A hybrid noise canceller with a real‐time adaptive Wiener filter and a geometric‐based voice–activity detector for an automotive application

In real‐time environments a speech recognition system in a car has to receive the driver's voice only while suppressing the background noise. This paper presents a hybrid real‐time adaptive filter that operates within a geometrical zone defined around the head of the desired speaker. Any sound outside of this zone is considered to be noise and is suppressed. As this defined geometrical zone is small, it is assumed that only driver's speech is incoming from this zone. The technique uses three microphones to define a geometric‐based voice–activity detector (VAD) to cancel the unwanted speech coming from outside of the zone. However, when unwanted speech and desired speech are incoming at the same time, the VAD fails to identify the unwanted speech or desired speech. In such a situation an adaptive Wiener filter is switched on for noise reduction. In the case of sole unwanted speech incoming from outside of a desired zone, this speech is muted at the output of the hybrid noise canceller. In the case of desired and unwanted speeches incoming together, the SNR is improved by as much as 20 dB. Copyright © 2009 John Wiley & Sons, Ltd.     

Bottom‐up performance analysis of focal‐plane mixed‐signal hardware for Viola–Jones early vision tasks

Focal‐plane mixed‐signal arrays have traditionally been designed according to the general claim that moderate accuracy in processing is affordable. The performance of their circuitry has been analyzed in these terms without a comprehensive study of the ultimate consequences of such moderate accuracy. In this paper, for the first time to the best of our knowledge, we do carry out this study. We move expectable performance of mixed‐signal image processing hardware directly into the vision algorithm making use of it. This permits to close a wider design loop, enabling a more aggressive design of this kind of hardware provided that the algorithm, at the highest level—semantic interpretation of the scene—, can afford it. Thus, we present a thorough analysis of the non‐idealities associated with the implementation of a QVGA array tailored for the distinctive characteristics of the Viola–Jones processing framework. The resulting deviation models are then introduced in the processing flow of this framework provided by the OpenCV library. We have found, contrary to what could be expected, that these deviations do not necessarily degrade the performance of the Viola–Jones algorithm. They could be even beneficial for certain high‐level specifications. Additionally, we demonstrate the architectural advantages of our approach: exploitation of focal‐plane distributed memory and ultra‐low‐power operation. Copyright © 2014 John Wiley & Sons, Ltd.     

FDCCII‐based electronically tunable voltage‐mode biquad filter

In this work, a voltage‐mode biquad filter realizing low‐pass, band‐pass and high‐pass characteristics is presented. The proposed filter, which employs two FDCCIIs, two grounded capacitors and two NMOS transistors, provides electronic tunability with the control voltage applied to the gate. NMOS transistors act as linear resistor. Furthermore, the proposed circuit still enjoys realization using a low number of active and passive components, no requirement with the component choice conditions to realize specific filtering functions, high input impedance, and low active and passive sensitivities performance. Simulation results using SPICE program are given to show the performance of the filter and verify the theory. Copyright © 2010 John Wiley & Sons, Ltd.     

Unsupervised adaptation of electroencephalogram signal processing based on fuzzy C‐means algorithm

This paper studies an unsupervised approach for online adaptation of electroencephalogram (EEG) based brain–computer interface (BCI). The approach is based on the fuzzy C‐means (FCM) algorithm. It can be used to improve the adaptability of BCIs to the change in brain states by online updating the linear discriminant analysis classifier. In order to evaluate the performance of the proposed approach, we applied it to a set of simulation data and compared with other unsupervised adaptation algorithms. The results show that the FCM‐based algorithm can achieve a desirable capability in adapting to changes and discovering class information from unlabeled data. The algorithm has also been tested by the real EEG data recorded in experiments in our laboratory and the data from other sources (set IIb of the BCI Competition IV). The results of real data are consistent with that of simulation data. Copyright © 2011 John Wiley & Sons, Ltd.     

