Improvement of the convergence characteristics of the NLMS algorithm has received attention in the area of adaptive filtering. A new variable stepsize NLMS method, in which the stepsize is updated optimally by using variances of the measured error signal and the estimated noise, is proposed. The optimal control equation of the stepsize has been derived from a convergence characteristic approximation. A new condition to judge convergence is introduced in this paper to ensure the fastest initial convergence speed by providing precise timing to start estimating noise level. And further, some adaptive smoothing devices have been added into the ADF to overcome the saturation problem of the identification error caused by some random deviations. By the simulation, The initial convergence speed and the identification error in precise identification mode is improved significantly by more precise adjustment of stepsize without increasing in computational cost. The results are the best ever reported performanced. This variable stepsize NLMS-ADF also shows good effectiveness even in severe conditions, such as noisy or fast changing circumstances.

In this letter, we introduce a predictor based least square (PLS) algorithm. By involving both order- and time-update recursions, the PLS algorithm is found to have a more stable performance compared with the stable version (Version II) of the RLS algorithm shown in Ref.[1]. Nevertheless, the computational requirement is about 50% of that of the RLS algorithm. As an application, the PLS algorithm can be applied to the fast Newton transversal filters (FNTF). The FNTF algorithms suffer from the numerical instability problem if the quantities used for extending the gain vector are computed by using the fast RLS algorithms. By combing the PLS and the FNTF algorithms, we obtain a much more stable performance and a simple algorithm formulation.

Design techniques are presented for high performance microstrip bandpass filters using GaAs FETs for loss compensation. The filters are based on conventional planar filter topologies with the addition of GaAs FET negative resistance circuits to amplify the signal within the resonators via a reflection-mode of amplification. Three practical filters have been demonstrated using these negative resistance techniques: (1) A filter employing an active loop configuration, (2) a dual-mode microstrip ring resonator filter, and (3) an end-coupled half-wavelength resonator filter. The investigation of this negative resistance method of loss compensation has led to the development of an exciting new type of miniaturised filter which employs MIC microstrip resonators with MMIC negative resistance chips bonded into the filter for loss compensation. This approach has the advantage of combining the proven capabilities of established MIC microstrip filter topologies with the excellent reproducibility of the MMIC loss compensation circuits.

A novel method is presented for designing discrete coeffcient FIR linear phase filters using Hopfield neural networks. The proposed method is based on the minimization of the energy function of Hopfield neural networks. In the proposed method, the optimal solution for each filter gain factor is first searched for, then the optimal filter gain factor is selected. Therefore, a good solution in the specified criterion can be obtained. The feature of the proposed method is that it can be used to design FIR linear phase filters with different criterions simultaneously. A design example is presented to demonstrate The effectiveness of the proposed method.

In this paper, we discuss design of quadrature mirror filter (QMF) banks using digital allpass networks in the frequency domain. In the QMF banks composed of a parallel connection of two allpass networks, both aliasing error and amplitude distortion are always completely canceled. Therefore, we only need to design the analysis filters and eliminate phase distortion of the overall transfer function. We consider design of the QMF banks in two cases where phase responses of the filters are repuired or not required. In the case where the phase responses are not required, the design problem can be reduced to design of phase difference of two allpass networks. In the case where the phase responses are required, we present a procedure for designing the QMF banks with both equiripple magnitude and phase responses.

This paper describes newly developed miniaturized stepped impedance resonators with a double coaxial structure (DC-SIR's) and their application to bandpass filters. The new DC-SIR's using dielectric material are devised for more compact and lower frequency bandpass filters. Fundamental characteristics such as resonance properties and unloaded-Q make it clear that DC-SIR's have attractive features that miniaturization can be achieved without Q-factor degradation. Trial 400 MHz bandpass filters incorporating DC-SIR's are also made. Experimental results of bandpass filters proved that DC-SIR's are applicable to lower frequency band radio equipment and able to contribute to the expansion of applicable frequency ranges of dielectric coaxial resonators.

