Previous approaches to question retrieval in community-based question answering rely on statistical translation techniques to match users' questions (queries) against collections of previously asked questions. This paper presents a simple but effective method for computing word relatedness to improve question retrieval based on word co-occurrence information directly extracted from question and answer archives. Experimental results show that the proposed approach significantly outperforms translation-based approaches.

The proposed quasi-resonant (QR) zero current switching (ZCS) switched-capacitor (SC) converter is a new type of bidirectional power flow control conversion scheme. The proposed converter is able to provide voltage conversion ratios from -3/- (triple-mode/ trisection-mode) to -n/- (-n-mode/--mode) by adding a different number of switched-capacitors and power MOSFET switches with a small series connected resonant inductor for forward and reverse power flow control schemes. It possesses the advantages of low switching losses and current stress in this QR ZCS SC converter. The principle of operation, theoretical analysis of the proposed triple-mode/ trisection-mode bidirectional power conversion scheme is described in detail with circuit model analysis. Simulation and experimental studies are carried out to verify the performance of the proposed inverting type ZCS SC QR bidirectional converter. The proposed converters can be applied to battery equalization for battery management system (BMS).

In many engineering problems it is required to convert a polynomial into another polynomial through a transformation. Due to its wide range of applications, the polynomial transformation has received much attention and many techniques have been developed to compute the coefficients of a transformed polynomial from those of an original polynomial. In this letter a new result is presented concerning the transformation matrix for arbitrary polynomial transformation. A simple algorithm is obtained which enables one to successively compute transformation matrices of various order.

It is essential, as bandwidths of wireless communications get wider, to evaluate the imbalances among quadrature mixer ports, in terms of carrier phase offset, IQ gain imbalance, and IQ skew. Because it is time consuming to separate skew, gain imbalance and carrier phase offset evaluation during test is often performed using a composite value, without separation of the imbalance factors. This paper describes an algorithm for enabling separation among quadrature mixer gain imbalance, carrier phase offset, and skew. Since the test time is reduced by the proposed method, it can be applied during high volume production testing.

Logical operations in CMOS digital integration are highly prone to fail as the amount of power supply (PS) drop approaches to failure threshold. PS voltage variation is characterized by built-in noise monitors in a 32-bit microprocessor of 90-nm CMOS technology, and related with operation failures by instruction-level programming for logical failure analysis. Combination of voltage drop size and activated logic path determines failure sensitivity and class of failures. Experimental observation as well as simplified simulation is applied for the detailed understanding of the impact of PS noise on logical operations of digital integrated circuits.

In this paper, we propose a distance-based howling canceller with high speech quality. We have developed a distance-based howling canceller that uses only distance information by noticing the property that howling occurs according to the distance between a loudspeaker and a microphone. This method estimates the distance by transmitting a pilot signal from the loudspeaker to the microphone. Multiple frequency candidates for each howling are computed from the estimated distance and eliminated by cascading notch filters that have nulls at them. However degradation of speech quality occurs at the howling canceller output. The first cause is a shot noise occurrence at the beginning and end of the pilot signal transmission due to the discontinuous change of the amplitude. We thus develop a new pilot signal that is robust against ambient noises. We can then reduce the shot noise effect by taking the amplitude small. The second one is a speech degradation caused from overlapped stopbands of the notch filters. We thus derive a condition on the bandwidths so that stopbands do not overlap, and propose an adaptive bandwidth scheme which changes the bandwidth according to the distance.

As we scale toward nanometer technologies, the increase in interconnect parameter variations will bring significant performance variability. New design methodologies will emerge to facilitate construction of reliable systems from unreliable nanometer scale components. Such methodologies require new performance models which accurately capture the manufacturing realities. In this paper, we present a Linear Fractional Transform (LFT) based model for interconnect parametric uncertainty. The new model formulates the interconnect parametric uncertainty as a repeated scalar uncertainty structure. With the help of generalized Balanced Truncation Realization (BTR) and Linear Matrix Inequalities (LMI's), the porposed model reduces the order of the original interconnect network while preserves the stability. The LFT based new model even guarantees passivity if the BTR reduction is based on solutions to a pair of Linear Matrix Inequalities (LMI's) generated from Lur'e equations. In case of large number of uncertain parameters, the new model may be applied successively: the uncertain parameters are partitioned into groups, and with regard to each group, LFT based model is applied in turns.

