The hand posture estimation system by searching a similar image from a vast database, such as our previous research, may cause the increase of processing time, and prevent realtime controlling of a robot. In this study, the authors proposed a new estimation method of human hand posture by rearranging a large-scale database with the Self-Organizing Map including self-reproduction and self-annihilation, which enables two-step searches of similar image with short period of processing time, within small errors, and without deviation of search time. The experimental results showed that our system exhibited good performance with high accuracy within processing time above 50 fps for each image input with a 2.8 GHz CPU PC.

This paper describes the design of a small-offset 12-bit CMOS charge-redistribution DAC using a weighted-mean flip-around sample-and-hold circuit (S/H). Flip-around S/H topology can realize small-offset characteristics, and it is effective to reduce power dissipation and chip area because independent feedback capacitors are not necessary. In this DAC the small-offset characteristic remains not only in amplification phase but also in sampling phase with the circuit technique. The design of 1.8 V, 50 MS/s fully differential DAC with output swing of 2 Vp-p has very small offset of 100 µV for the reset switch mismatch of 2%. A technique to improve dynamic performance measured by SFDR using damping resistors and switches at the output stage is also presented. The designed 12-bit DAC with 0.25 µm CMOS technology has low-power dissipation of 35 mW at 50 MS/s.

In this paper, the confinement loss of octagonal photonic crystal fibers (PCFs) with an isosceles triangle lattice of air-holes are numerically investigated. Taking into account the confinement loss, the mode field diameter (MFD), the effective area (Aeff) and the chromatic dispersion of octagonal PCFs are calculated, compared to conventional hexagonal PCFs. It is found from confinement loss and MFD results that the octagonal PCFs can confine the field strongly than the hexagonal PCFs due to the different air filling fraction. Moreover, it is shown that the octagonal PCFs are obtained not only positive but also negative larger dispersion values and smaller Aeff values compared to the hexagonal PCFs.

This paper describes the Differential Pass Transistor Pulsed Latch (DPTPL) which enhances D-Q delay and reduce power consumption using NMOS pass transistors and feedback PMOS transistors. The proposed flip-flop uses the characteristic of stronger drivability of NMOS transistor than that of transmission gate if the sum of total transistor width is the same. Positive feedback PMOS transistors enhance the speed of the latch as well as guarantee the full-swing of internal nodes. Also, the power consumption of proposed pulsed latch is reduced significantly due to the reduced clock load and smaller total transistor width compared to conventional differential flip-flops. DPTPL reduces ED by 45.5% over ep-SFF. The simulations were performed in a 0.1 µm CMOS technology at 1.2 V supply voltage with 1.25 GHz clock frequency.

This paper presents a new statistical model-based voice activity detection (VAD) algorithm in the wavelet domain to improve the performance in non-stationary environments. Due to the efficient time-frequency localization and the multi-resolution characteristics of the wavelet representations, the wavelet transforms are quite suitable for processing non-stationary signals such as speech. To utilize the fact that the wavelet packet is very efficient approximation of discrete Fourier transform and has built-in de-noising capability, we first apply wavelet packet decomposition to effectively localize the energy in frequency space, use spectral subtraction, and employ matched filtering to enhance the SNR. Since the conventional wavelet-based spectral subtraction eliminates the low-power speech signal in onset and offset regions and generates musical noise, we derive an improved multi-band spectral subtraction. On the other hand, noticing that fixed threshold cannot follow fluctuations of time varying noise power and the inability to adapt to a time-varying environment severely limits the VAD performance, we propose a statistical model-based VAD algorithm in wavelet domain with an adaptive threshold. We perform extensive computer simulations and compare with the conventional algorithms to demonstrate performance improvement of the proposed algorithm under various noise environments.

In this paper, we propose a combination-based novel technique of dithered subband coding with spectral subtraction for improving the perceptual quality of coded audio at low bit rates. It is well known that signal-correlated distortion is audible when the audio signal is quantized at bit rates lower than the lower bound of perceptual coding. We show that this problem can be overcome by applying the dithering quantization process in each subband. Consequently, the quantization noise is rendered into a signal-independent white noise; this noise is then estimated and removed by spectral subtraction at the decoder. Experimental results show an effective improvement by the proposed method over the conventional one in terms of better SNR and human listening test results. The proposed method can be combined with other existing or future coding methods such as perceptual coding to improve their performance at low bit rates.

