Question answering (QA) is the task of retrieving an answer in response to a question by analyzing documents. Although most of the efforts in developing QA systems are devoted to dealing with electronic text, we consider it is also necessary to develop systems for document images. In this paper, we propose a method of document image retrieval for such QA systems. Since the task is not to retrieve all relevant documents but to find the answer somewhere in documents, retrieval should be precision oriented. The main contribution of this paper is to propose a method of improving precision of document image retrieval by taking into account the co-occurrence of successive terms in a question. The indexing scheme is based on two-dimensional distributions of terms and the weight of co-occurrence is measured by calculating the density distributions of terms. The proposed method was tested by using 1253 pages of documents about the major league baseball with 20 questions and found that it is superior to the baseline method proposed by the authors.

We calculated the shielding characteristic of a three-dimensional array of strip conductors by using the electric field integral equation method and its expansion to an array structure. From reflection coefficients, the effective permittivity of the array is calculated. The effective permittivity becomes negative in the frequency range above resonance, in which the electromagnetic waves travel through the material in an evanescent mode and the transmission coefficient becomes very small.

This paper proposes a new class of polyphase ZCZ (zero-correlation zone) sequence sets which satisfy a mathematical upper bound. The proposed ZCZ sequence sets are obtained from DFT matrices and unitary matrices. In addition, this paper discusses the cross-correlation property between different ZCZ sequence sets which belong to the proposed class.

In this letter, using techniques from linear algebra and coding theory, we characterize the quadratic Boolean functions represented by trace. We show that a linear combination of trace-terms over finite field can be determined to be bent by a polynomial GCD computation. Then we derive some new families of bent functions.

This paper proposes a method for estimating the number and locations of multiple targets distributed in the ocean. This is achieved by calculating the cross spectral density matrix (CSDM) generated from individual sound sources and applying them to a minimum variance distortionless response (MVDR), a nonlinear processor. The individual CSDMs are calculated by separating and extracting a Sequence CLEAN-based data vector from the CSDM of the data vector received from multiple targets. Numerical simulations demonstrate that the proposed method improves the MVDR performance in the case of multiple targets.

The direct optical switching CDMA radio-on-fiber network is proposed as multiplexing method for radio base stations in microcell/picocell technologies. In this system, aliasing distortions degrade the received signal quality and decreases the number of connectable radio base stations (RBSs) when the encoding rate becomes lower than the twice of bandwidth of RF signal. This paper proposes the optical switching speed reduction method and clarifies that the chaotic spreading sequences can suppress the aliasing distortion without additional processing at the receiver even if the encoding rate becomes lower than the RF signal bandwidth. The effect in switching speed reduction is theoretically investigated and the effect in aliasing distortions suppression is experimentally investigated by using the proposed method.

In an Orthogonal Frequency Division Multiplexing (OFDM) systems, the Peak to Average power Ratio (PAR) is high. The clipping signal scheme is a useful and simple method to reduce the PAR. However, it introduces additional noise that degrades the systems performance. We propose an oversampling scheme to deal with the received signal in order to reduce the clipping noise by using finite impulse response (FIR) filter. Coefficients of the filter are obtained by correlation function of the received signal and the oversampling information at receiver. The performance of the proposed technique is evaluated for frequency selective channel. Results show that the proposed scheme can mitigate the clipping noise significantly for OFDM systems and in order to maintain the system's capacity, the clipping ratio should be larger than 2.5.

This paper presents a study on the blind separation of a convoluted mixture of speech signals using Frequency Domain Independent Component Analysis (FDICA) algorithm based on the negentropy maximization of Time Frequency Series of Speech (TFSS). The comparative studies on the negentropy approximation of TFSS using generalized Higher Order Statistics (HOS) of different nonquadratic, nonlinear functions are presented. A new nonlinear function based on the statistical modeling of TFSS by exponential power functions has also been proposed. The estimation of standard error and bias, obtained using the sequential delete-one jackknifing method, in the approximation of negentropy of TFSS by different nonlinear functions along with their signal separation performance indicate the superlative power of the exponential-power-based nonlinear function. The proposed nonlinear function has been found to speed-up convergence with slight improvement in the separation quality under reverberant conditions.

The present paper introduces a new construction of a class of binary sequence set having a zero-correlation zone (hereafter binary zcz sequence set). The cross-correlation function and the side-lobe of the auto-correlation function of the proposed sequence set is zero for the phase shifts within the zero-correlation zone. The present paper shows that such a construction generates a binary zcz sequence set by using a primitive linear recursion over GF(2), the finite field of integers modulo 2.

Jacobian Adaptation (JA) has been successfully used in Automatic Speech Recognition (ASR) systems to adapt the acoustic models from the training to the testing noise conditions. In this work we present an improvement of JA for speaker verification, where a specific training noise reference is estimated for each speaker model. The new proposal, which will be referred to as Model-dependent Noise Reference Jacobian Adaptation (MNRJA), has consistently outperformed JA in our speaker verification experiments.

In this paper, we propose an OFDM scheme with pre-IDFT/DFT and the frequency domain equalization on frequency-selective Rayleigh fading channels. In this scheme, a two-dimensional block interleaving is used to randomize the correlated noise caused by the frequency domain linear equalizer. Then, the pre-DFT averages the interleaved noise enhancement and improves the error performance of the proposed scheme. Computer simulations confirm the bit error probability of the proposed scheme for multilevel modulations.

An approach based on signal subspace analysis is proposed to blind estimation of the PN (Pseudo Noise) sequence from lower SNR (Signal to Noise Ratios) DS/SS (Direct Sequence Spread Spectrum) signals. The received signal is divided into vectors according to a temporal window, from which an autocorrelation matrix is computed and accumulated. The PN sequence can be reconstructed from principal eigenvectors of the matrix.

