This work proposes new features to improve the pathological voice quality classification performance. They are the means, the variances, and the perturbations of the higher-order statistics (HOS) such as the skewness and the kurtosis. The HOS-based features show meaningful differences among normal, grade 1, grade 2, and grade 3 voices classified in the GRBAS scale. The jitter, the shimmer, the harmonic-to-noise ratio (HNR), and the variance of the short-time energy are utilized as the conventional features. The performances are measured by the classification and regression tree (CART) method. Specifically, the CART-based method by utilizing both the conventional features and the HOS-based ones shows its effectiveness in the pathological voice quality measurement, with the classification accuracy of 87.8%.

The complexity of Finite Impulse Response (FIR) filters is mainly dominated by the number of adders (subtractors) used to implement the coefficient multipliers. It is well known that Common Subexpression Elimination (CSE) method based on Canonic Signed Digit (CSD) representation considerably reduces the number of adders in coefficient multipliers. Recently, a binary-based CSE (BSE) technique was proposed, which produced better reduction of adders compared to the CSD-based CSE. In this paper, we propose a new 4-bit binary representation-based CSE (BCSE-4) method which employs 4-bit Common Subexpressions (CSs) for implementing higher order low-power FIR filters. The proposed BCSE-4 offers better reduction of adders by eliminating the redundant 4-bit CSs that exist in the binary representation of filter coefficients. The reduction of adders is achieved with a small increase in critical path length of filter coefficient multipliers. Design examples show that our BCSE-4 gives an average power consumption reduction of 5.2% and 6.1% over the best known CSE method (BSE, NR-SCSE) respectively, when synthesized with TSMC-0.18 µm technology. We show that our BCSE-4 offers an overall adder reduction of 6.5% compared to BSE without any increase in critical path length of filter coefficient multipliers.

In this paper, we propose a method for selecting n-mode singular vectors in higher-order singular value decomposition. We select the minimum number of n-mode singular vectors, when the upper bound of a least-squares cost function is thresholded. The reduced n-ranks of all modes of a given tensor are determined automatically and the tensor is represented with the minimum number of dimensions. We apply the selection method to simultaneous low rank approximation of matrices. Experimental results show the effectiveness of the n-mode singular vector selection method.

Transmit diversity systems based on orthogonal space-time block coding (OSTBC) usually suffer from rate loss and power spreading. Proper antenna selection scheme can help to more effectively utilize the transmit antennas and transmission power in such systems. In this paper, we propose a new antenna selection scheme for such systems based on the idea of antenna switching. In particular, targeting at reducing the number of pilot channels and RF chains, the transmitter now replaces the antennas with the lowest received SNR with unused ones if the output SNR of space time decoder at the receiver is below a certain threshold. With this new scheme, not only the number of pilot channels and RF chains to be implemented is decreased, the average amount of feedback information is also reduced. To analyze the performance of this scheme, we derive the exact integral closed form for the probability density function (PDF) of the received SNR. We show through numerical examples that the proposed scheme offers better performance than traditional OSTBC systems using all available transmitting antennas, with a small amount of feedback information. We also examine the effect of different antenna configuration and feedback delay.

This paper describes the matrix order reduction method by the nodal analysis formulation and the application of relaxation-based simulation technique to interconnect and plane networks. First, the characteristics of the power/ground plane networks are considered. Next, the formulation of the plane network by nodal analysis (NA) method is suggested. Furthermore, application and estimation results of the relaxation-based numerical analyses are shown. Finally, it is confirmed that the relaxation-based methods improved by the suggested formulation are much more efficient than the conventional direct-based methods.

Modern microprocessors achieve high application performance at the acceptable level of power dissipation. In terms of power to performance trade-off, the instruction window is particularly important. This is because enlarging the window size achieves high performance but naive scaling of the conventional instruction window can severely increase the complexity and power consumption. In this paper, we propose low-power instruction window techniques for contemporary microprocessors. First, the small reorder buffer (SROB) reduces power dissipation by deferred allocation and early release. The deferred allocation delays the SROB allocation of instructions until their all data dependencies are resolved. Then, the instructions are executed in program order and they are released faster from the SROB. This results in higher resource utilization and low power consumption. Second, we replace a conventional issue queue by a direct lookup table (DLT) with an efficient tag translation technique. The translation scheme resolves the instruction dependency, especially for the case of one producer to multiple consumers. The efficiency of the translation scheme stems from the fact that the vast majority of instruction dependency exists within a basic block. Experimental results show that our proposed design reduces the power consumption significantly for SPEC2000 benchmarks.

