At frequencies in the GHz range, an electrical connector must be considered as part of an electromagnetic transmission line. This paper reviews the effect of signal frequency on constriction resistance, interfacial capacitance and contact inductance at an electrical interface in a high speed connector. The deleterious effects of contact degradation at pin-receptacle junctions on transmitted signal integrity, are addressed. For frequencies in the GHz range, an electrical interface becomes capacitively coupled if contact resistance increases sufficiently. Contact deterioration may also lead to the generation of parasitic third-order harmonics that contribute to loss of signal integrity.

This paper presents a theoretical convergence analysis of a CORDIC-based adaptive ARMA lattice filter. In previous literatures, several investigation methods for adaptive lattice filters have been proposed; however, they are available only for AR-type filters. Therefore, we have developed a distinct technique that can reveal the convergence properties of the CORDIC ARMA lattice filter. The derived technique provides a quantitative convergence analysis, which facilitates an efficient hardware design for the filter. Moreover, our analysis technique can be applied to popular multiplier-based filters by slight modifications. Hence, the presented convergence analysis is significant as a leading attempt to investigate ARMA lattice filters.

A speech signal captured by a distant microphone is generally smeared by reverberation, which severely degrades both the speech intelligibility and Automatic Speech Recognition (ASR) performance. Previously, we proposed a single-microphone dereverberation method, named "Harmonicity based dEReverBeration (HERB)." HERB estimates the inverse filter for an unknown room transfer function by utilizing an essential feature of speech, namely harmonic structure. In previous studies, improvements in speech intelligibility was shown solely with spectrograms, and improvements in ASR performance were simply confirmed by matched condition acoustic model. In this paper, we undertook a further investigation of HERB's potential as regards to the above two factors. First, we examined speech intelligibility by means of objective indices. As a result, we found that HERB is capable of improving the speech intelligibility to approximately that of clean speech. Second, since HERB alone could not improve the ASR performance sufficiently, we further analyzed the HERB mechanism with a view to achieving further improvements. Taking the analysis results into account, we proposed an appropriate ASR configuration and conducted experiments. Experimental results confirmed that, if HERB is used with an ASR adaptation scheme such as MLLR and a multicondition acoustic model, it is very effective for improving ASR performance even in unknown severely reverberant environments.

Beam control using active antenna arrays with self-oscillating harmonic mixers has been investigated. The active antenna is composed of a patch antenna receiving RF signal and a parallel feedback type oscillator which operates as the self-oscillating harmonic mixer, and down-converts the received RF signal into IF signal. The mixer has two ports for local oscillating (LO) signal. One is an output port extracting the LO signal. The other is an input port for an injection signal to synchronize the local oscillation. The mixers can be coupled unilaterally without other nonreciprocal components by connecting the output port to the input port in the next mixer. In the unilaterally coupled array, the phase differences of the LO signals between the adjacent mixers can be varied without phase shifters in injection locking state by changing the local free-running frequencies of the self-oscillating mixers. The receiving pattern can be controlled by combining the IF signals from the individual active antennas, which have phases associated with the LO signals. The IF is difference between the RF and double of the LO frequency so that arbitrary phase differences from 0 to 2π radian can be provided to the output IF signals. The experiments using the two- and three-element arrays demonstrated beam control capability.

An even-harmonic mixer using a bipolar differential pair (bipolar harmonic mixer;BHMIX) is theoretically analyzed from the direct conversion point of view; i.e, conversion gain, third-order input intercept point (IIP3), self-mixing induced dc offset level, and second-order input intercept point (IIP2). Also, noise are analyzed based on nonlinear large-signal model, and numerical results are given. Noises are treated as cyclostationary noises, thus all the folding effects are taken into account. Factors determining IIP3, IIP2, dc offset, and noise are identified and estimation procedures for these characteristics are obtained. For example, design guidelines for the optimal noise performance are given. Measured results support all the analysis results, and they are very useful in the practical BHMIX design.

In this paper, the previously introduced periodic Fourier transform concept is extended to a two-dimensional case. The relations between the periodic Fourier transform, harmonic series representation and Fourier integral representation are also discussed. As a simple application of the periodic Fourier transform, the scattering of a scalar wave from a finite periodic surface with weight is studied. It is shown that the scattered wave may have an extended Floquet form, which is physically considered as the sum of diffraction beams. By the small perturbation method, the first order solution is given explicitly and the scattering cross section is calculated.

