Feedback-type Adaptive Array Antenna has been proposed for frequency division duplexed (FDD) system, where the mobile station (MS) measures channel characteristics and sends those back to the base station (BS). Using a higher number of feed-back bits provides better performance. However it wastes channel capacity of the up-link. On the other hand, error in feedback signals transmission causes significant performance degradation. To solve these problems, this paper proposes a method that the MS sends back the difference between the optimum weights calculated at the MS and weights which are currently used at the BS. Bit error rate performance of the system is shown under a realistic propagation condition.

This paper presents a new statistical model-based voice activity detection (VAD) algorithm in the wavelet domain to improve the performance in non-stationary environments. Due to the efficient time-frequency localization and the multi-resolution characteristics of the wavelet representations, the wavelet transforms are quite suitable for processing non-stationary signals such as speech. To utilize the fact that the wavelet packet is very efficient approximation of discrete Fourier transform and has built-in de-noising capability, we first apply wavelet packet decomposition to effectively localize the energy in frequency space, use spectral subtraction, and employ matched filtering to enhance the SNR. Since the conventional wavelet-based spectral subtraction eliminates the low-power speech signal in onset and offset regions and generates musical noise, we derive an improved multi-band spectral subtraction. On the other hand, noticing that fixed threshold cannot follow fluctuations of time varying noise power and the inability to adapt to a time-varying environment severely limits the VAD performance, we propose a statistical model-based VAD algorithm in wavelet domain with an adaptive threshold. We perform extensive computer simulations and compare with the conventional algorithms to demonstrate performance improvement of the proposed algorithm under various noise environments.

In wireless local area networks (WLANs), the necessity of quality-of-service (QoS) control for uplink flows is increasing because interactive applications are becoming more popular. Fairness between flows transmitted by stations with different physical transmission rates must be ensured in QoS control for link-adaptive WLANs, which are widely used nowadays. We propose a novel distributed access scheme called QC-DCA to satisfy these requirements. QC-DCA adaptively controls the parameters of carrier sense multiple access with collision avoidance (CSMA/CA). QC-DCA has two QoS control functions: guarantee and classification. QC-DCA guarantees target throughputs and packet delays by quickly adjusting CSMA/CA parameters. In QoS classification, the difference of throughputs and packet delays between different QoS classes is maintained. These two functions allow QC-DCA to suppress the unfairness caused by differences of transmission rates in the physical layer. We evaluated the throughput and delay performances of our scheme using computer simulations. The results show the viability of our scheme.

We consider a hybrid direct-sequence frequency-hopped (DS/FH) code division multiple access (CDMA) communication system, where the transmission power, data rate (i.e. spreading gain), and hopping frequency are adapted relative to the channel variations. Instead of random frequency hopping, hopping pattern is adaptively adjusted to obtain the maximum channel gain among available frequency slots. Transmission power and/or data rate are also adapted such that a target transmission quality is maintained. It is shown that the proposed scheme provides a higher average data rate than pure DS/CDMA with power and rate adaptations, subject to the identical bandwidth and average transmission power constraints.

A new stereo echo canceler with input slides and counter-lateralization is proposed. Convergence of filter coefficients to the correct echo paths is obtained by pre-processing which delays the input signal periodically by one sample in one of the two channels. The time difference between the two stereo components of the input signals causes a shift of the sound image. This shift is compensated for by presenting the delayed component of the stereo signals to a loudspeaker at a higher intensity, and the other component at a lower intensity. Correct echo-path identification is analytically shown in a more general form than in the preceding literatures. A subjective listening test shows that this method is more effective for vocal musics. The processed signals are scored 0.45 lower than the original input signals, using the ITU-R five-grade impairment scale.

Software-Defined Radio (SDR) represents a major paradigm shift in the design of radios, allowing a large fraction of the functionality to be implemented through programmable signal processing devices, enabling the radio to change its operating parameters to accommodate new air interface, features and capabilities. However, the actual realization of innovative and software-reconfigurable receiver diversity at mobile handsets in intermediate frequency band to provide wide-ranging benefits, including more effective filtered result, less cost of the mixed channel access, improved capacity, better link reliability, and reduced power consumption, has been slowed down largely due to an absence of effective architecture reducing the complexity of adaptive combining algorithms. This paper proposes a novel reconfigurable architecture for adaptive space diversity at handsets in MC-CDMA (multicode code-division multiple-access) systems. The key to which is the development of a valid and effective alternative to the time-consuming multiplication operation and despreading acquisition. A software definable algorithm can become a multiplier-free architecture if it can restrict the weight factors to power-of-two values and repetitive gradient search procedure to contain shift operations and predicate functions. The results of numerical simulation and experimentation confirm the expectation that the constrained approach should perform comparably to, but not better than the traditional diversity algorithm. That is, the feasibility of SDR depends on its trading some performance for reduced computational complexity, improved area efficiency and less power consumption.

