This paper describes a spatial spectral subtraction method by using the complementary beamforming microphone array to enhance noisy speech signals for speech recognition. The complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this paper, it is shown that the nonlinear subtraction processing with complementary beamforming can result in a kind of the spectral subtraction without the need for speech pause detection. In addition, the optimization algorithm for the directivity pattern is also described. To evaluate the effectiveness, speech enhancement experiments and speech recognition experiments are performed based on computer simulations under both stationary and nonstationary noise conditions. In comparison with the optimized conventional delay-and-sum (DS) array, it is shown that: (1) the proposed array improves the signal-to-noise ratio (SNR) of degraded speech by about 2 dB and performs more than 20% better in word recognition rates under the conditions that the white Gaussian noise with the input SNR of -5 or -10 dB is used, (2) the proposed array performs more than 5% better in word recognition rates under the nonstationary noise conditions. Also, it is shown that these improvements of the proposed array are same as or superior to those of the conventional spectral subtraction method cascaded with the DS array.

As a means of CDMA network evolution toward future wireless services, a spectral overlay of narrowband CDMA (N-CDMA) and wideband CDMA (W-CDMA) systems is proposed in [8]. In order to justify the overlaying strategy, the reverse link capacity is examined in the same work. Although the capacity of conventional CDMA cellular systems is usually limited by the reverse link, the limit could occur at the forward link depending on the transmission technologies adopted by specific CDMA proposals. Especially, the number of users that can be simultaneously accommodated in the system would be limited by the forward link in future mobile service environments where unequal traffic is offered between two links. In this paper, we first examine the forward link capacity of the spectrally overlaid narrowband and wideband CDMA (N/W CDMA) system. And we compare it with the reverse link capacity to obtain the overall performance. The effects of various parameters on the capacity of N/W CDMA system are numerically evaluated for different mobile environments.

We propose an optical spread-time code-division multiple-access (ST-CDMA) with pulse position modulation (PPM) signaling for high-speed communication networks. We obtain a union upper bound on the bit error rate (BER) considering the multi-access interference (MAI), shot noise and thermal noise at the receiver. As a result, we show that the optical ST-CDMA with PPM signaling improves the BER performance at the same received power and bit rate compared to that with OOK signaling. This leads to an increase of the bit rate at the same BER. Moreover, we show that the proposed system can relax the requirement for spectral resolution compared to the optical ST-CDMA with OOK signaling under the received power and bit rate constraints.

The formulation of the wall admittance of a circular microstrip antenna by the spectral domain method is presented. The circular microstrip antenna is calculated using the cavity model. The electromagnetic fields within the antenna cavity are determined from the impedance boundary condition at the side aperture. The contribution from the region outside the antenna is taken into account by the wall admittance. The wall admittance is defined by the magnetic field produced by the equivalent magnetic current at the aperture. The magnetic field is calculated by the spectral domain method. The wall admittances obtained by this method are compared with the results calculated by Shen. The calculated input impedances of the microstrip antenna agree fairly well with the experimental data for the substrate thickness of up to 0.048λg. The formulation of wall admittance presented here is easily applicable to arbitrarily shaped microstrip antennas.

An Acoustic echo canceller has problems adaptating under noisy or double-talk conditions. The adaptation process requires a precise identification of the temporarily changed room impulse response. To do this, both minimizing the step size parameter of the Least Mean Square (LMS) method to be as small as possible and giving up on updating the adaptive filter coefficients have been considered. This paper describes an adaptive cross-spectral technique that is robust to adaptive filtering under noisy or double-talk conditions and for colored signals such a speech signal. The cross-spectral technique was originally developed to measure the impulse response in a linear system. Here we apply in the adaptive cross-spectral technique to solve the acoustic echo cancelling problem. This cross-spectral technique takes the ensemble average of the cross spectrum between input and error signals and the averaged cross spectrum is divided by the averaged power spectrum of the input signal to update the filter coefficients. We have confirmed that the echo signal is suppressed by about 15 dB even under double-talk conditions. We also explain that this method has a systematic error due to using a short time block for estimating the room impulse response. Then we investigate overlapping every last half block by the following first half block in order to reduce the effect of the systematic error. Finally, we compare our method with the Frequency-domain Block LMS (FBLMS) method because both methods are implemented in the frequency domain using a short time block.

This paper proposes a method of improving the accuracy of the attenuation constant estimate obtained by using the cross-spectral technique. In the cross-spectral technique, the envelope of the estimated impulse response is deformed due to the use of a time window. As a result, the estimated impulse response decays more rapidly than the real impulse response does, and the attenuation constant obtained by the estimated impulse response becomes larger than the real value. This paper first describes how the attenuation constant changes in the process of impulse response estimation. Next, we propose a method of improving the accuracy of the estimation. The effect of the proposed method is confirmed by computer simulation.