In this letter, a novel design of a branch-line coupler with considerable reduction in its size and suppressed harmonic passband is proposed. By embedding a defected ground structure (DGS) unit cell under a microstrip line, compact branch-line couplers are easily achieved. The electrical length is scaled appropriately according to the slow-wave effect. In this case, the experimental coupling (S21 or S31) is comparable to that of conventional branch-line couplers. Also, experimental results indicate that DGS section is quite effective for the suppression of higher order harmonics.

For the power amplifier used in CDMA cellular phones, the supply voltage is switched between high and low at a transmission power several decibels higher than 10 dBm using a DC-DC converter to improve operational efficiency. The longer the operation time under low supply voltage, the lower the current consumption of the cellular phone. In order to increase the output power under low supply voltage, we applied the 2nd harmonic-injection technique, which is useful for distortion compensation. With 2nd harmonic-injection, there is an inflectional power point. The distortion increases rapidly when output power goes beyond the inflectional power point. It is important to make this inflectional power point high in order to compensate for distortion in the high output-power region. We report here that the inflectional power point can be increased by connecting a 2nd harmonic short circuit to the drain terminal of the FET to which the 2nd harmonic for distortion compensation is injected. A prototype of the final stage of the power amplifier under a supply voltage of Vdd=1.5 V is presented. We report that applying a CDMA uplink signal, 1.5 dB higher output power and 12% higher drain efficiency is achieved compared when only 2nd harmonic injection is employed.

In this report, we propose a tracking algorithm of speaker direction using microphones located at vertices of an equilateral triangle. The method realizes tracking by minimizing a performance index that consists of the cross spectra at three different microphone pairs in the triangular array. We adopt the steepest descent method to minimize it, and for guaranteeing global convergence to the correct direction with high accuracy, we alter the performance index during the adaptation depending on the convergence state. Through some computer simulation and experiments in a real acoustic environment, we show the effectiveness of the proposed method.

This paper proposes a new robust adaptive processing algorithm that is based on the extended least squares (ELS) method with running spectrum filtering (RSF). By utilizing the different characteristics of running spectra between speech signals and noise signals, RSF can retain speech characteristics while noise is effectively reduced. Then, by using ELS, autoregressive moving average (ARMA) parameters can be estimated accurately. In experiments on real speech contaminated by white Gaussian noise and factory noise, we found that the method we propose offered spectrum estimates that were robust against additive noise.

This paper presents a novel method to speed up neural network (NN) based face detection systems. NN-based face detection can be viewed as a classification and search problem. The proposed method formulates the face search problem as an integer nonlinear optimization problem (INLP) and expands the basic particle swarm optimization (PSO) to handle it. PSO works with a population of particles, each representing a subwindow in an input image. The subwindows are evaluated by how well they match a NN based face filter. A face is indicated when the filter response of the best particle is above a given threshold. Experiments on a set of 42 test images show the effectiveness of the proposed approach. Moreover, the effect of PSO parameter settings on the search performance was investigated.

Borrowing the notion of instantaneous frequency that was developed in the context of time-frequency signal analysis, an instantaneous frequency amplitude spectrum (IFAS) is introduced for estimating fundamental frequency of speech signal in both noiseless and adverse environments. We define harmonicity measure as a quantity that indicates degree of periodical regularity in the IFAS and that shows substantial difference between periodic signal and noise-like waveform. The harmonicity measure is applied to estimate the existence of fundamental frequency. We provide experimental examples to demonstrate the general applicability of the harmonicity measure and apply the proposed procedure to Japanese continuous speech signals. The results show that the proposed method outperforms the conventional methods with or without the presence of noise.

This investigation proposes a novel dual-mode patch bandpass filter (BPF) that uses defect ground structure (DGS) to suppress spurious response. The proposed dual-mode patch BPF has exhibits a wide stopband characteristic owing to that uses the bandgap resonant characteristic of DGS in the harmonic frequency of the dual-mode patch BPF. The novel proposed filter demonstrates the frequency characteristics with center frequency f0 = 2.2 GHz, 3-dB bandwidth (FBW) of 8% and wider stopband from 2.6 to 6 GHz at the level of -35 dB. The experimental and simulated results agree.

This paper deals with the problem of multiple object tracking with the condensation algorithm, applied to tracking of soccer players. To solve the problem of failures in tracking multiple players under overlapping, we introduce occlusion alarm probability, which attracts or repels particles based on their posterior distribution of previous time step. Real experiments showed a robust performance.

