describe the effect of flow generated around the object on the radiation pressure when the object is smaller.
Abstract: The generation mechanism of acoustic streaming and radiation pressure are discussed along with the KZK equation, which explains successfully the propagation of finite-amplitude sound waves. The energy loss of a sound beam in a viscous fluid generates the driving force of streaming which induces mass flow in the beam. The radiation pressure acts on the surface of an object whose acoustic impedance is different from the fluid. The effect of flow generated around the object on the radiation pressure is greater when the object is smaller. The main purpose of this paper is to describe the unified relationships among waveform distortion, acoustic streaming, and radiation pressure. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(2): 76–82, 1999
10 citations
TL;DR: In the objective test, the proposed speech morphing algorithm can reduce the spectral distortion by 1.9 dB in comparison with the method based on progressive substitution of spectra, when it is used for interpolating two vowels in a real speech.
Abstract: A speech morphing algorithm based on progressive interpolation of spectral envelopes and source signals is proposed. The basic morphing scheme is; 1) determine the time correspondence for unit waveforms of original and target speech, 2) separate speech spectra into an envelope and a sound source, 3) obtain the frequency correspondence for spectral channels of original and target speech for each envelope, 4) interpolate both source signal and envelope, 5) construct a unit waveform, and 6) generate morphing speech by PSOLA. In the objective test, the proposed method can reduce the spectral distortion by 1.9 dB in comparison with the method based on progressive substitution of spectra, when it is used for interpolating two vowels in a real speech. The effectiveness of the method is also confirmed by a subjective test in which 89% (male to female) or 93% (female to male) subjects preferred the proposed method. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(3): 22–30, 1999
10 citations
TL;DR: In this article, a time series is represented by a convolution of the input signal and the impulse response function, expanded by using a set of scaling functions, and a prediction method is derived by using the fact that the impulse reaction retains self-similarity even if the time scale is expanded, due to the fractal geometry of the time series.
Abstract: This paper deals with the prediction of time series that have fractal geometry. We also describe the prediction error, the estimation of fractal dimension of the times series, and several applications. First, we assume that the time series is represented by a convolution of the input signal and the impulse response function, expanded by using a set of scaling functions. Then, a prediction method is derived by using the fact that the impulse response retains self-similarity even if the time scale is expanded, due to the fractal geometry of the time series. We demonstrate the ability of the prediction method for various fractal dimensions by showing that the one-step-ahead prediction error is very small, and also that the n-step-ahead prediction error can be made small by using adaptive correction of the data in which the prediction is successively used as an observation. We also describe the estimation of the fractal dimension by using the covariance matrix and the shape of the spectrum of the time series. As an application, the option price of the stock index estimated by the method is compared to the price obtained by conventional estimation, and our strategy is shown to give a greater profit. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(3): 31–39, 1999
9 citations
TL;DR: In this article, a pre-sented formula for recovering the original signal from its irregularly sampled values using wavelets, which extendsthe Walter sampling theorem to the irregular sampling case and generalizes the Paley-Wiener 1/4-Theorem by remov-ing the symmetricity constraint for sampling, is pre-specified.
Abstract: SUMMARY A formula for recovering the original signal from itsirregularly sampled values using wavelets, which extendsthe Walter sampling theorem to the irregular sampling caseand generalizes the PaleyŒWiener 1/4-Theorem by remov-ing the symmetricity constraint for sampling, is pre-sented.a 1999 Scripta Technica, Electron Comm Jpn Pt 3,82(5): 65Œ71, 1999Key words: Wavelet; sampling theorem; scalingfunction; orthogonality; biorthogonality. 1. Introduction In digital signal and image processing, digital com-munications, and so forth, a continuous signal is usuallyrepresented and processed by using its discrete samples.How, then, are we to reconstruct the original signal from itsdiscrete samples? The classical Shannon sampling theoremgives the following formula for band-limited finite energysignals.For a finite energy s-band continuous signalf( t), 2t ˛ R , that is, supp f^(w) I [- s , s] and f ˛ L (R), it canbe recovered by the formulawhere f^ is the Fourier transform of f(t) defined byIf we let s =
7 citations
TL;DR: In this article, the maximum output power of a piezoelectric transformer is expressed as the product of the v2max, the square of the power coefficient on the output side A2 and the load resistance RL.