A single‐bit continuous‐time delta‐sigma modulator using clock‐jitter and inter‐symbol‐interference suppression technique

This paper presents a technique for mitigating two well‐known DAC non‐idealities in continuous‐time delta‐sigma modulators (CTDSMs), particularly in wide‐band and low over‐sampling‐ratio (OSR) cases. This technique employs a special digital‐to‐analog convertor (DAC) waveform, called modified return‐to‐zero (MRZ), to reduce the time uncertainty effect because of the jittered clock at the sampling time instances and eliminate the effect of inter‐symbol‐interference (ISI) which degrades the modulator performance, especially when non‐return‐to‐zero (NRZ) DAC waveform is chosen in the modulator design. A third‐order single‐bit CTDSM is designed based on the proposed technique and step‐by‐step design procedure at circuit and system levels, considering clock jitter and ISI, is explained. Circuit simulations in 180‐nm CMOS technology show that in the presence of circuit non‐idealities which generate jitter and asymmetrical rise and fall times in the DAC current pulse, signal‐to‐noise‐distortion‐ratio (SNDR) of the proposed modulator is higher than the conventional modulator with NRZ waveform by about 10 dB. In these simulations, clock jitter standard deviation is 0.3% of the sampling period (T_S) and the difference between fall/rise times in the DAC current pulse is 4%T_S. Simulated at 600‐MHz sampling frequency (f_S) with an oversampling ratio (OSR) of 24, SNDR figure of merit (FOM_SNDR) of the proposed modulator in 180‐nm CMOS is 300 fj/conversion. Copyright © 2016 John Wiley & Sons, Ltd.     

Multi‐command SSAEP‐based BCI system with training sessions for SSVEP during an eye fatigue state

This paper proposes a steady‐state auditory stimulus modality and a detection algorithm to replace steady‐state visual evoked potential (SSVEP )‐based brain–computer interface (BCI ) systems during visual fatigue periods. The optimal speaker position for the steady‐state auditory evoked potential (SSAEP )‐based BCI system and possible electrode positions are investigated. Using the proposed system, an accuracy of 85% for two commands was achieved based on the T3–T5 and T4–T6 electrode positions using only one speaker. SSAEP is a promising BCI modality for mitigating the problem of eye fatigue that often occurs during the use of SSVEP ‐based BCI systems. However, SSAEP ‐based BCI systems suffer from low accuracy. To increase accuracy, we propose a new enhanced SSAEP training method. The training process was enhanced by instructing users to control their attention levels while simultaneously detecting an auditory stimulus frequency. Furthermore, we propose a corresponding single‐frequency, multi‐command BCI paradigm for the proposed training method. With the proposed paradigm, four commands can be detected using only one auditory stimulus frequency. The proposed training system yielded an accuracy of ∼81% compared to 66% for sessions performed without the proposed training method. © 2017 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

Design of a near‐optimal adaptive filter in digital signal processor for active noise control

Adaptive filter has been applied in adaptive feedback and feedforward control systems, where the filter dimension is often determined by trial‐and‐error. The controller design based on a near‐optimal adaptive filter in digital signal processor (DSP) is developed in this paper for real‐time applications. The design integrates the adaptive filter and the experimental design such that their advantages in stability and robustness can be combined. The near‐optimal set of controller parameters, including the sampling rate, the dimension of system identification model, the dimension (order) of adaptive controller in the form of an FIR filter, and the convergence rate of adaptation is shown to achieve the best possible system performance. In addition, the sensitivity of each design parameter can be determined by analysis of means and analysis of variance. Effectiveness of the adaptive controller on a DSP is validated by an active noise control experiment. Copyright © 2007 John Wiley & Sons, Ltd.     