The proposed DOA (Direction Of Arrival) estimation method by integrating the frequency array data generated from microphone pairs in an equilateral-triangular microphone array is extended here. The method uses four microphones located at the apices of regular tetrahedron to enable to estimate the elevation angle from the array plane as well. Furthermore, we introduce an idea for separate estimation of azimuth and elevation to reduce the computational loads.

A new excitation enhancement technique based on a harmonic model is proposed in this paper to improve the speech quality of low-bit-rate speech coders. This technique is employed only in the decoding process of speech coders and improves high-frequency components of excitation. We develop the procedure of harmonic model parameters estimation and harmonic generation and apply the technique to a current state-of-art low bit rate speech coder. Experiments on spectrum reading and spectrum distortion measurement show that the proposed excitation enhancement technique improves speech quality.

A fully differential direct conversion receiver IC for W-CDMA is presented. The receiver IC consists of an LNA, a quadrature demodulator, low-pass filters (LPFs), and variable gain amplifiers (VGAs). In order to suppress DC offset, which is the most important issue in a direct conversion system, an active harmonic mixer is applied to the quadrature demodulator. Furthermore, a receiving system, including the LNA and an RF filter, adopts a differential architecture to reduce local signal leakage, which generates DC offset. Performance of the entire receiving system was evaluated and DC offset in steady state was measured at only 40 mV. Moreover, DC offset variation at the LNA gain change, which has the largest affect on the receiving performance, was limited to 70 mV, which is less than -10 dB compared to desired signal strength. It was confirmed by computer simulation that the DC offset variation at the LNA gain change did not degrade bit error rate (BER) performance at all.

In this paper, we propose a DOA (Direction Of Arrival) estimation method of speech signal using three microphones. The angular resolution of the method is almost uniform with respect to DOA. Our previous DOA estimation method using the frequency-domain array data for a pair of microphones achieves high precision estimation. However, its resolution degrades as the propagating direction being apart from the array broadside. In the method presented here, we utilize three microphones located at vertices of equilateral triangle and integrate the frequency-domain array data for three pairs of microphones. For the estimation scheme, the subspace analysis for the integrated frequency array data is proposed. Through both computer simulations and experiments in a real acoustical environment, we show the efficiency of the proposed method.

This paper proposes a robust method for estimating the fundamental frequency (F0) in real environments. It is assumed that the spectral structure of real environmental noise varies momentarily and its energy does not distribute evenly in the time-frequency domain. Therefore, segmenting a spectrogram of speech mixed with environmental noise into narrow time-frequency regions will produce low-noise regions in which the signal-to-noise ratio is high. The proposed method estimates F0 from the periodic and harmonic features that are clearly observed in the low-noise regions. It first uses two kinds of spectrogram, one with high frequency resolution and another with high temporal resolution, to represent the periodic and harmonic features corresponding to F0. Next, the method segments these two kinds of feature plane into narrow time-frequency regions, and calculates the probability function of F0 for each region. It then utilizes the entropy of the probability function as weight to emphasize the probability function in the low-noise region and to enhance noise robustness. Finally, the probability functions are grouped in each time, and F0 is obtained as the frequency with the highest probability of the function. The experimental results showed that, in comparison with other approaches such as the cepstrum method and the autocorrelation method, the developed method can more robustly estimate F0s from speech in the presence of band-limited noise and car noise.

This paper describes the performance improvement of power amplifiers by defected ground structure (DGS). Due to the excellent capability of harmonic rejection and tuning, DGS plays a great role in improving the major nonlinear behaviors of power amplifier such as output power, harmonics, power added efficiency (PAE), and the ratio between the carrier and the third order intermodulation distortion (C/IMD3). In order to verify the improvement of performances by DGS, measured data for a power amplifier, which adopts a 30 Watts LDMOS device for the operation at 2.1-2.2 GHz, are illustrated under several operating bias currents for two cases, i.e., with and without DGS attached. The principle of the improvement is described by the simple Volterra nonlinear transfer functions with the consideration of different operating classes. The obtained improvement of the 30 Watts power amplifier, under 400 mA of IdsQ as an example, includes the reduction in the second and third harmonics by 17 dB and 20 dB, and the increase in output power, PAE, and C/IMD3 by 1.3 Watts, 3.4%, and 4.7 dB, respectively.