Abstract: Considering the characteristics of materials required for piezoelectric transformers, and introducing the maximum vibration velocity vmax that generates a fixed temperature rise, we can express the maximum output power of a piezoelectric transformer as the product of this v2max, the square of the power coefficient on the output side A2 and the load resistance RL. Moreover, the power coefficient on the output side A2 is proportional to the product of the square root of Young's modulus YE, the square root of the electromechanical coupling coefficient k, and the permittivity ϵ33. This is verified by using two types of piezoelectric transformers in an inverter where the output section may be of the Rosen type or the transverse-effect type. It is shown that the contributions made by the structure of the piezoelectric transformer and the load resistance differ in each case. The Rosen structure is used in the piezoelectric transformer for the inverter, and it is shown that a material with large maximum vibration velocity vmax is useful in this case. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(11): 86–92, 1999
5 citations
TL;DR: In this article, a rotary linear motor with a pair of two-dimensional ultrasonic actuators is presented, which can rotate and move directly, simultaneously, or independently in two orthogonal directions independently.
Abstract: There is a need to draw various patterns on the inner wall of a small pipe several millimeters in diameter to develop a precise linear scale. Since the authors have already developed a two-dimensional ultrasonic actuator that can move an object is two orthogonal directions independently, they now offer a rotary linear motor, in which the motor shaft is driven by a pair of two-dimensional ultrasonic actuators. Hence, the motor can rotate and move directly, simultaneously, or independently.
Good positioning accuracy was obtained for simultaneous two directional drive by controlling the motor with a proportional control: the steady errors were 1.41° and 0.73 mm in the rotary and linear directions, respectively. However, the positioning accuracy greatly decreased for one-directional drive: the steady errors were 27.5° for rotational motion and 20.3 mm for linear motion. To improve these poor positioning accuracies for the one-directional drive, RC series circuits were connected parallel to the piezoelectric devices of a two-dimensional ultrasonic actuator in order to reduce the interference due to the self-EMF induced by the ultrasonic vibration of one pair of piezoelectric devices on another pair. In addition, proportional control was replaced with proportional and integral control. As the result, the positioning accuracy for one-directional drive has been greatly improved, to 0.05° for rotational motion only and 0.006 mm for linear motion only. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(2): 48–57, 1999
TL;DR: In this article, a block length control is proposed to improve the convergence of the estimation precision, as in step gain control, and a method of reducing the convergence time is proposed, utilizing the fact that saturation of the convergence behavior can be observed iteratively by extending the block.
Abstract: It is known in the normalized least mean square (NLMS) algorithm that the convergence time can be reduced by applying a control in which the step gain is gradually reduced, as the estimation precision is improved. A problem in realizing this idea is that the instantaneous value of the estimation precision which is needed in the control of the step gain cannot be observed directly from the residual signal which contains external disturbances. This paper considers a block length control, which can be used in the “summational” NLMS method. The method is able to improve the convergence of the estimation precision, as in step gain control, and a method of reducing the convergence time is proposed, utilizing the fact that saturation of the convergence behavior can be observed iteratively by extending the block. In addition, it is highly likely in an actual adaptive system that the reference signal, the power of the external disturbance, and the impulse response of the unknown system to be estimated as the coefficients of the finite impulse response (FIR) filter will change. It is not realistic to ignore the possibility of these changes in discussing the control of convergence. In the last part of this paper, it is shown that the proposed control method can handle these changes. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(12): 54–64, 1999
TL;DR: The transformed facial images were successfully judged in terms of the degree of transformation (cardioidal strain degree) and from the subjects' estimation of actual age, the effect of the transformation degree differed between female and male facial images.
Abstract: In this study, we transformed the real front view of facial images by cardioidal strain change. Subjects then made relative and actual age estimations of the facial images. The results were as follows. First, the transformed facial images were successfully judged in terms of the degree of transformation (cardioidal strain degree). Second, from the subjects' estimation of actual age, the effect of the transformation degree differed between female and male facial images. The effect was weaker for the male facial images than for the female. Additionally, for the male facial images the effect of the transformation differed according to face type; there was a significant effect for childlike faces. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(5): 39–48, 1999
TL;DR: A stepped-FM coded radar that is free from radar mutual interference, has high resolution, and is referred to as FM-CW radar is described.