A spread-spectrum clock generator (SSCG) using fractional-N phase-locked loop (PLL) with an extended range sigma-delta (ΣΔ) modulator is presented in this paper. The proposed ΣΔ modulator simply adds an extra output bit in the first stage modulator. It can enlarge the input range about three times as compared to the conventional modulator and solve the saturation problem when the input exceeds the boundary of the conventional modulator. A flexible digital modulation controller can generate center and down spread-spectrum modulation and each has spread ratios of 0.4%, 0.8%, 1.6% and 3.2%. The proposed SSCG has been fabricated in TSMC 0.35-µm double-poly quadruple-metal CMOS process with output frequency of 300 MHz. The active area is 0.630.62 mm2 and the power consumption is 17.5 mW.

Constraint-based software specifications enable run-time monitoring to detect probable risk events and ensure the desired system behavior. SpecTRM-RL is a well-developed constraint-based specification method for computer-controlled systems. However, it is desirable to express constraints in familiar visual models. To provide better visualization and popularity, we developed methods to represent all the SpecTRM-RL constraint types in UML. We have also extended SpecTRM's constraints by adding relational and global constraints, and then expressed them in OCL. Safety verification of these specifications is also proposed. We developed a systematic way to construct fault trees for safety analysis based on UML diagrams. Due to the generality of UML as well as the defensive manner of constraints and fault tree analysis, our approach can be adapted for both general applications and safety-critical applications.

In the European Telecommunication Standards Institute (ETSI), Distributed Speech Recognition (DSR) front-end, the distortion added due to feature compression on the front end side increases the variance flooring effect, which in turn increases the identification error rate. The penalty incurred in reducing the bit rate is the degradation in speaker recognition performance. In this paper, we present a nontraditional solution for the previously mentioned problem. To reduce the bit rate, a speech signal is segmented at the client, and the most effective phonemes (determined according to their type and frequency) for speaker recognition are selected and sent to the server. Speaker recognition occurs at the server. Applying this approach to YOHO corpus, we achieved an identification error rate (ER) of 0.05% using an average segment of 20.4% for a testing utterance in a speaker identification task. We also achieved an equal error rate (EER) of 0.42% using an average segment of 15.1% for a testing utterance in a speaker verification task.

Software inspection is a widely acknowledged effective quality improvement method in software development by detecting defects involved in software artifacts and removing them. In research on software inspection, constructing computer supported inspection systems is a major topic in the field. A lot of systems have been reported. However few inspection support systems for model diagrams, especially UML diagrams, have been emerged. We identified four key requirements an inspection support system for UML diagrams should have. They are as follows: 1) direct annotations are given to model diagrams, 2) version management is provided so that evolution of artifacts can be managed, 3) the whole inspection process should be supported, 4) horizontal and vertical readings are supported. This paper describes design and implementation of our inspection support system for UML diagrams to realize the four requirements.

Recently, system level design languages (SLDL), which can describe both hardware and software aspects of the design, are receiving attention. Mixed-signal extensions of SLDL enable current discrete-oriented SLDLs to describe and simulate not only digital systems but also digital-analog mixed-signal systems. The synchronization between discrete and continuous behaviors is widely regarded as a critical part in the extensions. In this paper, we present an event-driven synchronization mechanism for both timed and untimed system level designs through which discrete and continuous behaviors are synchronized via AD events and DA events. We also demonstrate how the synchronization mechanism can be incorporated into the kernel of SLDL, such as SpecC. In the extended kernel, a new simulation cycle, the AMS cycle, is introduced. Three case studies show that the extended SpecC-based system level design environment using our synchronization mechanism works well with timed/untimed mixed-signal system level description.

