In this paper, a new methodology to estimate the number of competing stations in an IEEE 802.11 network, is proposed. Due to the nonlinear nature of the measurement model, an iterative nonlinear filtering algorithm, called the Scaled Unscented Filter (SUF), is employed. The SUF can provide a superior alternative to nonlinear filtering than the conventional Extended Kalman Filter (EKF), since it avoids errors associated with linearization. This approach demonstrates both high accuracy in addition to prompt reactivity to changes in the network occupancy status. In particular, the proposed algorithm shows superior performance in non saturated conditions when compared to the EKF. Numerical results demonstrate that the proposed algorithm provides a more viable method for estimation of the number of competing stations in an IEEE 802.11 network, than estimators based on the EKF.

Three miniaturized lumped-element power dividers with a filtering function for use in quadrature mixers are described. Simulation results showed that they can be miniaturized, as compared to conventional ones with open/short stubs, while maintaining the filter characteristics. A fabricated 0.95-GHz 0power divider with a filtering function had a chip size about half that of a conventional lumped-element one. Its insertion loss at 0.950.05 GHz was 4.00.1 dB.

In this article, we propose a vehicle positioning method that can estimate positions of cars even in areas where the GPS is not available. For the estimation, each car measures the relative distance to a car running in front, communicates the measurements with other cars, and uses the received measurements for estimating its position. In order to estimate the position even if the measurements are received with time-delay, we employed the time-delay tolerant Kalman filtering. For sharing the measurements, it is assumed that a car-to-car communication system is used. Then, the measurements sent from farther cars are received with larger time-delay. It follows that the accuracy of the estimates of farther cars become worse. Hence, the proposed method manages only the states of nearby cars to reduce computing effort. The authors simulated the proposed filtering method and found that the proposed method estimates the positions of nearby cars as accurate as the distributed Kalman filtering.

RFID event filtering is an important issue of RFID data management. Tag read events from readers have some problems like unreliability, redundancy, and disordering of tag readings. Duplicated events lead to performance degradation of RFID systems with a flood of similar tag information. Therefore, this paper proposes a fuzzy logic-based quantized event filter. In order to reduce duplicated tag readings and solve disordering of tag readings, the filter applies a fuzzy logic system to control a filtering threshold by the change in circumstances of readers. Continuous tag readings are converted into discrete values for event generation by the filtering threshold. And, the filter generates as many events as the discrete values at a point of event generation time. Experimental results comparing the proposed filter with existing RFID event filters, such as the primitive event filter and the smoothing event filter, verify effectiveness and efficiency of the fuzzy logic-based quantized event filter.

To improve the performance of the optimal pilot sequences over multiple OFDM symbols in fast time-varying channels, this letter proposes a novel channel estimation method using virtual pilot tones in multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) systems. Assuming that the superimposed virtual pilot tones at the data locations over the specific sub-carriers are transmitted from all transmit antennas, the corresponding virtual received pilot signals at the same locations are obtained from the neighboring real received pilot signals over the same sub-carriers by Wiener filter. Based on the least squares (LS) channel estimation, the channel parameters can be obtained from the combination of the virtual and real received pilot signals over one OFDM symbol. Simulation results show that the proposed channel estimation method greatly outperforms the previous method for the optimal pilot sequences over multiple OFDM symbols in fast time-varying channels, as well as approaches the method for the comb-type optimal pilot sequences in performance.

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.

To realize the compact ultra-wideband (UWB) bandpass filters, a novel filter prototype with two short-circuited stubs loaded at both sides of a stepped-impedance resonator (SIR) via the parallel coupled lines is proposed based on a distributed filter synthesis theory. The equivalent circuit of this filter is established, while the corresponding 7-pole Chebyshev-type transfer function is derived for filter synthesis. Then, a distributed-circuit-based technique was presented to synthesize the elements' values of this filter. As an example, a FCC UWB filter with the fractional bandwidth (FWB) @ -10 dB up to 110% was designed using the proposed prototype and then re-modeled by commercial microwave circuit simulator to verify the correctness and accuracy of the synthesis theory. Furthermore, in terms of EM simulator, the filter was further-optimized and experimentally-realized by using microstrip line. Good agreements between the measurement results and theoretical ones validate the effectiveness of our technique. In addition, compared with the conventional SIR-type UWB filter without short-circuited stubs, the new one significantly improves the selectivity and out-of-band characteristics (especially in lower one -45 dB@1-2 GHz) to satisfy the FCC's spectrum mask. The designed filter also exhibits very compact size, quite low insertion loss, steep skirts, flat group delay and the easily-fabricatable structure (the coupling gap dimension in this filter is 0.15 mm) as well. Moreover, it should be noted that, in terms of the presented design technique, the proposed filter prototype can be also used to easily realize the UWB filters with other FBW even greater than 110%.

