In this paper, we investigated the effect of chromaticity and luminance of surround to decide subject neutral white, and conducted a mathematical model of adapting degree for environment. Factors for adapting degree consist of two parts, adapting degree of ambient chromaticity and color saturation. These can be applied to color appearance models (CAM), actually improve the performance of color matching of CAM, hence would produce the method of image reproduction to general display systems.

A robust multichannel sound reproduction system that utilizes the relationship between the width of the actual control area and the control frequency of the control points is proposed. The reproduction accuracy of a conventional sound reproduction system is reduced by room environment variations when fixed inverse filter coefficients are used. This tendency becomes more significant when control points are arranged more closely. To resolve this problem, the frequency control band at every control point is switched to avoid degrading the reproduced sound in low frequencies, so the pass band range of the control points at both ears is only high-range. That of the other control points is the entire control range. Numerical simulation with real environmental data showed that improvement of the reproduction accuracy is about 6.1 dB on average, even with a temperature fluctuation of 5C as an environmental variation in the listening room.

Auditory artifacts due to switching head-related transfer functions (HRTFs) are investigated, using a software-implemented dynamic virtual auditory display (DVAD) developed by the authors. The DVAD responds to a listener's head rotation using a head-tracking device and switching HRTFs to present a highly realistic 3D virtual auditory space to the listener. The DVAD operates on Windows XP and does not require high-performance computers. A total system latency (TSL), which is the delay between head motion and the corresponding change of the ear input signal, is a significant factor of DVADs. The measured TSL of our DVAD is about 50 ms, which is sufficient for practical applications and localization experiments. Another matter of concern is the auditory artifact in DVADs caused by switching HRTFs. Switching HRTFs gives rise to wave discontinuity of synthesized binaural signals, which can be perceived as click noises that degrade the quality of presented sound image. A subjective test and excitation patterns (EPNs) analysis using an auditory filter are performed with various source signals and HRTF spatial resolutions. The results of the subjective test reveal that click noise perception depends on the source signal and the HRTF spatial resolution. Furthermore, EPN analysis reveals that switching HRTFs significantly distorts the EPNs at the off signal frequencies. Such distortions, however, are masked perceptually by broad-bandwidth source signals, whereas they are not masked by narrow-bandwidth source signals, thereby making the click noise more detectable. A higher HRTF spatial resolution leads to smaller distortions. But, depending on the source signal, perceivable click noises still remain even with 0.5-degree spatial resolution, which is less than minimum audible angle (1 degree in front).

The TFT-LCD image has non-uniform brightness that is the major difficulty of finding the visible defect called Mura in the field. To facilitate Mura detection, background signal shading should level off and Mura signal must be amplified. In this paper, Mura signal amplification and background signal flattening method is proposed based on human visual system (HVS). The proposed DC normalized contrast sensitivity function (CSF) is used for the Mura signal amplification and polynomial regression (PR) is used to level off the background signal. In the enhanced image, tri-modal thresholding segmentation technique is used for finding Dark and White Mura at the same time. To select reliable defect, falsely detected invisible region is eliminated based on Weber's Law. By the experimental results of artificially generated 1-d signal and TFT-LCD image, proposed algorithm has novel enhancement results and can be applied to real automated inspection system.

Although it is very important to conduct listening tests when constructing a practical sound field reproduction system based on wave field synthesis, listening tests are very expensive. A localization model of synthesized sound images that predicts the results of listening tests is proposed. This model reduces the costs of constructing a reproduction system because it makes it possible to omit the listening tests. The proposed model uses the precedence effect and predicts the direction of synthesized sound images based on the inter-aural time difference. A comparison of the results predicted by the proposed model and the localized results of listening tests shows that the model accurately predicts the localized results.

Speech dereverberation is one of the most difficult tasks in acoustic signal processing. Of the various problems involved in this task, this paper highlights "over-whitening," which flattens the characteristics of recovered speech. This distortion sometimes happens when inverse filters are directly calculated from microphone signals. This paper reviews two studies related to this problem. The first study shows the possibility of compensating for such over-whitening to achieve precise speech-dereverberation. The second study presents a new approach for approximating the original speech by removing the effect of late reflections from observed reverberant speech.

In this paper the transfer function of a system with windowed current integration is discussed. This kind of integration is usually used in a sampling mixer and the current is generated by a transconductance amplifier (TA). The parasitic capacitance (Cp) and the output resistance of the TA (Ro,TA) before the sampling mixer heavily affect the performance. Calculations based on a model including the parasitic capacitance and the output resistance of the TA is carried out. Calculation results show that due to the parasitic capacitance, a notch at the sampling frequency appears, which is very harmful because it causes the gain near the sampling frequency to decrease greatly. The output resistance of the TA makes the depth of the notches shallow and decreases the gain near the sampling frequency. To suppress the effect of Cp and Ro,TA, an operational amplifier is introduced in parallel with the sampling capacitance (Cs). Simulation results show that there is a 17 dB gain increase while Cs is 1,pF, gm is 9,mS, N is 8 with a clock rate of 800,MHz.

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.

Many studies of on-line comprehension of semantic violations have shown that the human sentence processor rapidly constructs a higher-order semantic interpretation of the sentence. What remains unclear, however, is the amount of time required to detect semantic anomalies while concatenating two words to form a phrase with very rapid stimuli presentation. We aimed to examine the time course of semantic integration in concatenating two words in phrase structure building, using magnetoencephalography (MEG). In the MEG experiment, subjects decided whether two words (a classifier and its corresponding noun), presented each for 66 ms, form a semantically correct noun phrase. Half of the stimuli were matched pairs of classifiers and nouns. The other half were mismatched pairs of classifiers and nouns. In the analysis of MEG data, there were three primary peaks found at approximately 25 ms (M1), 170 ms (M2) and 250 ms (M3) after the presentation of the target words. As a result, only the M3 latencies were significantly affected by the stimulus conditions. Thus, the present results indicate that the semantic integration in concatenating two words starts from approximately 250 ms.

This paper proposes a new control scheme in which the base station (BS) controls terminal's transmit beamforming in time-division duplex (TDD)/multi-input multi-output (MIMO) systems. In the proposed scheme, the BS transmits pilot signals using appropriate downlink beams to instruct a terminal on target transmit beamforming. Using responses of the downlink pilot signals, the terminal can perform transmit beamforming close to the target one. Our theoretical investigation reveals that the BS can control multiple terminals' transmit beamforming simultaneously. Furthermore, an efficient signal processing at the terminal is investigated to obtain precise weight of transmit beamforming in noise environments. Numerical results show that the terminal can perform precise transmit beamforming close to the target one in noise environments. It is also shown that the amount of downlink control signalling in the proposed scheme is much less than that in codebook-based approach.

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.

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 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.

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.

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.

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.

Approximate pattern matching plays an important role in various applications. In this paper we focus on (δ, γ)-matching, where a character can differ at most δ and the sum of these errors is smaller than γ. We show how to find these matches when the pattern is transformed by y=αx + β, without knowing α and β in advance.

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.

This paper discusses the design of configurations of videophone equipment aimed at online sign interpretation. We classified interpretation services into three types of situations: on-site interpretation, partial online interpretation, and full online interpretation. For each situation, the spatial configurations of the equipment are considered keeping the issue of nonverbal signals in mind. Simulation experiments of sign interpretation were performed using these spatial configurations and the qualities of the configurations were assessed. The preferred configurations had the common characteristics that the hearing subject could see the face of his/her principal conversation partner, that is, the deaf subject. The results imply that hearing people who do not understand sign language utilize nonverbal signals for facilitating interpreter-mediated conversation.