The k-edge-connectivity augmentation problem for a specified set of vertices of a graph with degree constraints, kECA-SV-DC, is defined as follows: "Given an undirected multigraph G = (V,E), a specified set of vertices S ⊆V and a function g: V → Z+ ∪{∞}, find a smallest set E' of edges such that (V,E ∪ E') has at least k edge-disjoint paths between any pair of vertices in S and such that, for any v ∈ V, E' includes at most g(v) edges incident to v, where Z+ is the set of nonnegative integers." This paper first shows polynomial time solvability of kECA-SV-DC and then gives a linear time algorithm for 2ECA-SV-DC.

We consider a watermark application to assist in the integrity maintenance and verification of the associated images. There is a great benefit in using WM in the context of authentication since it does not require any additional storage space for supplementary metadata, in contrast with cryptographic signatures, for instance. However there is a fundamental problem in the case of exact authentication: How to embed a signature into a cover message in such a way that it would be possible to restore the watermarked cover image into its original state without any error? There are different approaches to solve this problem. We use the watermarking method consisting of modulo addition of a mark and investigate it in detail. Our contribution lies in investigating different modified techniques of both watermark embedding and detection in order to provide the best reliability of watermark authentication. The simulation results for different types of embedders and detectors in combination with the pictures of watermarked images are given.

It is well known that TCP does not fully utilize the available bandwidth in fast long-distance networks. To solve this scalability problem, several high speed transport protocols have been proposed. They include HighSpeed TCP (HS-TCP), Scalable TCP (S-TCP), Binary increase control TCP (BIC-TCP), and H-TCP. These protocols increase (decrease) their window size more aggressively (slowly) compared to standard TCP (STD-TCP). This paper aims at evaluating and comparing these high speed transport protocols through computer simulations. We select six metrics that are important for high speed protocols; scalability, buffer requirement, TCP friendliness, TCP compatibility, RTT fairness, and responsiveness. Simulation scenarios are carefully designed to investigate the performance of these protocols in terms of the metrics. Results clarify that each high speed protocol successfully solves the problem of STD-TCP. In terms of the buffer requirement, S-TCP and BIC-TCP have better performance. For TCP friendliness and compatibility, HS-TCP and H-TCP offer better performance. For RTT fairness, BIC-TCP and H-TCP are superior. For responsiveness, HS-TCP and H-TCP are preferred. However, H-TCP achieves a high degree of fairness at the expense of the link utilization. Thus, we understand that all the proposed high speed transport protocols have their own shortcomings. Thus, much more research is needed on high speed transport protocols.

IP multicast technology is highly advantageous for various applications and future needs in the Internet. Yet, it is generally recognized that the IP multicast routing protocol is fairly complex and non-scalable and requires additional maintenance and operational cost to network administrators. Although there has been much research related to IP multicast and most router vendors already support basic IP multicast routing protocols, there is still a big gap between what is reported as the state-of-the-art in the literature from what is implemented in practice. In this paper, we clarify the complexities of traditional multicast communication and describe possible solutions using the one-to-many multicast communication model called Source-Specific Multicast (SSM). We explain this communication model and the corresponding routing architecture and examine the statistics obtained for the number of multicast routing entries in our backbone router, which is connected to the international backbone. We also introduce our international collaboration activities that are contributing to the deployment and promotion of IP multicast services in the Internet.

We theoretically analyze the performance of coherent ultrashort light pulse code-division multiple-access (CDMA) communication systems with a nonlinear optical thresholder. The coherent ultrashort light pulse CDMA is a promising system for an optical local area network (LAN) due to its advantages of asynchronous transmission, high information security, multiple access capability, and optical processing. The nonlinear optical thresholder is based on frequency chirping induced by self-phase modulation (SPM) in optical fiber, and discriminates an ultrashort pulse from multiple access interference (MAI) with picosecond duration. The numerical results show that the thermal noise caused in a photodetector dominates the bit error rate (BER). BER decreases as the fiber length in the nonlinear thresholder and the photocurrent difference in the photodetector increase. Using the nonlinear optical thresholder allows for the response time of the photodetector to be at least 100 times the duration of the ultrashort pulses. We also show that the optimum cut-off frequency at the nonlinear thresholder to achieve the minimum BER increases with fiber length, the total number of users, and the load resistance in the photodetector.