A highly integrated frequency quadrupler MMIC that uses three-dimensional MMIC (3D-MMIC) technology is presented. It consists of four driver amplifiers, two doublers, and a 2-band elimination filter. These seven circuits are integrated in only a 2.36 mm2 area. The filter sufficiently suppresses spurious output components. The third and fifth harmonic components, which are the spurious components nearest to the desired component, are well suppressed. The desired/undesired ratio is about 40 dB. The driver amplifiers make the quadrupler output a constant power of the desired multiplied signal under low input power. The MMIC supplies +5 dBm of the fourth harmonic component in the input power range from -10 dBm to +5 dBm. The power dissipation of the MMIC is only 160 mW.

A new elliptic-function bandpass filter (BPF) is proposed, which utilizes an inter-digital coupled line (IDCPL) as a left-handed transmission line. The IDCPL is employed in order to realize a negative coupling between non-adjacent resonators in a wideband BPF. As the authors' knowledge, the left-handed operations of the IDCPL has rarely utilized before, although the IDCPL itself has been widely used in many microwave circuits without being paid attention to the left-handed operations. Measured characteristics of two BPFs are presented in this paper, one is targeted for 3-4 GHz WiMAX systems, and the other is for 3-5 GHz ultra wideband communication systems (UWB).

Rugate thin film optical filters are useful for designing arbitrary-shaped spectra, such as multistep or triangular spectra. A technique for synthesizing the refractive index distribution of rugate filters was used to suppress unwanted ripples on the spectrum. The refractive index of an amorphous hydrogenated silicon oxide (a-SiOx:H) rugate thin film was minutely controlled with a resolution of 0.001 using radio-frequency (RF) magnetron sputtering. The fabricated rugate filters had multistep bands over a wavelength range of 1260-1670 nm or good linearity over 1290-1650 nm.

In this paper, a new formulation of equal-length three-section open stubs having two zeros located on the unit circle and one zero at z=-1 (θ=π) in the Z-plane is presented. In particular, new filter configurations consisting of equal-length two-section open stubs, cascade lines, open stubs, and three-section open stubs are employed to emulate the discrete-time filters. To examine the validity of our formulation, we realized two discrete-time Chebyshev type II low-pass filters in the form of microstrip lines. The frequency responses of these two filters are measured to validate this new formulation.

In this paper, we present and analyze a predictive closed-loop power control (CLPC) scheme which employs a comb-type sample arrangement to effectively compensate multiple power control group (PCG) delays over mobile fading channels. We consider both least squares and recursive least squares filters in our CLPC scheme. The effects of channel estimation error, prediction filter error, and power control bit transmission error on the performance of the proposed CLPC method along with competing non-predictive and predictive CLPC schemes are thoroughly investigated. Our results clearly indicate the superiority of the proposed scheme with its improved robustness under non-ideal conditions. Furthermore, we carry out a Monte-Carlo simulation study of a 55 square grid cellular network and evaluate the user capacity. Capacity improvements up to 90% are observed for a typical cellular network scenario.

A residual acoustic echo cancellation method that employs the masking property is proposed to enhance the speech quality of hands-free communication devices in an automobile environment. The conventional masking property is employed for speech enhancement using the masking threshold of the desired clean speech signal. In this Letter, either the near-end speech or residual noise is selected as the desired signal according to the double-talk detector. Then, the residual echo signal is masked by the desired signal (masker). Experiments confirm the effectiveness of the proposed method by deriving the echo return loss enhancement and by examining speech waveforms and spectrograms.