Enhancement of steady‐state visual evoked potential‐based brain–computer interface systems via a steady‐state motion visual stimulus modality

Steady‐state visual evoked potential (SSVEP)‐based brain–computer interface (BCI) systems are among the most accurate assistive devices for patients with severe disabilities. However, existing visual stimulation patterns lead to eye fatigue, which affects the system performance. Therefore, in this study, we propose two improvements to SSVEP‐based BCI systems. First, we propose a novel visual stimulator that incorporates a visual motion stimulus for the steady‐state visual stimulus to reduce eye fatigue while maintaining the advantages associated with SSVEPs. We also propose two corresponding feature extraction algorithms, i.e. SSVEP detection and visual attention detection, to capture the phenomena of steady‐state motion visual stimulus responses. The accuracy of the test was ∼83.6%. Second, we propose a novel hybrid BCI‐SSVEP system and a motion visual stimulus hybrid BCI system to enhance the SSVEP‐based BCI system during a state of eye fatigue. Participants used the SSVEP system until reaching a fatigued state and then began using a hybrid motion visual stimulus. The accuracy of the proposed system was ∼85.6%. The proposed improvements can be incorporated into practical BCI systems for wheelchair control. © 2017 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

Modeling and real‐time simulation of UPFC

We have developed a digital real‐time simulator of Power Electronics Controllers, so‐called FACTS (Flexible AC Transmission Systems) Controllers and/or Custom Power by using MATLAB^TM/SIMULINK^TM and dSPACE^TM System. This paper describes the modeling and the calculation accuracy of a UPFC (Unified Power Flow Controller) model. Hence, the developed simulator operates at a large time step, in order to improve simulation accuracy, a correction processing of the switching delay is implemented into the UPFC model. Calculation accuracy of the real‐time UPFC model is at the same level as EMTDC^TM results. We confirm stable operation of the developed UPFC model by connecting a commercial real‐time digital simulator (RTDS^TM). © 2006 Wiley Periodicals, Inc. Electr Eng Jpn, 155(1): 19–26, 2006; Published online in Wiley InterScience ( www.interscience.wiley.com ). DOI 10.1002/eej.20200     

All‐pass dispersion synthesis using microwave C‐sections

A closed‐form iterative procedure for synthesizing quasi‐arbitrary phase responses with cascaded microwave C‐section all‐pass phasers is presented. The synthesis consists in mapping the transmission poles of the cascaded C‐section structure onto the transmission poles of the specified transfer function, where the latter poles are computed using a closed‐form polynomial generation method. The real and complex transmission poles of the specified transfer function are realized using C‐sections of different lengths and different couplings coefficients. The proposved synthesis is validated by both full‐wave analysis and measured multilayer prototypes. Copyright © 2013 John Wiley & Sons, Ltd.     

CNN‐based architecture for real‐time object‐oriented video coding applications

This paper presents a new CNN‐based architecture for real‐time video coding applications. The proposed approach, by exploiting object‐oriented CNN algorithms and MPEG encoding capabilities, enables low bit‐rate encoder/decoder to be designed. Simulation results using Claire video sequence show the effectiveness of the proposed scheme. Copyright © 2005 John Wiley & Sons, Ltd.     

Time–frequency higher‐order spectra with adjustment to the instantaneous frequency variation

Novel time–frequency techniques are proposed: the short time instantaneous higher‐order spectra (HOS) with adjustment to measured or known a priori time variation of the instantaneous frequency of transient signals. It is shown that the proposed transforms are more effective than the non‐instantaneous HOS (i.e. without adjustment to time variation of the instantaneous frequency) in recognizing a non‐stationary nonlinearity. Copyright © 2008 John Wiley & Sons, Ltd.     