Abstract: This paper focuses on radar that is used in automotive applications. It describes a stepped-FM coded radar that is free from radar mutual interference, has high resolution, and is referred to as FM-CW radar. It also describes a ranging method based on one radar profile
TL;DR: In this article, a new RLS criterion that includes a strong parameter energy factor was proposed, and experiments show that adaptive algorithms that meet this criterion are numerically more stable than conventional RLS algorithms.
Abstract: One reason that recursive least-squares (RLS) algorithms are not used as widely as LMS algorithms is an instability problem resulting from finite precision errors in the calculations; leaky LMS algorithms that are numerically stable have been proposed in the LMS literature. In this paper, we offer a new RLS criterion that includes a strong parameter energy factor. Experiments show that adaptive algorithms that meet this criterion are numerically more stable than conventional RLS algorithms. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(4): 26–37, 1999
TL;DR: In this paper, the authors proposed an adaptive algorithm for an echo canceler, which is referred to as subband ES (exponentially weighted stepsize), where the frequency bands of the received input signal and echo signal are divided into multiple subbands, and echo is independently canceled in each subband.
Abstract: This paper proposes a novel adaptive algorithm for an echo canceler. In this algorithm, the number of operations and memory capacity are equivalent to those of the conventional NLMS algorithm but the convergence speed is twice that using the conventional algorithm. This adaptive algorithm is referred to as subband ES (exponentially weighted stepsize). In the algorithm, the frequency bands of the received input signal and echo signal are divided into multiple subbands, and echo is independently canceled in each subband. Each adaptive filter in each subband has independent coefficients with an independent stepsize. The stepsize is time-independent and its weight is exponentially proportional to the change of the impulse response within the frequency region, such as the expected value of the difference between the waveforms of two impulse responses. As a result, the characteristic of the acoustic echo path in each frequency band is analyzed using the adaptive algorithm to improve the convergence characteristic. Using the results of computer simulation and experimental results obtained via an experimental setup with DSP, it is shown that the convergence speed with respect to input voice signal can be about 4 times faster when using echo cancellation based on the new algorithm than in conventional full-band echo cancellation based on the NLMS algorithm. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(3): 49–57, 1999
TL;DR: How the angle of depression affects the driver's perception of HUD images and the forward view is described and why misperception occurs, what causes it, and how to design HUDs to avoid it are discussed, based on experimental results derived from an analysis of drivers' eye movements.
Abstract: Although HUDs (head-up displays) were originally developed for aircraft use, in the past decade they have found application in automobiles.
The angle of depression of the displayed image is one of the most important parameters for designing automotive HUDs. Generally, drivers can perceive more information displayed by HUDs when the angle of depression is smaller. However, it has been observed in experiments that decreasing the angle in an actual automotive HUD, where the distance to the displayed image is not great, can result in misperception of the displayed information.
This article describes how the angle of depression affects the driver's perception of HUD images and the forward view. It also discusses why misperception occurs, what causes it, and how to design HUDs to avoid it, based on experimental results derived from an analysis of drivers' eye movements. ©1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(4): 1–9, 1999
TL;DR: In this article, a Braille type input/output device was proposed for deaf-blind communication, in which the palm sides of the forefinger, the middle finger, and the ring finger near the proximal and distal finger joints are used for presenting Braille information.
Abstract: This paper proposes a Braille type input/output device as a communication aid for the deaf-blind. By means of this device, both input and output operations can be handled in 6-point braille format. The input/output unit of this device has a structure in which a vibration motor for the pager and a push-button switch are combined. Braille information can be input or presented using the same hand position. In presenting the Braille information, the palm sides of the forefinger, the middle finger, and the ring finger near the proximal and distal finger joints are used, to allow for ease during the input operation. In order to determine the optimal position for presenting Braille information in this device, an experiment was performed on single-point vibration stimulation sensitivity as well as two-point simultaneous vibration stimulation sensitivity. The experiment showed that the visually impaired subjects were able to recognize Braille information with a recognition rate of approximately 84%, after training for ten minutes. The input of Braille information was performed by a sighted subject, who could input the information at a rate of approximately 36 character/min. after training for five hours. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(1): 9–18, 1999