The passive and sparse reduced-order modeling of a RLC network is presented, where eigenvalues and eigenvectors of the original network are used, and thus the obtained macromodel is more accurate than that provided by the Krylov subspace methods or TBR procedures for a class of circuits. Furthermore, the proposed method is applied to low pass filtering of a reduced-order model produced by these methods without breaking the passivity condition. Therefore, the proposed eigenspace method is not only a reduced-order macromodeling method, but also is embedded in other methods enhancing their performances.

In this letter, we propose a simple detection ordering which can be applied to successive interference cancellation in low-complexity equalization of OFDM in doubly selective channels. Since the detection ordering can be determined solely on the basis of pilot symbol arrangement, the proposed scheme can lead to a significant performance improvement without additional complexity on the premise that pilot symbols are employed primarily for channel estimation.

The H.264/AVC standard provides several new error-resilient features to enable the reliable transmission of compressed video signals over lossy packet networks. Flexible Macroblock Ordering (FMO) is one of the most interesting resilient features within the H.264/AVC standard. Unlike former standards, in which slices were constructed out of consecutive raster scan macroblocks, FMO suggests new slices composed of spatially distributed Macroblocks (MBs), and organized in a mixed-up fashion. H.264/AVC specifies seven types of FMO. The standard defines also an explicit FMO type (Type 6), which allows explicitly assignment of each MB within the frame to any available slice groups. Therefore new FMO types can be used and integrated into H264/AVC without violating the standard. In this paper we propose a new Explicit Chessboard-Wipe (ECW) Flexible Macroblocks Ordering (FMO) technique, which outperforms all other FMO types. The new ECW ordering results in effective error scattering which maximizes the number of correctly received macroblocks located around corrupted macroblocks, leading to better error concealment. Performance evaluations demonstrate that the proposed Explicit FMO approach outperforms all the FMO types. Both subjective and objective visual quality comparative study has been also carried out in order to validate the proposed approach.

In this research, the practical OTA-based inductors of all structures have been studied and their complete passive equivalent circuit models, where the effects of both parasitic elements and finite opened-loop bandwidth have been taken into account, also contain only the conventional standard linear elements i.e. the ordinary resistor, inductor and capacitor, without any infeasible high order element e.g. super inductor etc., have been proposed. The resulting models have been found to be excellently accurate, excellently straight forward, far superior to the previously proposed ones and completely realizable by the passive elements. Hence, the proposed passive equivalent circuit models have been found to be the convenience and versatile tools for the implementation of any analog and mixed signal processing circuits and systems.

Simulations are often deployed to evaluate proposed mechanisms or algorithms in Mobile Ad Hoc Networks (MANET). In MANET, the impacts of some simulation parameters are noticeable, such as transmission range, data rate etc. However, the effect of mobility model is not clear until recently. Random Waypoint (RWP) is one of the most applied nodal mobility models in many simulations due to its clear procedures and easy employments. However, it exhibits the two major problems: decaying average speed and border effect. Both problems will overestimate the performance of the employed protocols and applications. Although many recently proposed mobility models are able to reduce or eliminate the above-mentioned problems, the concept of Diverse Average Speed (DAS) has not been introduced. DAS aims to provide different average speeds within the same speed range. In most mobility models, the average speed is decided when the minimum and maximum speeds are set. In this paper, we propose a novel mobility model, named General Ripple Mobility Model (GRMM). GRMM targets to provide a uniform nodal spatial distribution and DAS without decaying average speed. The simulations and analytic results have demonstrated the merits of the outstanding properties of the GRMM model.

A low distortion and low noise differential amplifier using the difference between the even- and odd-mode impedances is proposed. In order to realize an amplifier with high OIP3 and low NF characteristics, the impedance of the bias circuit should be low (<300 Ω) at the difference frequency and high (>4 kΩ) at the carrier frequency. Although the frequency response of the bias circuit impedance can only meet these conditions with difficulty, owing to the 20 MHz Tx signal bandwidth for 3G LTE, the proposed amplifier can achieve the impedance difference using the properties of a differential configuration where the difference frequency signal is the even-mode and the carrier frequency is the odd-mode. It has been demonstrated that the NF of the proposed amplifier, which has been fabricated in 0.18 µm SiGe BiCMOS technology operating at 2.14 GHz, can be kept to 1.6 dB or less and an OIP3 of 9.0 dBm can be achieved, which is 3 dB higher than that of a conventional amplifier, in the condition where the power gain is greater than 18 dB.

In the design of Space Time Block Coding (STBC), for an arbitrary complex signal constellation with a size above 2 as well as a real signal matrix with a size above 8, it is difficult to acquire full code rate and full transmit diversity simultaneously. In this letter, an efficient selective receiver switching scheme is proposed for STBC with the full code rate and non-orthogonal design with the example of a 4-by-4 matrix. In the proposed scheme with the aid of beamforming, we divide the received signals into two groups according to the encoded matrix. By this way, we can eliminate the interference from the neighboring signals by more than half.

