In this letter, a computational approach for solving cross-coupled algebraic Riccati equations (CAREs) is investigated. The main purpose of this letter is to propose a new algorithm that combines Newton's method with a gradient-based iterative (GI) algorithm for solving CAREs. In particular, it is noteworthy that both a quadratic convergence under an appropriate initial condition and reduction in dimensions for matrix computation are both achieved. A numerical example is provided to demonstrate the efficiency of this proposed algorithm.

In this letter, we examine the linear complexity of some 3-ary sequences, proposed by No, of period 3n-1(n=3ek, e, k integer) with the ideal autocorrelation property. The exact value of linear complexity k(6e)w is determined when the parameter r =. Furthermore, the upper bound of the linear complexity is given when the other forms of the value r is taken. Finally, a Maple program is designed to illustrate the validity of the results.

Since a receiver (RX) and a transmitter (TX) are operated simultaneously in a WCDMA transceiver, noise and intermodulation distortion performances of a low-noise amplifier (LNA) are degraded by a large leakage signal from the TX. The degradation of the distortion due to the large leakage signal from the TX has been reported in some previous works, but to our best knowledge, there are no reports about the degradation of noise figure (NF) in a LNA due to the large leakage signal from the TX. In this paper, a 900-MHz LNA for WCDMA terminal with high tolerance for a leakage signal from the TX is proposed. Suitable designs of an input matching circuit and a trap circuit are adopted to improve the tolerance for the leakage signal from the TX. The LNA using the proposed techniques is fabricated using SiGe-BiCMOS process. The measured degradation of NF due to the leakage signal from the TX is suppressed to only 0.12 dB.

A method for extracting a sound signal from a particular area that is surrounded by multiple ambient noise sources is proposed. This method performs several fixed beamformings on a pair of small microphone arrays separated from each other to estimate the signal and noise power spectra. Noise suppression is achieved by applying spectrum emphasis to the output of fixed beamforming in the frequency domain, which is derived from the estimated power spectra. In experiments performed in a room with reverberation, this method succeeded in suppressing the ambient noise, giving an SNR improvement of more than 10 dB, which is better than the performance of the conventional fixed and adaptive beamforming methods using a large-aperture microphone array. We also confirmed that this method keeps its performance even if the noise source location changes continuously or abruptly.

Synthesis of quantum circuits is essential for building quantum computers. It is important to verify that the circuits designed perform the correct functions. In this paper, we propose an algorithm which can be used to verify the quantum circuits synthesized by any method. The proposed algorithm is based on BDD (Binary Decision Diagram) and is called X-decomposition Quantum Decision Diagram (XQDD). In this method, quantum operations are modeled using a graphic method and the verification process is based on comparing these graphic diagrams. We also develop an algorithm to verify reversible circuits even if they have a different number of garbage qubits. In most cases, the number of nodes used in XQDD is less than that in other representations. In general, the proposed method is more efficient in terms of space and time and can be used to verify many quantum circuits in polynomial time.

In this paper, a CMOS analog-to-digital converter (ADC) with a 6-bit 100 MSPS at 1.8 V is described. The architecture of the proposed ADC is based on a folding type with a resistive interpolation technique for low power consumption. To reduce the power consumption, a folder reduction technique to decrease the number of folding blocks (NFB) by half of the conventional ones, an averaging folder technique, and a compensated resistive interpolation technique are proposed. Further, an auto-switching encoder for efficient digital processing is also presented. With the clock speed of 100 MSPS, the ADC achieves an effective resolution bandwidth (ERBW) of 50 MHz, while consuming only 4.5 mW of power. The measured result of figure-of-merit (FoM) is 0.93 [pJ/convstep]. The active chip occupies an area of 0.28 mm2 in 0.18 µm CMOS technology.

This letter deals with computationally efficient maximum-likelihood (ML) detection for the quasi-orthogonal space-time block code (QOSTBC) with four transmit antennas. The proposed ML detector uses a permutation based real-valued equivalent channel matrix representation. As a result, the complexity of ML detection problem is moderated from O(2|A|2) to O(4|A|), where |A| is modulation order. Numerical results show that the proposed ML detector provides ML performance and achieves greatly high computational savings.

This paper presents a new interactive learning method for spoken word acquisition through human-machine audio-visual interfaces. During the course of learning, the machine makes a decision about whether an orally input word is a word in the lexicon the machine has learned, using both speech and visual cues. Learning is carried out on-line, incrementally, based on a combination of active and unsupervised learning principles. If the machine judges with a high degree of confidence that its decision is correct, it learns the statistical models of the word and a corresponding image category as its meaning in an unsupervised way. Otherwise, it asks the user a question in an active way. The function used to estimate the degree of confidence is also learned adaptively on-line. Experimental results show that the combination of active and unsupervised learning principles enables the machine and the user to adapt to each other, which makes the learning process more efficient.