Low‐power,low‐noise,active‐RC band‐pass filters using a ‘lossy’ LP–BP transformation

A new and straightforward design procedure for simple canonical topologies of allpole, active‐RC, low‐selectivity band‐pass (BP) filters, with low sensitivity to component tolerances is presented. The procedure is primarily intended for discrete‐component, low‐power filter applications using just one amplifier for relatively high‐order filters. The design procedure starts out with an ‘optimized’ low‐pass (LP) prototype filter, yielding an ‘optimized’ BP filter, whereby the wealth of ‘optimized’ single‐amplifier LP filter designs can be exploited. Using a so‐called ‘lossy’ LP–BP transformation, closed‐form design equations for the design of second‐ to eighth‐order, single‐amplifier BP filters are presented. The low sensitivity, low power consumption, and low noise features of the resulting circuits, as well as the influence of the finite gain‐bandwidth product and component spread, are demonstrated for the case of a fourth‐order filter example. The optimized single‐opamp fourth‐order filter is compared with other designs, such as the cascade of optimized Biquads. Using PSpice with a TL081 opamp model, the filter performance is simulated and the results compared and verified with measurements of a discrete‐component breadboard filter using 1% resistors, 1% capacitors, and a TL081 opamp. Copyright © 2011 John Wiley & Sons, Ltd.     

Small‐signal stability based on a synchronous generator model with field mutual‐leakage reactance

Power system small‐signal stability studies have been carried out using a synchronous machine model with a field mutual‐leakage reactance called the Canay reactance in the d‐axis equivalent circuit. The influence of the Canay reactance on the effectiveness of a power system stabilizer (PSS) was evaluated by the eigenvalue calculation method. The values of the Canay reactance reported in the literature were also investigated statistically, and an approximate expression applicable to power system stability studies was proposed. © 2005 Wiley Periodicals, Inc. Electr Eng Jpn, 154(2): 19–29, 2006; Published online in Wiley InterScience ( www.interscience.wiley.com ). DOI 10.1002/eej.20163     

An adaptive noise reduction filter for discrete signal by use of sandglass‐type neural network

An adaptive noise reduction filter composed of a sandglass‐type neural network (SNN) noise reduction filter (RF) is proposed in this paper. SNN was originally devised to work effectively for information compression. It is a hierarchial network and is symmetrically structured. SNN consists of the same number of units in the input and output layers and a smaller number of units in the hidden layer. It is known that SNN has signal processing performance which is equivalent to Karhunen–Loeve expansion after learning. We proved the theoretical suitability of SNN for an adaptive noise reduction filter for discrete signals. The SNNRF behaves optimally when the number of units in the hidden layer is equal to the rank of the covariance matrix of the signal components included in the input signal. Further we show by applying the recursive least squares method to learning of the SNNRF that the filter can process signals for on‐line adaptive noise reduction. This is an extremely desirable feature for practical application. In order to verify the validity of SNNRF, we performed computer experiments examining how the noise reduction ability of SNNRF is affected by altering the properties of the input pattern, learning algorithm, and SNN. The results confirm that the SNNRF acquired appropriate characteristics for noise reduction from the input signals, and markedly improved the SNR of the signals. © 1999 Scripta Technica, Electr Eng Jpn, 127(4): 39–51, 1999     

1.5 V square‐root domain universal biquad filters

Square‐root domain universal biquad topologies are introduced in this paper. One of them is single input multiple output, while the other one is multiple input single output biquad. Important benefits offered by the proposed topologies are the electronic adjustment of the resonant frequency and the capability for operating in a low‐voltage environment; also, the resonant frequency could be adjusted without disturbing the Q factor and vice‐versa. Simulation results using the Spectre simulator of the Analog Design Environment of Cadence software validate the correct operation of the proposed topologies and provide important performance characteristics. Copyright © 2011 John Wiley & Sons, Ltd.     

Simple and accurate approaches to implement the complex trans‐conductance suited for time‐domain simulators for small‐signal and large‐signal table‐based models