Affine projection algorithms perform well for acoustic echo cancellation and adaptive equalization. Although these algorithms typically provide fast convergence, they are unduly complex when updating the weights of the associated adaptive filter. In this paper, we propose a new subband affine projection (SAP) algorithm and a facile method for its implementation. The SAP algorithm is derived by combining the affine projection algorithm and the subband adaptive structure with the maximal decimation. In the proposed SAP algorithm, the derived weight-updating formula for the subband adaptive filter has a simple form as compared with the normalized least mean square (NLMS) algorithm. The algorithm gives improved convergence and reduced computational complexity. The efficiency of the proposed algorithm for a colored input signal is evaluated experimentally.

This paper deals with graph colouring games, an example of pseudo-telepathy, in which two players can convince a verifier that a graph G is c-colourable where c is less than the chromatic number of the graph. They win the game if they convince the verifier. It is known that the players cannot win if they share only classical information, but they can win in some cases by sharing entanglement. The smallest known graph where the players win in the quantum setting, but not in the classical setting, was found by Galliard, Tapp and Wolf and has 32,768 vertices. It is a connected component of the Hadamard graph GN with N=c=16. Their protocol applies only to Hadamard graphs where N is a power of 2. We propose a protocol that applies to all Hadamard graphs. Combined with a result of Frankl, this shows that the players can win on any induced subgraph of G12 having 1609 vertices, with c=12. Moreover combined with a result of Godsil and Newman, our result shows that all Hadamard graphs GN (N ≥ 12) and c=N yield pseudo-telepathy games.

Post-FFT type orthogonal frequency division multiplexing (OFDM) adaptive array antennas can reduce the co-channel interference with a few antenna elements under multi-path fading environments. However, the Post-FFT type OFDM adaptive array antennas require a lot of pilot symbols in order to determine the optimal weights in each subcarrier. In packet communication systems, since the data are transmitted burst by burst, the ratio of the effective data in a channel decreases when the long pilot symbols are used. Recursive least squares (RLS) algorithm is one of the weight optimization algorithm with fast convergence based on minimum mean square errors (MMSE). However, the optimal weight determination with a few pilot symbols is difficult. Therefore, in this paper, we propose a novel multi-stage RLS OFDM adaptive array antenna for realizing weight determination with a few pilot symbols. In the proposed method, the weights are optimized by using a multiple stage structure with the stored pilot symbols. Here, the initial weights and the initial inverse matrix of correlation matrix are decided by the results of the weight determination in the adjacent subcarriers of the previous stage. As a result, the weight determination with a few pilot symbols can be achieved.

Adaptive modulation and coding scheme (AMC) is an effective way to achieve high data rate communication. In AMC system, the key issue is to determine the rule for switching among different Modulation and Coding Schemes (MCSs). In this paper, adaptive threshold controlling scheme for AMC is proposed. The proposed scheme controls switching thresholds according to target block error rate. Simulation results have shown that the throughput performance of the proposed scheme is very close to the performance, which obtained by the optimum SIR thresholds. We also proposed downlink transmission power control (TPC) scheme suitable for AMC. The throughput of the lowest MCS is improved and the tranmission power of the highest MCS can be reduced with the proposed algorithm.

Combined input-crosspoint buffered (CICB) switches relax arbitration timing and provide high-performance switching for packet switches with high-speed ports. It has been shown that these switches, with one-cell crosspoint buffer and round-robin arbitration at input and output ports, provide 100% throughput under uniform traffic. However, under admissible traffic patterns with nonuniform distributions, only weight-based selection schemes are reported to provide high throughput. This paper proposes a round-robin based arbitration scheme for a CICB packet switch that provides 100% throughput for several admissible traffic patterns, including those with uniform and nonuniform distributions, using one-cell crosspoint buffers and no speedup. The presented scheme uses adaptable-size frames, where the frame size is determined by the traffic load.

A simultaneous equations method is one of active noise control algorithms without estimating an error path. This algorithm requires identification of a transfer function from a reference microphone to an error microphone containing the effect of a noise control filter. It is achieved by system identification of an auxiliary filter. However, the introduction of the auxiliary filter requires more number of samples to obtain the noise control filter and brings a requirement of some undesirable assumption in the multiple channel case. In this paper, a new simultaneous equations method without the identification of the auxiliary filter is proposed. By storing a small number of input signals and error signals, we avoid this identification. Therefore, we can reduce the number of samples to obtain the noise control filters and can avoid the undesirable assumption. From simulation examples, it is verified that the merits of the ordinary method is also retained in the proposed method.

A per-user AGC technique is proposed to combat the signal level variation of an individual user in a DS-CDMA receiver. A simple signal model for a Rake receiver is derived, and the potential cause of the signal variation in the Rake receiver output is discussed. The adaptive scheme is also compared with a conventional fixed quantization scheme in simulations.

