A three-phase complex filter for a balanced three-phase analog signal processing is proposed. The proposed three-phase active-RC Tow-Thomas biquad complex filter can reduce total resistance by 10 percent, total capacitance by 25 percent, and power consumption by 22 percent compared to a conventional fully differential quadrature complex one.

A deterministic pushdown automaton (dpda) having just one stack symbol is called a deterministic restricted one-counter automaton (droca). When it accepts an input by empty stack, it is called strict. This paper is concerned with a subclass of real-time strict droca's, called Szilard strict droca's, and studies the problem of identifying the subclass in the limit from positive data. The class of languages accepted by Szilard strict droca's coincides with the class of Szilard languages (or, associated languages) of strict droca's and is incomparable to each of the class of regular languages and that of simple languages. After providing some properties of languages accepted by Szilard strict droca's, we show that the class of Szilard strict droca's is polynomial time identifiable in the limit from positive data in the sense of Yokomori. This identifiability is proved by giving an exact characteristic sample of polynomial size for a language accepted by a Szilard strict droca. The class of very simple languages, which is a proper subclass of simple languages, is also proved to be polynomial time identifiable in the limit from positive data by Yokomori, but it is yet unknown whether there exists a characteristic sample of polynomial size for any very simple language.

This paper presents a very wideband active RC polyphase filter (ARCPF). We propose a unit section of the ARCPF, which is an ordinary RCPF followed by opamps with parallel RC feedback. In the proposed unit section, pole and zero can be assigned independently. By using the unit ARCPFs, a very wideband image rejection filter can be realized by cascading the sections, which can greatly reduce the element value spread. To realize this, CMOS inverter based fully differential OTA which can operate under low supply voltage is also presented. This paper describes a six-stage active RC polyphase filter with 1-100 MHz passband in 0.18 µm CMOS technology.

A framework for generating facial expressions from emotional states in daily conversation is described. It provides a mapping between emotional states and facial expressions, where the former is represented by vectors with psychologically-defined abstract dimensions, and the latter is coded by the Facial Action Coding System. In order to obtain the mapping, parallel data with rated emotional states and facial expressions were collected for utterances of a female speaker, and a neural network was trained with the data. The effectiveness of proposed method is verified by a subjective evaluation test. As the result, the Mean Opinion Score with respect to the suitability of generated facial expression was 3.86 for the speaker, which was close to that of hand-made facial expressions.

This paper proposes an antenna selection method for terminal antennas employing orthogonal polarizations and patterns, which is suitable for outdoor MultiUser Multi-Input Multi-Output (MU-MIMO) systems. In addition, this paper introduces and verifies two other antenna selection methods for comparison. For the sake of simplicity, three orthogonal dipoles are considered, and this antenna configuration using the proposed selection method is compared to an antenna configuration with three vertical or horizontal dipoles. In the proposed antenna selection method, we always choose the vertical dipole, and choose one of two horizontal dipoles, which are orthogonal to each other, based on the Signal-to-Noise Ratio (SNR). We measured the MU-MIMO transmission properties and found that the proposed selection method employing the antenna with orthogonal polarizations and patterns can offer fairly high channel capacity in a multiuser scenario.

A compact built-in handset antenna for multiple-input multiple-output (MIMO) system at 2 GHz, comprising two elements array of newly proposed L-shaped folded monopole antenna (LFMA), is evaluated under the multipath radio wave propagation environments. By analyzing the fundamental characteristics, mean effective gain (MEG), correlation, and channel capacity, the significant enhancement in the capability, as a handset MIMO antenna under practical use conditions, was confirmed. The performances were also compared to those of an array antenna comprising two planar inversed-F antenna (PIFA) elements in order to verify the effectiveness of the proposed antenna. The results show that the equivalent or improved performances can be realized, by using the proposed LFMA array with a compact size, taking only the volume of 44% of a PIFA array. The LFMA array provides almost the same bandwidth and enhanced isolation compared with a PIFA array, and the sufficiently low correlation and acceptable effective gain are obtained under the multipath radio wave propagation environments. In addition, a greater channel capacity than a PIFA array is achieved especially when the proposed LFMA array is inclined for the display-viewing mode, and moreover, an almost doubled increase in the channel capacity is obtained by using MIMO transmission compared with single-input single-output (SISO). This study also show that the MEG has much effects on the channel capacity, rather than the correlations, for the proposed antenna.

