Following the developments in the use of ID-based schemes and smart cards, Yang and Shieh proposed two password authentication schemes to achieve two purposes: (1) to allow users to choose and change their passwords freely, and (2) to make it unnecessary for the remote server to maintain a directory of passwords or a verification table to authenticate users. Recently, Chan and Cheng showed that Yang and Shieh's timestamp-based password authentication scheme is insecure against forgery. In this paper, we point out that Chan and Cheng's forgery attack can not work. Thus, we further examine the security of Yang and Shieh's password authentication schemes and find that they are insecure against forgery because one adversary can easily pretend to be a valid user and pass the server's verification which allows the adversary to login to the the remote server.

In this letter, we propose a new technique that reduces the computation time to derive the MEI coefficients in solving scattering problems by the Measured Equation of Invariance (MEI) methods. Methods that use the MEI technique spend most of the computation time in the integration process to derive the MEI coefficients. Moreover, in the conventional solution process, some repeated computation of metron fields to derive the MEI coefficients is included. To avoid the repeated operations and thus save computation time, we propose a matrix localization technique in computing the MEI coefficients. The time comparison for the computation of MEI coefficients of a cylinder and a sphere is given to verify the CPU time reduction of our new technique.

We propose a new algorithm for blind source separation (BSS), in which frequency-domain independent component analysis (FDICA) and time-domain ICA (TDICA) are combined to achieve a superior source-separation performance under reverberant conditions. Generally speaking, conventional TDICA fails to separate source signals under heavily reverberant conditions because of the low convergence in the iterative learning of the inverse of the mixing system. On the other hand, the separation performance of conventional FDICA also degrades significantly because the independence assumption of narrow-band signals collapses when the number of subbands increases. In the proposed method, the separated signals of FDICA are regarded as the input signals for TDICA, and we can remove the residual crosstalk components of FDICA by using TDICA. The experimental results obtained under the reverberant condition reveal that the separation performance of the proposed method is superior to those of TDICA- and FDICA-based BSS methods.

This paper proposes a low power consumption Analog Matched Filter (AMF) that utilizes capacitor multiply-and-accumulate operations. A high-speed, high-precision Analog-to-Digital (A/D) converter is unnecessary because the proposed circuit directly samples received analog signals. A code-shifting MF structure is used to prevent errors from accumulating. A 15-tap AMF circuit was fabricated using 0.35 µm CMOS technology. Power consumption for the 128-tap circuit is estimated to be 22.3 mW at 25 MHz and 3.3 V, and the area is estimated to be 0.33 mm2. The proposed circuit will thus be a useful LSI for mobile terminals.

The label placement problem is one of the most important problems in geographic information systems, cartography, graph drawing and graphical interface design. In this paper, we consider the problem of labeling points and curves in maps drawn from digital data. In digital maps, a curve is represented as a set of points and consists of many small segments. The label for each curve must be placed alongside the corresponding curve. We define a continuous labeling space for points and curves, and present an algorithm using this space for positioning labels. Computational results for subway and JR train maps in Tokyo are presented.

This paper proposes a lossless scalable audio coding scheme and quality enhancement processing at the decoder to compensate for some missing scalable units of information. The bit rate scalability is achieved by combining high-compression coding, such as MPEG-4, and horizontal bit slicing of the PCM-coded error signal between the original waveform and the locally reconstructed MPEG-4 signal. The horizontally sliced stream may be transported through an IP network with priority. Even if some units are missing at the decoder, reasonable quality waveform can be reconstructed by means of preserving the important packets. In addition, quality enhancement procedures including scale adjustment and post-processing have been proposed. The scale adjustment eliminates unnecessary zero's, and the post-processing recovers the spectral envelope characteristics of the original input signal. As a result of objective quality evaluation, the two techniques are confirmed to be useful for quality enhancement when lower priority packets are lost. This scheme enables graceful degradation by supporting lossless, near lossless, and high-compression coding within a single scalable framework, and is useful for narrowband to broadband audio streaming.

This paper presents a method of temporal decomposition (TD) for line spectral frequency (LSF) parameters, called "Modified Restricted Temporal Decomposition" (MRTD), and its application to low rate speech coding. The LSF parameters have not been used for TD due to the stability problems in the linear predictive coding (LPC) model. To overcome this deficiency, a refinement process is applied to the event vectors in the proposed TD method to preserve their LSF ordering property. Meanwhile, the restricted second order TD model, where only two adjacent event functions can overlap and all event functions at any time sum up to one, is utilized to reduce the computational cost of TD. In addition, based on the geometric interpretation of TD the MRTD method enforces a new property on the event functions, named the "well-shapedness" property, to model the temporal structure of speech more effectively. This paper also proposes a method for speech coding at rates around 1.2 kbps based on STRAIGHT, a high quality speech analysis-synthesis method, using MRTD. In this speech coding method, MRTD based vector quantization is used for encoding spectral information of speech. Subjective test results indicate that the speech quality of the proposed speech coding method is close to that of the 4.8 kbps FS-1016 CELP coder.