Recently, there has been growing interest in the design of wireless cooperative protocol to achieve higher diversity-multiplexing tradeoff among single antenna devices. We propose an automatic request for cooperation (ARC) scheme for wireless networks which can achieve higher order diversity by selecting the best relay. In this scheme, a source transmits a data packet towards a destination and a group of relays. The destination tries to decode the information from the source and if the detection is correct the process will stop. Otherwise, the destination transmits an ARC towards the relays. We utilize this ARC signal for selecting the best relay from the set of relays that have successfully decoded the source packet. The selected relay generates and transmits redundant information for the source packet. The destination combines the two packets received from the source and the best relay to improve the reliability of the packet. We analyze the packet error rate, spectral efficiency and diversity-multiplexing tradeoff of our proposal and compare them with some existing protocols. Analysis shows that our proposal can achieve higher diversity multiplexing tradeoff than conventional cooperative protocols.

In this letter, we propose a computationally efficient search space for QRM-MLD that is used for spatially multiplexed multiple antenna systems. We perform a set of computer simulations to show that the proposed method achieves a performance that is near to that of the original QRM-MLD, while its computational complexity is near to that of rank-QRM-MLD.

A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decomposing algorithm as well as the virtual source representation are confirmed.

In this letter, we propose a computationally effient equalization technique that employs block minimum mean squared error (MMSE) depending on LDLH factorization. Parallel interference cancellation (PIC) is executed with pre- obtained output to provide more reliable symbol detection. In particular, the band structure of the frequency domain channel matrix is exploited to reduce the implementation complexity. It is shown through computer simulation that the proposed technique requires lower complexity than the conventional algorithm to obtain the same performance, and that it exhibits better performance than the conventional counterpart when the same complexity is assumed.

Similar to orthogonal frequency-division multiplexing (OFDM) systems, orthogonal frequency-division multiple access (OFDMA) is vulnerable to carrier frequency offset (CFO). Since the CFO of each user is different, CFO compensation in OFDMA uplink is much more involved than that in OFDM systems. It has been shown that the zero-forcing (ZF) compensation method is a simple yet effective remedy; however, it requires the inversion of a large matrix and the computational complexity can be very high. Recently, we have developed a low-complexity iterative method to alleviate this problem. In this paper, we consider the theoretical aspect of the algorithm. We specifically analyze the output signal-to-interference-plus-noise-ratio (SINR) of the algorithm. Two approaches are used for the analysis; one is simple but approximated, and the other is complicated but exact. The convergence problem is also discussed. In addition to the analysis, we propose a pre-compensation (PC) method enhancing the performance of the algorithm. Simulations show that our analysis is accurate and the PC method is effective.

This letter corrects some errors on a previous letter concerning the derivation of the covariance matrix of phase noise. This derivation doesn't affect the results of the previous letter.

The observed phenomena in actual sound and electromagnetic environment are inevitably contaminated by the background noise of arbitrary distribution type. Therefore, in order to evaluate sound and electromagnetic environment, it is necessary to establish some signal processing methods to remove the undesirable effects of the background noise. In this paper, we propose noise cancellation methods for estimating a specific signal with the existence of background noise of non-Gaussian distribution from two viewpoins of static and dynamic signal processing. By applying the well-known least mean squared method for the moment statistics with several orders, practical methods for estimating the specific signal are derived. The effectiveness of the proposed theoretical methods is experimentally confirmed by applying them to estimation problems in actual sound and magnetic field environment.

In this paper, we propose a rate control technique for H.264/AVC using subjective quality of video for off line video coding. We propose to use Video Quality Metric (VQM) with an evolution strategy algorithm, which is capable of identifying the best possible quantization parameters for each frame/macroblock to encode the video sequence such that it would maximize the subjective quality of the entire video sequence subjected to the target bit rate. Simulation results suggest that the proposed technique can improve the RD performance of the H.264/AVC codec significantly. With the proposed technique, up to 40% bit rate reduction can be achieved at the same video quality. Furthermore, results show that the proposed technique can improve the subjective quality of the encoded video significantly for video sequences especially with high motion.

Distributed Video Coding (DVC) is an emerging video coding paradigm that is characterized by a flexible architecture for designing very low cost video encoders. This feature could be very effectively utilized in a number of potential many-to-one type video coding applications. However, the compression efficiency of the latest DVC implementations still falls behind the state-of-the-art in conventional video coding technologies, namely H.264/AVC. In this paper, a novel non-linear quantization algorithm is proposed for DVC in order to improve the rate-distortion (RD) performance. The proposed solution is expected to exploit the dominant contribution to the picture quality from the relatively small coefficients when the high concentration of the coefficients near zero as evident when the residual input video signal for the Wyner-Ziv frames is considered in the transform domain. The performance of the proposed solution incorporating the non-linear quantizer is compared with the performance of an existing transform domain DVC solution that uses a linear quantizer. The simulation results show a consistently improved RD performance at all bitrates when different test video sequences with varying motion levels are considered.