While significant work has been dedicated to analyzing the performance of antennas for multiple-input multiple-output (MIMO) and diversity systems, little has appeared on synthesizing optimal antennas for these systems. This paper explores optimal antenna characteristics given understanding about the average structure of the propagation environment, making the results applicable for time-varying channels created by mobile nodes or scatterer motion. Specifically, it examines optimal antenna designs for the cases where 1) the antennas reside in a fixed aperture or 2) the number of antenna elements is fixed (under fast-fading conditions with spatially correlated signals).

This paper illustrates a feasible health informatics domain knowledge management process which helps gather useful technology information and reduce many knowledge misunderstandings among engineers who have participated in the IBM mainframe rightsizing project at National Taiwan University (NTU) Hospital. We design an asynchronously sharing mechanism to facilitate the knowledge transfer and our health informatics domain knowledge management process can be used to publish and retrieve documents dynamically. It effectively creates an acceptable discussion environment and even lessens the traditional meeting burden among development engineers. An overall description on the current software development status is presented. Then, the knowledge management implementation of health information systems is proposed.

In this paper, we present a new three-dimensional (3D) virtual environment (3DVE) system named "QuViE/P", which can enhance quality of service (QoS), that users actually feel, as good as possible when resources of computers and networks are limited. To realize this, we focus on characteristics of user's perceptual quality evaluation on 3D objects. We propose an effective QoS control scheme for QuViE/P by introducing relationships between system's internal quality parameters and user's perceptual quality parameters. This scheme can appropriately maintain the QoS of the 3DVE system and it is expected to improve convenience when using 3DVE system where resources are insufficient. We designed and implemented a prototype of QuViE/P using a multiagent framework. The experiment results show that even when the computer resource is reduced to 20% of the required amount, the proposed scheme can maintain the quality of important objects to a certain level.

We propose a new robust frequency domain acoustic echo cancellation filter that employs a normalized residual echo enhancement. By interpreting the conventional robust step-size control approaches as a statistical-model-based residual echo enhancement problem, the optimal step-size introduced in the most of conventional approaches is regarded as optimal only on the assumption that both the residual echo and the outlier in the error output signal are described by Gaussian distributions. However, the Gaussian-Gaussian mixture assumption does not always hold well, especially when both the residual echo and the outlier are speech signals (known as a double-talk situation). The proposed filtering scheme is based on the Gaussian-Laplacian mixture assumption for the signals normalized by the reference input signal amplitude. By comparing the performances of the proposed and conventional approaches through the simulations, we show that the Gaussian-Laplacian mixture assumption for the normalized signals can provide a better control scheme for the acoustic echo cancellation.

This paper proposes a combination of adaptive equalizer and Least Mean Square-RAKE (LMS-RAKE) combining scheme receiver system for Direct Sequence-Ultra Wideband (DS-UWB) multipath channel model. The main purpose of the proposed system is to overcome the performance degradation for UWB transmission due to the occurrence of Inter-Symbol Interference (ISI) during high speed transmission of ultra short pulses in a multipath channel. The proposed system improves the system performance by mitigating the multipath effect using LMS-RAKE receiver and suppressing the ISI effect with the adaptive equalizer. Simulation results verify that significant equalization gain can be obtained by the proposed system especially in UWB multipath channel models such as channel CM3 and channel CM4 that suffered severe ISI effect.

This paper proposes a low power and high speed track and hold circuit (T/H circuit) based on the two-stage structure. The proposed circuit consists of two internal T/H circuits connected in cascade. The first T/H circuit converts an input signal into a step voltage and it is applied to the following second T/H circuit which drives large load capacitors and consumes large power. Applying the step voltage to the second T/H circuit prevents the second T/H circuit from charging and discharging its load capacitor during an identical track phase and enables low power operation. Thanks to the two-stage structure the proposed T/H circuit can save 29% of the power consumption compared with the conventional one. An optimum design procedure of the proposed two stage T/H circuit is explained and its validity is confirmed by HSPICE simulations.