Combination of mutually complementary features is necessary to cope with various changes in pattern classification between normal and pathological voices. This paper proposes a method to improve pathological/normal voice classification performance by combining heterogeneous features. Different combinations of auditory-based and higher-order features are investigated. Their performances are measured by Gaussian mixture models (GMMs), linear discriminant analysis (LDA), and a classification and regression tree (CART) method. The proposed classification method by using the CART analysis is shown to be an effective method for pathological voice detection, with a 92.7% classification performance rate. This is a noticeable improvement of 54.32% compared to the MFCC-based GMM algorithm in terms of error reduction.

This paper proposes several cepstral statistics compensation and normalization algorithms which alleviate the effect of additive noise on cepstral features for speech recognition. The algorithms are simple yet efficient noise reduction techniques that use online-constructed pseudo-stereo codebooks to evaluate the statistics in both clean and noisy environments. The process yields transformations for both clean speech cepstra and noise-corrupted speech cepstra, or for noise-corrupted speech cepstra only, so that the statistics of the transformed speech cepstra are similar for both environments. Experimental results show that these codebook-based algorithms can provide significant performance gains compared to results obtained by using conventional utterance-based normalization approaches. The proposed codebook-based cesptral mean and variance normalization (C-CMVN), linear least squares (LLS) and quadratic least squares (QLS) outperform utterance-based CMVN (U-CMVN) by 26.03%, 22.72% and 27.48%, respectively, in relative word error rate reduction for experiments conducted on Test Set A of the Aurora-2 digit database.

A rate control scheme is described for zero-forcing beamforming (ZFBF) multiuser multiple-input and multiple-output (MU-MIMO) systems with a QR-decomposition maximum likelihood detector (MLD) at the receiver. For selected users, a modulation-and-coding set is selected for each substream by estimating the per-substream post-MLD signal-to-interference-plus-noise ratio. Iterative modified QR-decomposition MLD is employed at the receiver to achieve the throughput expected from the transmitter. The simulation results demonstrated that the proposed rate-control scheme achieved the target packet error rate while increasing the throughout for ZFBF-MU-MIMO systems as the number of user candidates increases.

A new subfamily of sequences with optimal correlation properties is constructed for the generalized Kasami set. A lower bound on the linear span is established. It is proved that with suitable choices of parameters, this subfamily has exponentially larger linear spans than either No sequences or TN sequences. A class of sequences with ideal autocorrelation is also proved to have large linear span.

This paper introduces an adaptive low complexity pre-processing filter to improve the coding performance of seriously degraded video sequences that is caused by the additive noise. The additive noise leads to a decrease in coding performance due to the high frequency components. By incorporating local statistics and quantization parameter into filtering process, the spurious noise is significantly attenuated and coding efficiency is improved for given quantization step size. In order to reduce the complexity of the pre-processing filter, the simplified local statistics and quantization parameter are introduced. The simulation results show the capability of the proposed algorithm.

Multi-carrier code division multiple access (MC-CDMA) has been considered as one of the promising techniques for the next generation of mobile communication systems because of its efficient bandwidth usage, robustness to the multi-path fading and simple channel-sharing scheme. However, MC-CDMA cannot be employed in the uplink communication where the transmitted signal from each user propagates through the different multi-path fading channel, and the received signals are no longer orthogonal at the base station. As a result, bit error rate (BER) performance in the uplink MC-CDMA communication would be strongly degraded due to the occurrence of multi-user interference (MUI). To solve the MUI problem in the uplink MC-CDMA, the pre-equalization method was proposed in which the uplink signal is pre-equalized at the user terminal by using the channel response estimated from the downlink. Although the pre-equalization method is very effective for the stationary uplink channel with fixed users, it is hard to be employed in the time varying fading channel with mobile users, because there is a big difference in the channel responses between downlink and uplink. For the efficient MUI compensation, each user terminal would be required to predict the future channel conditions based on the current observation. This paper proposes a method for model based uplink channel response prediction by employing the spectral decomposition of the downlink channel impulse response. Computer simulation results show that the proposed method can achieve the accurate prediction of channel response for mobile users during the uplink transmission and allows the effective MUI compensation.

Consider an undirected multigraph G=(V,E) with n vertices and m edges, and let Ni denote the number of connected spanning subgraphs with i(min) edges in G. Recently, we showed in [3] the validity of (m-i+1)Ni-1>Ni for a simple graph and each i(min). Note that, from this inequality, 2 is easily derived. In this paper, for a multigraph G and all i(min), we prove (m-i+1)Ni-1(i-n+2)Ni, and give a necessary and sufficient condition by which (m-i+1)Ni-1=(i-n+2)Ni. In particular, this means that (m-i+1)Ni-1>Ni is not valid for all multigraphs, in general. Furthermore, we prove 2, which is not straightforwardly derived from (m-i+1)Ni-1(i-n+2)Ni, and also introduce a necessary and sufficent condition by which =2. Moreover, we show a sufficient condition for a multigraph to have Nn2>Nn-1Nn+1. As special cases of the sufficient condition, we show that if G contains at least +1 multiple edges between some pair of vertices, or if its underlying simple graph has no cycle with length more than 4, then Nn2>Nn-1Nn+1.