We propose a method to realize soft-decision decoding for hard-detected signals. In this paper, a novel concept is introduced as "error-detected reliability. " The method is very useful for optical fiber communications (OFC) as hard detection is the only detection method for the OFC systems. We demonstrate our proposed method using the turbo code in which soft information is required for decoding. As a result, the simulation shows slight difference in the range of moderate to high signal-to-noise ratio between the proposed decoding scheme and the conventional turbo decoding scheme. Moreover, the bit error rate of 10-11 can be achieved by serial concatenation of a Reed-Solomon code and a turbo code for Q-factor lower than 8.0 dB with a bandwidth expansion ratio of 33.3%.

In this paper, we propose a TCAD driven hot carrier reduction methodology of 3.3 V I/O pMOSFETs design. The hot carrier reliability of surface channel I/O pMOSFET having drain structure in common with core devices has a critical issue. It is substantially important for the high-reliability devices to reduce both drain avalanche and channel hot hole components. The drain structures are successfully optimized in short time by applications of TCAD local models. Considering tradeoffs between hot carrier injection (HCI) and drive current (ION), SDE/HALO of both core and I/O transistors can be totally optimized for reduction of process-steps and/or photo-masks.

This paper proposes a new multistage technique of algebraic codebook in CELP coders called Trellis Search inspired from the Trellis Coded Quantization (TCQ). This search technique is implemented into the fixed codebook of the standard G.729 for objective evaluation on a large corpus of a testing speech database. Simulations results show that in terms of computer execution time the proposed search scheme reduces the codebook search by approximately 23% compared to the time of focused search used in the standard G.729. This yields to a reduction of about 8% in the computer execution time of the coder at the cost of a slight degradation of speech quality but perceptually not noticeable. Moreover, this new technique shows better speech quality than the G.729A at the expense of a higher complexity.

In this letter, we study the performance of the iterative receiver as applied to the space division multiplexing/orthogonal frequency division multiplexing (SDM/OFDM) systems. The iterative receiver under consideration employs the soft in/soft out (SISO) decoding process, which operates iteratively in conjunction with channel estimation for performing data detection and channel estimation at the same time. As opposed to the previous studies in which the perfect channel state information is assumed, the effects of channel estimation are taken into account for evaluating the performance of the iterative receiver and it is shown that the channel estimation applied in every iteration step of the iterative receiver plays a crucial role to warrant the performance, especially at a low signal-to-noise power ratio (SNR).

Appropriate test signals defined by formula or generated by algorithm are used for measuring objective QoS (Quality of Services) for voice operated telecommunication devices such as telephone and speech codec (coder-decoder). However, that for measuring residual echo characteristics in hands-free telecommunications equipped with acoustic echo canceller is under study in ITU-T Recommendation G.167. This paper describes comparative assessment of test signals for measurement of residual echo characteristics. In hands-free telecommunications, acoustical echo canceller has been developed to remove a room echo signal through the loudspeaker to the microphone in the receiving end. Performance of the echo canceller system is evaluated by residual echo characteristics expressed in echo return loss enhancement (ERLE). The ERLE can be conventionally measured by putting white noise into the echo canceller system. However, white noise is not adequate as the test signal for measuring the performance of the echo canceller, since the performance may depend on the characteristics of input test signal, and the characteristics of the white noise differ from those of real voice. Therefore, this paper discusses appropriate characteristics of real voice required for objective quality evaluation of echo canceller system. The test signals used for this verification tests were real voice (RV), white noise (WN), frequency weighted noise (FWN), artificial voice (AV), and composite source signal (CSS) depending on the approximation of real voice characteristics. As the comparative assessment results, the ERLE characteristics measured by artificial voice conforming to ITU-T Recommendation P.50 having average characteristics of real voices in time and frequency domains are almost equivalent to those of real voice and best among those test signals. It is concluded that artificial voice P.50 is satisfied with measurement of residual echo characteristics.

The present paper deals with the blind deconvolution of a Multiple-Input Multiple-Output Finite Impulse Response (MIMO-FIR) system. To deal with the blind deconvolution problem using the second-order statistics (SOS) of the outputs, Hua and Tugnait considered it under the conditions that a) the FIR system is irreducible and b) the input signals are spatially uncorrelated and have distinct power spectra. In the present paper, the problem is considered under a weaker condition than the condition a). Namely, we assume that c) the FIR system is equalizable by means of the SOS of the outputs. Under b) and c), we show that the system can be blindly identified up to a permutation, a scaling, and a delay using the SOS of the outputs. Moreover, based on this identifiability, we show a novel necessary and sufficiently condition for solving the blind deconvolution problem, and then, based on the condition, we propose a new algorithm for finding an equalizer using the SOS of the outputs, while Hua and Tugnait have not proposed any algorithm for solving the blind deconvolution under the conditions a) and b).