In this paper, the correlation properties are used to develop two efficient encoding schemes for speech line spectrum frequency (LSF) parameters. The first scheme (1D KL), which exploits the intraframe correlation, is based on one-dimensional Karhunen-Loeve (KL) transformation; the second scheme, which requires some coding delays to further utilize the interframe correlation, uses two-dimensional (2D KL) transform in the frequency domain or one-dimensional KL transform co-operating with DPCM in the time domain. Moreover, since the KL transform is globally optimal, which is sensitive to the change of input data statistics, further two adaptive transform coding systems are also investigated in this paper. The performance of all systems for different bit rates is investigated and adequate comparisons are made. It is shown that the gain of using KL transformation to exploit the intraframe and interframe correlation is 3 and 4 bits/speech frame, respectively.

The presence of a new surface-wave-like mode on CPWs of infinite width produces a complex transition region at the onset of leakage, involving the unusual simultaneous combination of a coupling region and a spectral gap. An examination of this region leads to a clear physical explanation of why sharp minima occur in the leakage behavior.

This paper presents a variant of the "Third-Order Recursion (TOR)" method for bispectral estimation of transfer-function parameters of a non-minimum-phase all-poles system. The modification is based on the segmentation of system-output data into coupled records, instead of independent records. It consists of considering the available data at the left and the right of each record as not null and taking them as the data corresponding to the preceding and succeeding record respectively. The proposed variant can also be interpreted as a "Constrained Third-Order Mean (CTOM)" method with a new segmentation in overlap records. Simulation results show that this new segmentation procedure gives more precise system parameters than the TOR and CTOM methods, to be obtained. Finally, in order to justify the use of bispectral techniques, the influence of added white and colored Gaussian noise on the parameter estimation is also considered.

In mobile communications, power is a very important factor and nonlinear amplification of power amplifiers cannot be avoided due to their high power efficiency. This article presents the performance of π/2-shift BPSK modulation scheme used in DS/SS/CDMA wireless communications over multipath Rayleigh fading channel and compares the performance with the performance of conventional BPSK and offset QPSK CDMA systems. The performance parameters: Out-of-Band power, average Bit Error Rate (BER) and Spectral Efficiency have been evaluated. In order to obtain improved performance on fading channels, a RAKE receiver has been employed. Finally it is shown that π/2-shift BPSK outperforms conventional BPSK and offset QPSK in the presence of nonlinear amplification.

In this paper, to obtain higher spectral efficiency than a conventional M-ary/Spread Spectrum (SS) communication system using an orthogonal code, we propose Double M-ary/SS communication systems in which transmitting sequences are produced by multiplying two sequences obtained from different M-ary/SS systems. First, we estimate the system performance of a Double M-ary/SS in which transmitting sequences are composed of two kinds of sequences which have the same chip duration and the same sequence length in AWGN environment by theoretical analysis and computer simulations using random sequences. And we show that the bit error rate (BER) of the system can be improved, compared to that of the conventional M-ary/SS. Next, we propose two concrete examples of the Double M-ary/SS systems. We investigate the performance of these systems in AWGN environment by computer simulations, and show that the BER and spectral efficiency can be improved by the two systems compared to those of the conventional M-ary/SS. Further, we consider a modified method of the demodulation for one of the examples of Double M-ary/SS systems, and show that we can reduce quantity of calculations by the method.

Recent progress in neural network research has demonstrated the usefulness of neural networks in a variety of areas. In this work, its application in the spatial resolution improvement of a remotely sensed low resolution thermal infrared image using high spatial resolution of visible and near-infrared images from Landsat TM sensor is described. The same work is done by an algebraic method. The tests developed are explained and examples of the results obtained in each test are shown and compared with each other. The error analysis is also carried out. Future improvements of these methods are evaluated.

This paper describes a classification method for rotated and scaled textured images using invariant parameters based on spectral-moments. Although it is well known that rotation invariants can be derived from moments of grey-level images, the use is limited to binary images because of its computational unstableness. In order to overcome this drawback, we use power spectrum instead of the grey levels to compute moments and adjust the integral region of moment evaluation to the change of scale. Rotation and scale invariants are obtained as the ratios of the different rotation invariants on the basis of a spectral-moment property with respect to scale. The effectiveness of the approach is illustrated through experiments on natural textures from the Brodatz album. In addition, the stability of the invariants with respect to the change of scale is discussed theoretically and confirmed experimentally.