In laparoscopic surgery, the lack of tactile sensation and 3D visual feedback make it difficult to identify the position of a blood vessel intraoperatively. An unintentional partial tear or complete rupture of a blood vessel may result in a serious complication; moreover, if the surgeon cannot manage this situation, open surgery will be necessary. Differentiation of arteries from veins and other structures and the ability to independently detect them has a variety of applications in surgical procedures involving the head, neck, lung, heart, abdomen, and extremities. We have used the artery's pulsatile movement to detect and differentiate arteries from veins. The algorithm for change detection in this study uses edge detection for unsupervised image registration. Changed regions are identified by subtracting the systolic and diastolic images. As a post-processing step, region properties, including color average, area, major and minor axis lengths, perimeter, and solidity, are used as inputs of the LVQ (Learning Vector Quantization) network. The output results in two object classes: arteries and non-artery regions. After post-processing, arteries can be detected in the laparoscopic field. The registration method used here is evaluated in comparison with other linear and nonlinear elastic methods. The performance of this method is evaluated for the detection of arteries in several laparoscopic surgeries on an animal model and on eleven human patients. The performance evaluation criteria are based on false negative and false positive rates. This algorithm is able to detect artery regions, even in cases where the arteries are obscured by other tissues.

Two discrete-time Wirtinger-type inequalities relating the power of a finite-length signal to that of its circularly-convolved version are developed. The usual boundary conditions that accompany the existing Wirtinger-type inequalities are relaxed in the proposed inequalities and the equalizing sinusoidal signal is free to have an arbitrary phase angle. A measure of this sinusoidal signal's power, when corrupted with additive noise, is proposed. The application of the proposed measure, calculated as a ratio, in the evaluation of the power of a sinusoid of arbitrary phase with the angular frequency π/N, where N is the signal length, is thoroughly studied and analyzed under additive noise of arbitrary statistical characteristic. The ratio can be used to gauge the power of sinusoids of frequency π/N with a small amount of computation by referring to a ratio-versus-SNR curve and using it to make an estimation of the noise-corrupted sinusoid's SNR. The case of additive white noise is also analyzed. A sample permutation scheme followed by sign modulation is proposed for enlarging the class of target sinusoids to those with frequencies M π/N, where M and N are mutually prime positive integers. Tandem application of the proposed scheme and ratio offers a simple method to gauge the power of sinusoids buried in noise. The generalization of the inequalities to convolution kernels of higher orders as well as the simplification of the proposed inequalities have also been studied.

Slotted wireless ad hoc networks are drawing more and more attention because of their advantage of QoS (Quality of Service) support for multimedia applications owing to their collision-free packet transmission. Time slot assignment is an unavoidable and important problem in such networks. The existing time slot assignment methods have in general a drawback of limited available bandwidth due to their local assignment optimization without the consideration of directions of the radio wave transmission of wireless links along the routes in such networks. A new time slot assignment is proposed in this paper in order to overcome this drawback. The proposed assignment is different from the existing methods in the following aspects: a) consideration of link directions during time slot assignment; b) largest bandwidth to be achieved; c) feasibility in resource limited ad hoc networks because of its fast assignment. Moreover, the effectiveness of the proposal is confirmed by some simulation results.

It is a tedious and subjective task to measure severity of a dysarthria by manually evaluating his/her speech using available standard assessment methods based on human perception. This paper presents an automated approach to assess speech quality of a dysarthric speaker with cerebral palsy. With the consideration of two complementary factors, speech consistency and speech distinction, a speech quality indicator called speech clarity index (Ψ) is proposed as a measure of the speaker's ability to produce consistent speech signal for a certain word and distinguished speech signal for different words. As an application, it can be used to assess speech quality and forecast speech recognition rate of speech made by an individual dysarthric speaker before actual exhaustive implementation of an automatic speech recognition system for the speaker. The effectiveness of Ψ as a speech recognition rate predictor is evaluated by rank-order inconsistency, correlation coefficient, and root-mean-square of difference. The evaluations had been done by comparing its predicted recognition rates with ones predicted by the standard methods called the articulatory and intelligibility tests based on the two recognition systems (HMM and ANN). The results show that Ψ is a promising indicator for predicting recognition rate of dysarthric speech. All experiments had been done on speech corpus composed of speech data from eight normal speakers and eight dysarthric speakers.