The goal of this study was to design a practical fuzzy controller of the cycle-to-cycle control for multi-joint movements of swing phase of functional electrical stimulation (FES) induced gait. First, we designed three fuzzy controllers (a fixed fuzzy controller, a fuzzy controller with parameter adjustment based on the gradient descent method, and a fuzzy controller with parameter adjustment based on a fuzzy model) and two PID controllers (a fixed PID and an adaptive PID controllers) for controlling two-joint (knee and ankle) movements. Control capabilities of the designed controllers were tested in automatic generation of stimulation burst duration and in compensation of muscle fatigue through computer simulations using a musculo-skeletal model. The fuzzy controllers showed better responses than the PID controllers in the both control capabilities. The parameter adjustment based on the fuzzy model was shown to be effective when oscillating response was caused due to the inter-subject variability. Based on these results, we designed the fuzzy controller with the parameter adjustment realized using the fuzzy model for controlling three-joint (hip, knee, and ankle) movements. The controlled gait pattern obtained by computer simulation was not significantly different from the normal gait pattern and it could be qualitatively accepted in clinical FES gait control. The fuzzy controller designed for the cycle-to-cycle control for multi-joint movements during the swing phase of the FES gait was expected to be examined clinically.

The speech noise reduction system based on the frequency domain adaptive line enhancer using a windowed modified DFT (MDFT) pair is presented. The adaptive line enhancer (ALE) is effective for extracting sinusoidal signals blurred by a broadband noise. In addition, it utilizes only one microphone. Therefore, it is suitable for the realization of speech noise reduction in portable electronic devices. In the ALE, an input signal is generated by delaying a desired signal using the decorrelation parameter, which makes the noise in the input signal decorrelated with that in the desired one. In the present paper, we propose to set decorrelation parameters in the frequency domain and adjust them to optimal values according to the relationship between speech and noise. Such frequency domain decorrelation parameters enable the reduction of the computational complexity of the proposed system. Also, we introduce the window function into MDFT for suppressing spectral leakage. The performance of the proposed noise reduction system is examined through computer simulations.

An RF adaptive array antenna (RF-AAA) configured with variable capacitors is proposed. This antenna system can control the power combining ratio and phase value of received signals. In this paper, we focus on the diversity effects of RF-AAA. First, we show the design methodology of the combiner circuit to realize the effective combining. Second, the perturbation method and the steepest gradient method are compared for the optimization algorithms to provide fast convergence and suboptimum solutions among the variable circuit constants. Finally, in simulation, we show the RF-AAA can achieve diversity antenna gains of 7.7 dB, 10.9 dB and 12.6 dB for 2-branch, 3-branch and 4-branch configuration, respectively, which have higher performance than the selection combining.

This paper examines the bit and power allocation problem for orthogonal frequency division multiplexing systems in which the overall transmission power is minimized by constraining the fixed data rate and bit error rate. To provide the optimal allocation with less computational complexity, we propose new bit and power allocation schemes based on the Lagrangian method. Firstly, we propose an initial search range of the bisection search method to find the optimal Lagrangian multiplier efficiently. The simulation results verify that the proposed initial search range guarantees the optimal solution with less computational complexity. Secondly, a new iterative search method for the optimal Lagrangian multiplier is proposed using Newton's search method. The simulation results demonstrate that the proposed scheme has significant computational advantages over the conventional algorithms while providing optimal performance.

We have proposed two space division multiple access (SDMA) approaches for OFDM signals: "Virtual Subcarrier Assignment (VISA)" and "Preamble Subcarrier Assignment (PASA)," both of which can enhance the system capacity without significant change of transmitter/receiver structures for already-existing OFDM-based standards such as IEEE802.11a. In order to investigate the performance of the proposed approaches in real wireless scenarios, we conducted a measurement campaign to obtain real channel state data at 5-GHz band in an indoor environment. Using the measured channel data, we can make the performance evaluation realistic. In this paper, after the brief overview of the two proposed SDMA approaches, we describe our measurement campaign in detail. Furthermore, we evaluate the performance of VISA-based system and PASA-based system by computer simulations using the measured channel state data and present a comparative study on the performance of the two proposed SDMA approaches in the realistic wireless environment.

In this letter, a new formula is proposed for calculating the polarization entropy, based on the least square method. There is no need to calculate the eigenvalues of a covariance matrix as well as to use logarithms of values. So the time for computing the polarization entropy is reduced. Using polarimetric SAR data, the authors validate the effectiveness of the new formula.

This paper proposes a technique for synthesizing speech with a desired speaking style and/or emotional expression, based on model adaptation in an HMM-based speech synthesis framework. Speaking styles and emotional expressions are characterized by many segmental and suprasegmental features in both spectral and prosodic features. Therefore, it is essential to take account of these features in the model adaptation. The proposed technique called style adaptation, deals with this issue. Firstly, the maximum likelihood linear regression (MLLR) algorithm, based on a framework of hidden semi-Markov model (HSMM) is presented to provide a mathematically rigorous and robust adaptation of state duration and to adapt both the spectral and prosodic features. Then, a novel tying method for the regression matrices of the MLLR algorithm is also presented to allow the incorporation of both the segmental and suprasegmental speech features into the style adaptation. The proposed tying method uses regression class trees with contextual information. From the results of several subjective tests, we show that these techniques can perform style adaptation while maintaining naturalness of the synthetic speech.