The speech noise reduction system based on the frequency domain adaptive line enhancer using a windowed modified DFT (MDFT) pair is presented. The adaptive line enhancer (ALE) is effective for extracting sinusoidal signals blurred by a broadband noise. In addition, it utilizes only one microphone. Therefore, it is suitable for the realization of speech noise reduction in portable electronic devices. In the ALE, an input signal is generated by delaying a desired signal using the decorrelation parameter, which makes the noise in the input signal decorrelated with that in the desired one. In the present paper, we propose to set decorrelation parameters in the frequency domain and adjust them to optimal values according to the relationship between speech and noise. Such frequency domain decorrelation parameters enable the reduction of the computational complexity of the proposed system. Also, we introduce the window function into MDFT for suppressing spectral leakage. The performance of the proposed noise reduction system is examined through computer simulations.

A time domain (TD) speech enhancement technique to improve SNR in noise-contaminated speech is proposed. Additional supplementary scheme is applied to estimate the degree of noise of noisy speech. This is estimated from a function, which is previously prepared as the function of the parameter of the degree of noise. The function is obtained by least square (LS) method using the given degree of noise and the estimated parameter of the degree of noise. This parameter is obtained from the autocorrelation function (ACF) on frame-by-frame basis. This estimator almost accurately estimates the degree of noise and it is useful to reduce noise. The proposed method is based on two-stage processing. In the first stage, subtraction in time domain (STD), which is equivalent to ordinary spectral subtraction (SS), is carried out. In the result, the noise is reduced to a certain level. Further reduction of noise and by-product noise residual is carried out in the second stage, where blind source separation (BSS) technique is applied in time domain. Because the method is a single-channel speech enhancement, the other signal is generated by taking the noise characteristics into consideration in order to apply BSS. The generated signal plays a very important role in BSS. This paper presents an adaptive algorithm for separating sources in convolutive mixtures modeled by finite impulse response (FIR) filters. The coefficients of the FIR filter are estimated from the decorrelation of two mixtures. Here we are recovering only one signal of interest, in particular the voice of primary speaker free from interfering noises. In the experiment, the different levels of noise are added to the clean speech signal and the improvement of SNR at each stage is investigated. The noise types considered initially in this study consist of the synthesized white and color noise with SNR set from 0 to 30 dB. The proposed method is also tested with other real-world noises. The results show that the satisfactory SNR improvement is attained in the two-stage processing.

This paper presents an approach to specification-based testing of concurrent programs with representative test sequences generated from Statecharts. Representative test sequences are a subset of all possible interleavings of concurrent events that define the behaviors of a concurrent program. Because a program's correctness may be determined by checking whether a program implemented all behaviors in its specification or not, the program can be regarded as being correct if it can supply an alternative execution that has the same effects as the program's behavior with each representative test sequence. Based on this observation, we employ each representative test sequence as a seed to generate an automaton that accepts its equivalent sequences to reveal the same behavior. In order to effectively test a concurrent program, the automaton such generated accepts all sequences equivalent to the representative test sequence and it is used to control test execution. This paper describes an automated process of generating automata from a Statecharts specification and shows how the proposed approach works on Statecharts specifications through some examples.

This paper investigates limitations of adjacent channel power ratio (ACPR) improvement in predistortion (pre-D) linearizer used with nonlinear RF power amplifiers (PAs) when the PA model is not perfectly acquired in pre-D design. The error between the physical PA and the nonlinear model is expanded by pre-D function and its power spectral density (PSD) works as limitations in ACPR improvement of the pre-D linearizer. An analytical estimation of ACPR limitations in RF PAs driven by digitally modulated input signal is derived using a formulation of autocorrelation function. The analysis technique is validated with the example of the memory polynomial PA model with the quasi-memoryless pre-D linearizer. The technique is also verified by comparing predicted ACPR limitation with measured limitation for a RF PA with 802.11g input signal.