Modern applications of pattern recognition generate very large amounts of data, which require large computational effort to process. However, the majority of the methods intended for large-scale problems aim to merely adapt standard classification methods without considering if those algorithms are appropriated for large-scale problems. CombNET-II was one of the first methods specifically proposed for such kind of a task. Recently, an extension of this model, named CombNET-III, was proposed. The main modifications over the previous model was the substitution of the expert networks by Support Vectors Machines (SVM) and the development of a general probabilistic framework. Although the previous model's performance and flexibility were improved, the low accuracy of the gating network was still compromising CombNET-III's classification results. In addition, due to the use of SVM based experts, the computational complexity is higher than CombNET-II. This paper proposes a new two-layered gating network structure that reduces the compromise between number of clusters and accuracy, increasing the model's performance with only a small complexity increase. This high-accuracy gating network also enables the removal the low confidence expert networks from the decoding procedure. This, in addition to a new faster strategy for calculating multiclass SVM outputs significantly reduced the computational complexity. Experimental results of problems with large number of categories show that the proposed model outperforms the original CombNET-III, while presenting a computational complexity more than one order of magnitude smaller. Moreover, when applied to a database with a large number of samples, it outperformed all compared methods, confirming the proposed model's flexibility.

The Earth will require sustainable electricity sources equivalent to 3 to 5 times the commercial power presently produced by 2050. Solar Power Satellite (SPS) is one option for meeting the huge future energy demand. SPS can send enormous amounts of power to the Earth as the form of microwave (MW). A highly efficient microwave power transmission (MPT) system is needed for SPS. A critical goal of SPS is to maintain highest Beam Efficiency (BE) because the microwaves from SPS will be converted to utility power unlike the MW from communication satellites. Another critical goal of SPS is to maintain Side Lobe Levels (SLL) as small as possible to reduce interference to other communication systems. One way to decrease SLL and increase BE is the edge tapering of a phased array antenna. However, tapering the excitation requires a technically complicated system. Another way of achieving minimum SLL is with randomly spaced element position but it does not guarantee higher BE and the determination of random element position is also a difficult task. Isosceles Trapezoidal Distribution (ITD) edge tapered antenna was studied for SPS as an optimization between full edge tapering and uniform amplitude distribution. The highest Beam Collection Efficiency (BCE) and lowest SLL (except maximum SLL) are possible to achieve in ITD edge tapering and ITD edge tapered antenna is technically better. The performance of ITD is further improved from the perspective of both Maximum Side Lobe Level (MSLL) and BE by using unequal spacing of the antenna elements. A remarkable reduction in MSLL is achieved with ITD edge tapering with Unequal element spacing (ITDU). BE was also highest in ITDU. Determination of unequal element position for ITDU is very easy. ITDU is a newer concept that is experimented for the first time. The merits of ITDU over ITD and Gaussian edge tapering are discussed.

Energy-efficient transmission scheme is very essential for Wireless Personal Area Networks (WPNs) for maximizing the lifetime of energy-constrained wireless devices and assuring the required QoS in the actual physical transmission at each allocated TDMA time slot. We therefore propose the minimum energy (ME) criterion based adaptive transmission scheme which determines the optimum combination of transmit power, physical data rate and fragment size required to simultaneously minimize the energy consumption and satisfy the required QoS in each assigned time duration. The improved performances offered by the proposed algorithm are demonstrated via computer simulation in terms of throughput and energy consumption.

In this letter, a switched transmission technique is presented that can provide the orthogonal frequency division multiplexing (OFDM) systems with additional diversity gain. The space-time block coding (STBC) and space-frequency block coding (SFBC) are considered for the transmission of the OFDM signals. A proper coding scheme is selected based on the estimated normalized delay spread and normalized Doppler spread. The selection criterion is derived empirically. It is shown through computer simulations that the proposed switching technique can improve the bit error rate (BER) performance of an OFDM system when the corresponding wireless channel has some time variation of time selectivity as well as frequency selectivity.

We propose a novel asynchronous direct-sequence code-division multiple access (DS-CDMA) using feedback-controlled spreading sequences (FCSSs) (FCSS/DS-CDMA). At the receiver of FCSS/DS-CDMA, the code-orthogonalizing filter (COF) produces a spreading sequence, and the receiver returns the spreading sequence to the transmitter. Then the transmitter uses the spreading sequence as its updated version. The performance of FCSS/DS-CDMA is evaluated over time-dispersive channels. The results indicate that FCSS/DS-CDMA greatly suppresses both the intersymbol interference (ISI) and multiple access interference (MAI) over time-invariant channels. FCSS/DS-CDMA is applicable to the decentralized multiple access.