We examined how spatial disparity between the auditory and visual stimuli modulated the audio-visual (A-V) prior entry effect. Spatial and temporal proximity of multisensory stimuli are crucial factors for multisensory perception in most cases (e.g. [1],[2]). However our previous research[3],[4] suggested that this well-accepted hypothesis was not applicable to the A-V prior entry effect. In order to examine the effect of the spatial disparity on the A-V prior entry effect, six loudspeakers and two light emitting diodes (LEDs) were used as stimuli. The loudspeakers were located at 10, 25, and 90 degrees from the midline of the participants to both right and left sides. A preceding sound was presented from one of these six loudspeakers. After the preceding sound, two visual targets were presented successively at a short interval and participants judged which visual target was presented first. Two colour changeable ('red' or 'green') LEDs were used for the visual targets and participants judged the order of visual targets by their colour not by their side in order to avoid the response bias as much as possible. The visual targets were situated at 10 degrees or 25 degrees from the participants' midline to both right and left in the Experiment 1. Results showed a biased judgment that the visual target at the sound presented side was presented first. The amplitude of the A-V prior entry effect was greater when the preceding sound source was more apart from the midline of participants. This effect of spatial separation indicated that the clarity of either right or left side of the preceding sound enhanced the amplitude of the A-V prior entry effect (Experiment 2). These results challenge the belief that the spatial proximity of multisensory stimuli is a crucial factor for multisensory perception.

In this paper, we present a multiple capture approach to reducing the peak power as well as average power consumption during testing. The basic idea behind is to divide a scan chain into two sub-scan chains, and only one sub-scan chain will be enabled at a time during the scan shift or capture operations. We develop a pattern insertion technique to efficiently deal with the capture violation problem during the capture cycle. In order to alleviate the timing cost due to the insertion of redundant patterns, a scan chain partitioning method incorporated with test pattern reordering is developed to reduce the testing time. Experimental results for large ISCAS'89 benchmark circuits show that the proposed approach can efficiently reduce peak and average power with little timing overhead.

A novel design for temperature-compensated complementary metal-oxide semiconductor (CMOS) voltage reference sources by using the 1st order voltage reference taking into account the electrical property of the conventional current generator was proposed to minimize a temperature coefficient. A temperature coefficient of the proposed voltage reference source was estimated by using the current generator, which operated at smaller or larger temperature in comparison with the optimized operating temperature. The temperature coefficient at temperature range between -40 and 125, obtained from the simulated data by using hynix 0.35 µm CMOS technology, was 3.33 ppm/. The simulated results indicate that the proposed temperature-compensated CMOS voltage reference sources by using the 1st order voltage reference taking into account the electrical properties of the conventional current generator can be used to decrease the temperature coefficient.

We propose a fast decoding algorithm for the p-ary first-order Reed-Muller code guaranteeing correction of up to errors and having complexity proportional to nlog n, where n = pm is the code length and p is an odd prime. This algorithm is an extension in the complex domain of the fast Hadamard transform decoding algorithm applicable to the binary case.

This study deals with two linear precoders: the maximization of the minimum Euclidean distance between received symbol-vectors, called here max-dmin, and the maximization of the post-processing signal-to-noise ratio termed max-SNR or beamforming. Both have been designed for reliable MIMO transmissions operating over uncorrelated Rayleigh fading channels. Here, we will explain why performances in terms of bit error rates show a significant enhancement of the max-dmin over the max-SNR whenever the number of antennas is increased. Then, from theoretical developments, we will demonstrate that, like the max-SNR precoder, the max-dmin precoder achieves the maximum diversity order, which is warrant of reliable transmissions. The current theoretical knowledge will be applied to the case-study of a system with two transmit- or two receive-antennas to calculate the probability density functions of two channel parameters directly linked to precoder performances for uncorrelated Rayleigh fading channels. At last, this calculation will allow us to quickly get the BER of the max-dmin precoder further to the derivation of a tight semi-theoretical approximation.

Meeting the tough linearity and noise required by GSM and UMTS receivers in CMOS technology is challenging. A new IIP2 calibration technique based on canceling the second order nonlinearities of mixer, generated in the input RF transistors, is introduced. By using this technique about 22 dB mixer IIP2 improvement is achieved. The proposed calibration circuit can be used in multi-standard mixer because of high bandwidth of the calibration circuitry. Moreover it can work with voltage supplies as low as 1 V. Using this technique a multi-standard mixer supporting PCS, UMTS and IEEE802.11b-g is developed. The design is done in CMOS 65 nm technology with 1.2 V supply while it consumes about 7 mA current.

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.