The Heart Rate Variability (HRV) analysis has become vigorous these days. One reason for this is that the HRV analysis investigates the dynamics of the autonomic nervous system activities which control the HRV. The Integral Pulse Frequency Modulation (IPFM) model is a pulse generating mechanism model in the nervous system, that is one of the models which connects the HRV to the autonomic nervous system activities. The IPFM model is a single frequency component model; however, the real HRV has multiple frequency components. Moreover, there are refractory periods after generating action potentials are initiated. Nevertheless, the IPFM model does not consider refractory periods. In order to make sure of the accuracy and the effectiveness of the integral function (IF) method applied to the real data, we consider the absolute refractory periods and two frequency components. In this investigation, the simulated HRV was made with a single and double frequency component using the IPFM model with and without absolute refractory periods. The original generating function of the IPFM model was demodulated by using the instantaneous heart rate tachogram. The power of the instantaneous pulse rate per minute was analyzed by the direct FFT method, the IF FFT method without the absolute refractory periods, and the IF FFT method with the absolute refractory periods. It was concluded that the IF FFT method can demodulate the original generating function accurately.

In this paper we derive the expressions of the spectra of waveform relaxation operators for linear differential-algebraic equations which stem from circuit simulation. These expressions suggest ways to split the matrices of the circuit equations such that waveform relaxation will converge. Numerical experimental results are given.

The performance of a constrained (that is, minimal order) Yule-Walker (CYW) single tone frequency estimator is studied. A closed form expression for the asymptotic error variance is derived. It is shown that CYW does not satisfactorily utilize the informaiton in data, and estimators with improved performance are proposed. Simulation results which lend support to the theoretical findings are included.

Irreducible components of canonical graphs for second order spectral null constraints at a rational submultiple of the symbol frequency fsk/n are studied where fs is the symbol frequency. We show that if n is prime then a canonical graph consists of disjoint irreducible components. We also show that the number of irreducible components of a canonical graphs is finite if n is prime. For the case n = 2 and p O mod n, all aperiodic irreducible components are identified explicitly where p is a parameter of a canonical graph.

Spectral transform methods have been widely studied for classification and analysis of logic functions. Spectral methods have also been used for logic synthesis, and by use of BDDs, practical-sized synthesis problems have been solved. Wavelet theory has recently attracted the attention of researchers in the signal processing field. The Haar function is used in both spectral methods and in signal processing to obtain spectral coefficients of logic functions of signals. In this paper spectral transform-based analysis of neural nets verifying signal processing and discrete function is presented. A neural net element is defined as a discrete function with multi-valued input signals and multi-valued or binary outputs. The multi-valued variable is realized as a variable (V, W) formed by a pair of a binary value and a multi-value pulse width. The multi-valued encoding is used with the multi-valued Haar function to give meanings to the wavelet coefficients from the view of Boolean algebra. A design example shows that these conceptually different concepts are closely related.

This work presents a further development of the approach to modelling thermal (i.e. carrier-velocity-fluctuation) noise in semiconductor devices proposed in papers by the present authors. The basic idea of the approach is to apply classical theory of Ito's stochastic differential equations (SDEs) and stochastic diffusion processes to describe noise in devices and circuits. This innovative combination enables to form consistent mathematical basis of the noise research and involve a great variety of results and methods of the well-known mathematical theory in device/circuit design. The above combination also makes our approach completely different, on the one hand, from standard engineering formulae which are not associated with any consistent mathematical modelling and, on the other hand, from the treatments in theoretical physics which are not aimed at device/circuit models and design. (Both these directions are discussed in more detail in Sect. 1). The present work considers the bipolar transistor compact model derived in Ref. [2] according to theory of Ito's SDEs and stochastic diffusion processes (including celebrated Kolmogorov's equations). It is shown that the compact model is transformed into the Ito SDE system. An iterative method to determine noisy currents as entries of the stationary stochastic process corresponding to the above Ito system is proposed.

A new structure for adaptive AR spectral estimation based on multi-band decomposition of the linear prediction error is introduced and the mathematical background for the soulution of the related adaptive filtering problem is derived. The presented structure gives rise to AR spectral estimates that represent the true underlying spectrum with better fidelity than conventional LS methods by allowing an arbitrary trade-off between variance of spectral estimates and tracking ability of the estimator along the frequency spectrum. The linear prediction error is decomposed through a filter bank and components of each band are analyzed by different window lengths, allowing long windows to track slowly varying signals and short windows to observe fastly varying components. The correlation matrix of the input signal is shown to satisfy both time-update and order-update properties for rectangular windowing functions, and an RLS algorithm based on each property is presented. Adaptive forward and backward relations are used to derive a mathematical framework that serves as a basis for the design of fast RLS alogorithms. Also, computer experiments comparing the performance of conventional and the proposed multi-band methods are depicted and discussed.