This paper focuses on the implementation of table‐based models of high‐frequency transistors for time‐domain simulators at microwave and mm‐wave frequencies. In this frequency range, the channel is not capable of responding to the excitation instantaneously therefore, a delay‐time exists between the channel response and the channel excitation. This delay is represented by a complex trans‐conductance in terms of circuit elements. The high‐frequency models of transistors are required to have the implementation of complex trans‐conductance, where the complex part accounts mathematically for the delay‐time between the channel response and the channel excitation. This paper presents simple and accurate approaches to incorporate the complex trans‐conductance in both small‐signal and large‐signal table‐based models for time‐domain simulators (MOS‐AK International Meeting. Eindhoven, Netherlands, April 2008). Implementation approach for each model, small‐signal and large‐signal, is presented in separated sections. In the first step, the delay is realized by the introduction of an ideal transmission line between the channel excitation and the channel response. As transmission lines are not generally suitable for time‐domain simulations, a lumped element equivalent network is introduced in the second step. The latter approach is fully compatible with time‐domain simulators but frequency limitation, determined by the delay‐time value itself, is introduced. Then the implementation of the complex trans‐conductance in large‐signal model is introduced. In terms of large‐signal behavior, delay‐time is important to achieve a non‐quasi static model. Yet again there is limitation in terms of the frequency range that is determined by the delay value itself. The methodology is illustrated on the small‐signal and the large‐signal equivalent circuit of a Multi‐Fin MOSFET transistor. Simulations are carried out by Cadence Spectre and Agilent ADS simulators, and comparisons are carried out between the simulation results and the measurements. Copyright © 2010 John Wiley & Sons, Ltd.     

Single channel informed signal separation using artificial‐stereophonic mixtures and exemplar‐guided matrix factor deconvolution

In this paper, a method is proposed to tackle the problem of single channel audio separation. The proposed method leverages on the exemplar source is used to emulate the targeted speech signal. A multicomponent nonnegative matrix factor 2D deconvolution (NMF2D) is proposed to model the temporal and spectral changes and the number of spectral basis of the audio signals. The paper proposes an artificial auxiliary channel to imitate a pair of stereo mixture signals, which is termed as “artificial‐stereophonic mixtures.” The artificial‐stereophonic mixtures and the exemplar source are jointly used to guide the factorization process of the NMF2D. The factorization is adapted under a hybrid framework that combines the generalized expectation–maximization algorithm with multiplicative update adaptation. The proposed algorithm leads to fast and stable convergence and ensures the nonnegativity constraints of the solution are satisfied. Adaptive sparsity has also been introduced on each sparse parameter in the multicomponent NMF2D model when the exemplar deviates from the target signal. Experimental results have shown the competence of the proposed algorithms in comparison with other algorithms.     

1‐V continuous‐time equalizers for multi‐gigabit short‐haul optical fiber communications

Two new CMOS analog continuous‐time equalizers for high‐speed short‐haul optical fiber communications are presented in this paper. The proposed structures compensate the limited bandwidth‐length product of 1‐mm step‐index polymer optical fiber channels (45 MHz, 100 m) and have been designed in a standard 0.18‐µm CMOS process. The equalizers are aimed for multi‐gigabit short‐range applications, targeting up to 2 Gb/s through a 50‐m step‐index polymer optical fiber. The prototypes operate with a single supply voltage of only 1 V and overcome the severe limitations suffered by the widely used degenerated differential pair caused by the low supply voltage. Copyright © 2012 John Wiley & Sons, Ltd.     

Forecasting of solar irradiance with just‐in‐time modeling

PV power output mainly depends on solar irradiance, which is affected by various meteorological factors. Thus, it is required to predict solar irradiance in the future for the efficient operation of PV systems. In this paper we develop a novel approach for solar irradiance forecasting, in which we combine the black–box model (JIT modeling) with the physical model (GPV data). We investigate the predictive accuracy of solar irradiance over a wide controlled area of each electric power company by utilizing measured data from 44 observation points throughout Japan provided by JMA and 64 points around Kanto provided by NEDO. Finally, we propose an applied forecasting method of solar irradiance to deal with difficulties in compiling databases. We also consider the influence of different GPV default times on solar irradiance prediction. © 2012 Wiley Periodicals, Inc. Electr Eng Jpn, 182(4): 19–28, 2013; Published online in Wiley Online Library ( wileyonlinelibrary.com ). DOI 10.1002/eej.22338