In this paper, we propose a method that uses Simulated Annealing (SA) to estimate the linear and nonlinear parameters of a closed-box loudspeaker system for implementing effective Mirror filters. The nonlinear parameters determined by W. Klippel's method are sometimes inaccurate and imaginary. In contrast, the proposed method can estimate the parameters with satisfactory accuracy due to its use of SA; the resulting impedance and displacement characteristics match those of an actual equivalent loudspeaker. A Mirror filter designed around these parameters can well compensate the nonlinear distortions of the loudspeaker system. Experiments demonstrate that the method can reduce the levels of nonlinear distortion by 5 dB to 20 dB compared to the before compensation condition.

An indirectly reactive sputtering coater has been developed to deposit various high quality metallic and metal oxide films at high deposition rate. In this letter, several kinds of filters such as antireflection (AR) coating, IR-cut filter, and Rugate filter were deposited for the benchmark test of implemental capabilities. Our coater was established to be a powerful tool for both discrete multilayer and Rugate filters due to high stability and reproducibility of the refractive index and the deposition rate.

A novel electronically tunable high input impedance voltage-mode multifunction filter with single inputs and three outputs employing two single-output-operational transconductance amplifiers, one differential difference current conveyor and two capacitors is proposed. The presented filter can be realized the highpass, bandpass and lowpass functions, simultaneously. The input of the filter exhibits high input impedance so that the synthesized filter can be cascaded without additional buffers. The circuit needs no any external resistors and employs two grounded capacitors, which is suitable for integrated circuit implementation.

Finite impulse response (FIR) filtering is the most computationally intensive operation in the channelizer of a wireless communication receiver. Higher order FIR channel filters are needed in the channelizer to meet the stringent adjacent channel attenuation specifications of wireless communications standards. The computational cost of FIR filters is dominated by the complexity of the coefficient multipliers. Even though many methods for reducing the complexity of filter multipliers have been proposed in literature, these works focused on lower order filters. This paper presents a coefficient-partitioning-based binary subexpression elimination method for realizing low power FIR filters. We show that the FIR filters implemented using proposed method consume less power and achieve speed improvement compared to existing filter implementations. Design examples of the channel filters employed in the Digital Advanced Mobile Phone System (D-AMPS) and Personal Digital Cellular (PDC) receivers show that the proposed method achieved 23% average reductions of full adder and power consumption and 23.3% reduction of delay over the best existing method. Synthesis results show that the proposed method offers average area reduction of 8% and power reduction of 22% over the best known method in literature.

In this letter, a genetic programming method is used to synthesize filters. In order to improve the group delay characteristics, we propose a novel two-stage fitness function reflecting not only the frequency response but also the group delay characteristics of the evolved filters. We also deal with two practical design considerations, i.e., the filters include parasitic effects and are composed of elements with discrete values. The proposed method is applied to low-pass filter design cases. The experimental results show the method can effectively generate filters satisfying the design considerations and possessing improved group delay characteristics when compared with traditional filters.

This paper is about an efficient implementation of adaptive filtering for echo cancelers. The first objective of this paper is to propose a simplified method of the flexible block Multi-Delay Filter (MDF) algorithm in the time-domain. Then, we will derive a new method for the step-size adaptation coefficient. The second objective is about the realization of a Block Proportionate Normalized Least Mean Squares (BPNLMS++) with the simplified MDF (SMDF) implementation. Using the new step-size method and the smaller block dimension proposed by SMDF, we achieve a faster convergence of the adaptive process with a limited computational cost. Then, an efficient implementation of the new procedure (SMDF-BPNLMS++) block filtering is proposed using Fermat Number Transform, which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor.

Spectral subtraction is commonly used for speech enhancement in a single channel system because of the simplicity of its implementation. However, this algorithm introduces perceptually musical noise while suppressing the background noise. We propose a wavelet-based approach in this paper for suppressing the background noise for speech enhancement in a single channel system. The wavelet packet transform, which emulates the human auditory system, is used to decompose the noisy signal into critical bands. Wavelet thresholding is then temporally adjusted with the noise power by time-adapted noise estimation. The proposed algorithm can efficiently suppress the noise while reducing speech distortion. Experimental results, including several objective measurements, show that the proposed wavelet-based algorithm outperforms spectral subtraction and other wavelet-based denoising approaches for speech enhancement for nonstationary noise environments.