This paper focuses on the underdetermined blind source separation (BSS) of three speech signals mixed in a real environment from measurements provided by two sensors. To date, solutions to the underdetermined BSS problem have mainly been based on the assumption that the speech signals are sufficiently sparse. They involve designing binary masks that extract signals at time-frequency points where only one signal was assumed to exist. The major issue encountered in previous work relates to the occurrence of distortion, which affects a separated signal with loud musical noise. To overcome this problem, we propose combining sparseness with the use of an estimated mixing matrix. First, we use a geometrical approach to detect when only one source is active and to perform a preliminary separation with a time-frequency mask. This information is then used to estimate the mixing matrix, which allows us to improve our separation. Experimental results show that this combination of time-frequency mask and mixing matrix estimation provides separated signals of better quality (less distortion, less musical noise) than those extracted without using the estimated mixing matrix in reverberant conditions where the reverberant time (TR) was 130 ms and 200 ms. Furthermore, informal listening tests clearly show that musical noise is deeply lowered by the proposed method comparatively to the classical approaches.

In an empowered Internet with end-to-end Quality of Service (QoS) facilities, it is essential for content servers to minimize reserved network resources to achieve a reduction in transmission cost for the use of QoS. Resource reservation usage charging forces customers to efficiently use network resources. In this paper, we analyze a model that optimizes the total cost when semi-elastic traffic flows are delivered in a client-server scenario. The client uses the proposed analytical model to easily control its buffer occupancy and to determine at any time if it is needed to utilize resource reservation or best-effort transmission mode.

This paper presents an adaptive control design for the ABR traffic congestion control in ATM networks. Firstly, we consider a control-based mathematical model to the ABR traffic congestion control problem. Then the feedback pole placement control design is applied to the ATM ABR traffic congestion control problem for the case of known delays. Finally, by using the online plant parameter estimation algorithm and modifying the controller parameters adaptively in real time, a method to treat the case of unknown time-varying delays is proposed. Several design modifications are introduced to solve practical control issues such as bounded command rate constraint, output buffer saturation and bounded values to the plant parameter estimation algorithm. Simulations are implemented to verify the proposed control design. It is shown that while considering these practical control issues, the control method satisfies the requirements of fairness to users, network efficiency, unknown time-varying delays, queue length control and good convergence performance at an acceptable computation effort.

Severe frequency-selective fading, encountered in a broadband wireless mobile communication, significantly degrades the bit error rate (BER) performance of direct sequence spread spectrum (DSSS) signal transmission with rake combining. In this paper, frequency-domain pre-equalization transmission, called pre-FDE transmission, is presented for orthogonal multicode DSSS signal transmission. It is confirmed by the computer simulation that pre-FDE transmission can achieve a BER performance almost identical to that attainable by FDE reception.

This paper proposes per-tone equalization methods for single carrier block transmission with cyclic prefix (SC-CP) systems. Minimum mean-square-error (MMSE) based optimum weights of the per-tone equalizers are derived for SISO (single-input single-output), SIMO (single-input multiple-output), and MIMO (multiple-input multiple-output) SC-CP systems. Unlike conventional frequency domain equalization methods, where discrete Fourier transform (DFT) is employed, the per-tone equalizers utilize sliding DFT, which makes it possible to achieve good performance even when the length of the guard interval is shorter than the channel order. Computer simulation results show that the proposed equalizers can significantly improve the bit error rate (BER) performance of the SISO, SIMO, and MIMO SC-CP systems with the insufficient guard interval.

While mobile networks have been enhanced to support a variety of mobile multimedia services such as video telephony and rich data content delivery, a new challenge is being created by the remarkable development of micro-device technologies such as micro processor-chips, sensors, and RF tags. These developments suggest the rapid emergence of the ubiquitous computing environment; computers supporting human life without imposing any stress on the users. The combination of broadband global networks and ubiquitous computing environment will lead to an entirely new class of services, which we call ubiquitous networking services. This paper discusses how to create ubiquitous service environments comparing global networking approaches which are based on fixed and mobile networks. It is shown that the mobile approach is better from service applicability and reliability viewpoints. Networking architecture is proposed which expand 4G mobile cellular networks to real space via gateways on the edges of the mobile network (i.e. mobile terminals). A new set of technical requirements will emerge via this approach, which may accelerate the paradigm shift from the current mobile network architecture and even from the Internet of today.

This paper describes two way satellite system environments on geostationary orbit (GEO) and performance enhancement mechanisms which reduces round trip time (RTT) and supports real-time services. We use performance enhancing proxy (PEP) for reducing round trip time and user-level real-time scheduler for reducing deadline violation tasks. The user-level real-time scheduling method classifies priority of user process into four types and those are reflected in kernel. With these dual performance enhancement mechanisms, we can improve quality of service (QoS) of end-user who connects to the DVB-RCS system.

The latest video coding standard, H.264/AVC, adopts 44 approximate transform instead of 88 discrete cosine transform (DCT) to avoid the inverse transform mismatch problem. However, that is only one of the factors that make it difficult to transcode pre-coded video contents with the previous standards to H.264/AVC in the common domain without causing cascaded pixel-domain transcoding. In this paper, to support the existent DCT-domain transcoding schemes and to reduce computational complexity, we propose an efficient algorithm that converts the quantized 88 DCT block into four newly quantized 44 transformed blocks. The experimental results show that the proposed scheme reduces computational complexity by 5-11% and improves video quality by 0.1-0.5 dB compared with the cascaded pixel-domain transcoding scheme that exploits inverse quantization (IQ), inverse DCT (IDCT), DCT, and re-quantization (re-Q).