In this paper, a novel confidence scoring method that is applied to N-best hypotheses (word candidates) output from an HMM-based classifier is proposed. In the first pass of the proposed method, the HMM-based classifier with monophone models outputs N-best hypotheses and boundaries of all monophones in the hypotheses. In the second pass, an SM (Subspace Method)-based verifier tests the hypotheses by comparing confidence scores. To test the hypotheses, at first, the SM-based verifier calculates the similarity between phone vectors and an eigen vector set of monophones, then this similarity score is converted into a likelihood score with normalization of acoustic quality, and finally, an HMM-based likelihood of word level and an SM-based likelihood of monophone level are combined to formulate the confidence measure. Two kinds of experiments were performed to evaluate this confidence measure on speaker-independent word recognition. The results showed that the proposed confidence scoring method significantly reduced the word error rate from 4.7% obtained by the standard HMM classifier to 2.0%, and in an unknown word rejection, it reduced the equal error rate from 9.0% to 6.5%.

It is well known that KT method proposed by R. Kumaresan and D. W. Tufts is used as a popular parameter estimation method of exponentially damped signal. It is based on linear backward-prediction method and singular value decomposition (SVD). However, it is difficult to estimate parameters correctly by KT method in the case when high noise exists in the signal. In this paper, we propose a parameter (frequency components and damping factors) estimation method to improve the performance of KT method under high noise. In our proposed method, we find the signal zero groups by calculating zeros with different data record lengths according to the combination of forward-prediction and backward-prediction, the mean value of the zeros in the signal zero groups are calculated to estimate the parameters of the signal. The proposed method can estimate parameters correctly and accurately even when high noise exists in the signal. Simulation results are shown to confirm the effectiveness of the proposed method.

An adaptive array antenna for the suppression of high-power interference in direct-sequence code-division multiple access (DS-CDMA) systems is presented. Although DS-CDMA has sufficient flexibility to support a variety of services, from voice to moving-pictures, with high levels of quality, multiple access interference (MAI) is a problem. This is particularly so of the high-power interference which accompanies high-speed transmission in DS-CDMA. While the application of adaptive array antennas is an effective way of improving signal-to-interference-plus-noise ratio (SINR), problems with this approach include large levels of power consumption and the high costs of hardware and of implementing the antennas. Therefore, our main purpose is to realize a simple configuration for an adaptive array system. In order to reduce the required amounts of processing, a common beam provides suppression of high-power interference for the low-bit-rate users; this makes per-user preparation of weights unnecessary. This approach also reduces the consumption of power by the system. Interference is cancelled by minimization of the array output power (i.e., the application of a power inversion algorithm) before despreading. The approach also allows us to improve the implementation of the antenna elements by using small auxiliary antennas. The basic performance of the system is confirmed through numerical calculation and computer simulation. Furthermore, a real-time processing unit has been developed and the effectiveness of the approach is confirmed by an experiment in a radio-anechoic chamber.

This paper addresses the system throughput maximization problem for HAPS third generation cellular systems. We assume that the Stratospheric Platform is able to perform a perfect link gain estimation for all mobile terminals, such that a centralized resource allocation strategy is made possible. A classical 3G wireless scenario is considered, where traffics characterized by different bit rates coexist with Best Effort Traffic services without stringent bit rate constraints. In this scenario, we firstly envisage three Rate Assignment schemes for best effort terminals which aim at achieving the maximum system throughput subject to different bit rate constraints. For the second envisaged rate assignment scheme, which represents the best compromise between service fairness and throughput, we then propose a simplified approach that allows to noticeably decrease the implementation complexity with a slight performance degradation.

In many applications of Independent Component Analysis (ICA) and Blind Source Separation (BSS) estimated sources signals and the mixing or separating matrices have some special structure or some constraints are imposed for the matrices such as symmetries, orthogonality, non-negativity, sparseness and specified invariant norm of the separating matrix. In this paper we present several algorithms and overview some known transformations which allows us to preserve several important constraints.

It is still an open problem to elucidate the scaling merits of an embedded SRAM with Low Operating Power (LOP) MOSFETs fabricated in 50, 70 and 100 nm CMOS technology nodes. Taking into account a realistic SRAM cell layout, we evaluated the parasitic capacitance of the bit line (BL) as well as the word line (WL) in each generation. By means of a 3-Dimensional (3D) interconnect simulator (Raphael), we focused on the scaling merit through a comparison of the simulated SRAM BL delay for each CMOS technology node. In this paper, we propose two kinds of original interconnect structure which modify ITRS (International Technology Roadmap for Semiconductors), and make it clear that the original interconnect structures with reduced gate overlap capacitance guarantee the scaling merits of SRAM cells fabricated with LOP MOSFETs in 50 and 70 nm CMOS technology nodes.

This paper reviews research and development on the phased array antennas (PAAs) for several applications in Japan in over past two decades. First, the author shows the historical overview of the PAA for radar, satellite and mobile communication uses. Next, this paper introduces analysis methods for the PAA. It shows mutual coupling analysis methods and pattern synthesis methods for the PAA. Furthermore, the author discusses measurement methods for the PAA. Especially, he explains the rotating-element electric-field vector (REV) method for the Japanese original PAA calibration method. Finally, the author concludes and shows future PAA technologies.