This paper presents a new method for reconstruction of trigonometric polynomials, a specific class of bandlimited signals, from a number of integrated values of input signals. It is applied in signal reconstruction, spectral estimation, system identification, as well as in other important signal processing problems. The proposed method of processing can be used for precise rms measurements of periodic signal (or power and energy) based on the presented signal reconstruction. Based on the value of the integral of the original input (analogue) signal, with a known frequency spectrum but unknown amplitudes and phases, a reconstruction of its basic parameters is done by the means of derived analytical and summarized expressions. Subsequent calculation of all relevant indicators related to the monitoring and processing of ac voltage and current signals is provided in this manner. Computer simulation demonstrating the precision of these algorithms. We investigate the errors related to the signal reconstruction, and provide an error bound around the reconstructed time domain waveform.

In this letter, we propose effective feature vectors to improve the performance of voice activity detection (VAD) employing a support vector machine (SVM), which is known to incorporate an optimized nonlinear decision over two different classes. To extract the effective feature vectors, we present a novel scheme that combines the a posteriori SNR, a priori SNR, and predicted SNR, widely adopted in conventional statistical model-based VAD.

To improve the computation efficiency of the application specific instruction-set processor (ASIP), a strategy of hardware/software collaborative design is usually utilized. In this process, the auto-customization of specific instruction set has always been a key part to support the automated design of ASIP. The key issue of this problem is how to effectively reduce the huge exponential exploration space in the instruction identification process. To address this issue, we first formulate it as a feasible sub-graph enumeration problem under multiple constraints, and then propose a fast instruction identification algorithm based on a new model called basic convex pattern (BCP). The kernel technique in this algorithm is the transformation from the graph exploration to the formula-based computations. The experimental results have indicated that the proposed algorithm has a distinct reduction in the execution time.

In this paper, we propose a computationally efficient method of compensating temperature for the transaural stereo. The conventional method can be used to estimate the change in impulse responses caused by the fluctuation of temperature with high accuracy. However, the large amount of computation required makes real-time implementation difficult. Focusing on the fact that the amount of compensation depends on the length of the impulse response, we reduce the computation required by segmenting the impulse response. We segment the impulse responses in the time domain and estimate the effect of temperature fluctuation for each of the segments. By joining the processed segments, we obtain the compensated impulse response of the whole length. Experimental results show that the proposed method can reduce the computation required by a factor of nine without degradation of the accuracy.

In this study, we conducted a series of experiments using stimuli characterized by various attributes in order to understand the categorization process in an infant's pre-linguistic development. The infants are able to assign the same label to members within the same category by focusing attention on specific features or functions common to the members. The ability to categorize is likely to play an essential role in an infant's overall cognitive development. Specifically, we investigated how the infants use different strategies in the process of linguistic categorization. In one strategy, members of a single category are derived from perceptual similarities within the most representative members, i.e., the prototypical members. Alternatively, each membership is established by referring to the linguistic labels for each category provided by the caretaker, in a symbol grounding process. We found that the infant is able to employ these strategies in a flexible manner in its development. We discuss the interplay between different cognitive strategies, including the prototype effects in the infant's cognitive development and the implications for cortical mechanism involved.

In this letter, we develop a two-step receive antenna selection method to maximize channel capacity. Different from previous work, we first derive a lower bound on capacity based on Hadamard inequality and arithmetic-geometric mean inequality, which is then used to iteratively drop the worst-performing antennas according to their measure. The recursive nature of this method helps to largely reduce the computational complexity.

We study in this paper the subchannel access and the rate assignment for the multicarrier multi-cell networks. For subchannel access, we show from theoretic results and simulation results that the scheme with only one user per cell in each subchannel outperforms the scheme with multiple users per cell in each subchannel. For rate assignment, a distributed rate assignment is proposed to assign the rate for all subchannels. The proposed rate assignment need not measure the channel gains and uses only local information to iteratively adjust the transmitting power and data rate. We prove that the aggregate rate can be increased by increasing the number of iterations in the proposed rate assignment.

We present an interactive system for cosmetic makeup of a point-based face model acquired by 3D scanners. We first enhance the texture of a face model in 3D space using low-pass Gaussian filtering, median filtering, and histogram equalization. The user is provided with a stereoscopic display and haptic feedback, and can perform simulated makeup tasks including the application of foundation, color makeup, and lip gloss. Fast rendering is achieved by processing surfels using the GPU, and we use a BSP tree data structure and a dynamic local refinement of the facial surface to provide interactive haptics. We have implemented a prototype system and evaluated its performance.