The fundamental TE10 mode in a rectangular waveguide of a square cross section is degenerate with TE01 mode. A quarter wavelength resonator made of a dielectric square waveguide is, therefore, applied for a small-sized bandpass filter, just like dual mode filters for base stations in the mobile communication. In this paper, the methods to couple the two modes are first studied, including cutting a corner of the resonator and adding some metal electrodes on its end face. Both methods help to flow the rf current of the odd mode at the corner, resulting in decrease of the series inductance and thus increase of the resonant frequency. The coupling constant, that is proportional to the difference of the odd and even-mode's resonant frequency, can be controlled by the perturbations mentioned above. The coupling to the external circuit is adjusted by an electrode fabricated also on the end face. It is connected to a microstrip line and capacitively couples to the resonant modes. The coupling strength increases with the dimension of the electrode. The adjustment of the resonant frequency is carried out by the similar electrode on the end face and connected to the center of the side of the square cross section. The frequency decreases with the length of the electrode. The unloaded Q is measured to be of around 500 for 5510 mm resonator of εr=93. The optimum aspect ratio for the resonator is found in terms of the Q value. The simplest bandpass filter, i.e., a two-stage bandpass filter is designed and fabricated using 5510 mm resonator. It is mounted in a square hole made in a printed circuit board and excited by a microstrip line. The frequency characteristics are in good agreement with the expected values.

A fiber optic temperature sensor using a conventional graded index multimode optical fiber is proposed. The multimode fiber is excited by two selected modes using a computer-generated holographic filter. A clear periodic signal created by interference between two modes is observed in the experiment.

An adaptive joint-process IIR filter with generalized lattice structure is constructed. This filter can borrow both FIR and IIR features and simultaneously holds the well-known merits of lattice structure.

In this paper we present a design method for all-pass networks with consideration of the stability. It is based on the eigen filter method and Remez exchange algorithm is used to obtain the equiripple phase error solution. In the iteration of the proposed algorithm, the eigen values besides maximum eigen value are used in order to obtain a stable all-pass networks.

This paper presents a new structure for noise and echo cancelers based on a combined fast abaptive algorithm. The main purpose of the new structure is to detect both the double-talk and the unknown path change. This goal is accomplished by using two adaptive filters. A main adaptive filter Fn, adjusted only in the non-double-talk period by the normalized LMS algorithm, is used for providing the canceler output. An auxiliary adaptive filter Ff, adjusted by the fast RLS algorithm, is used for detecting the double-talk and obtaining a near optimum tap-weight vector for Fn in the initialization period and whenever the unknown path has a sudden or fast change. The proposed structure is examined through computer simulation on a noise cancellation problem. Good cancellation performance and stable operation are obtained when signal is a speech corrupted by a white noise, a colored noise and another speech signal. Simulation results also show that the proposed structure is capable of distinguishing the near-end signal from the noise path change and quickly tracking this change.

A novel adaptive pitch-synchronous analysis method is proposed to estimate simultaneously vocal tract (formant/antiformant) and voice source parameters from speech waveforms. We use the parametric Rosenberg-Klatt (RK) model to generate a glottal waveform and an autoregressive-exogenous (ARX) model to represent voiced speech production process. The Kalman filter algorithm is used to estimate the formant/antiformant parameters from the coefficient of the ARX model, and the simulated annealing method is employed as a nonlinear optimization approach to estimate the voice source parameters. The two approaches work together in a system identification procedure to find the best set of the parameters of both the models. The new method has been compared using synthetic speech with some other approaches in terms of accuracy of estimated parameter values and has been proved to be superior. We also show that the proposed method can estimate accurately the parameters from natural speech sounds. A major application of the analysis method lies in a concatenative formant synthesizer which allows us to make flexible control of voice quality of synthetic speech.

We propose an objective measure from assessing low-rate coded speech. The model for this objective measure, in which several known features of the perceptual processing of speech sounds by the human ear are emulated, is based on the Hertz-to-Bark transformation, critical-band filtering with preemphasis to boost higher frequencies, nonlinear conversion for subjective loudness, and temporal (forward) masking. The effectiveness of the measure, called the Bark spectral distortion rating (BSDR), was validated by second-order polynomial regression analysis between the computed BSDR values and subjective MOS ratings obtained for a large number of utterances coded by several versions of CELP coders and one VSELP coder under three degradation conditions: input speech levels, transmission error rates, and background noise levels. The BSDR values correspond better to MOS ratings than several commonly used measures. Thus, BSDR can be used to accurately predict subjective scores.

A multi-layer perceptron (MLP) acting directly in the time-domain is applied as a speech signal enhancer, and the performance examined in the context of three common classes of degradation, namely low bit-rate CELP degradation is non-linear system degradation, additive noise, and convolution by a linear system. The investigation focuses on two topics: (i) the influence of non-linearities within the network and (ii) network topology, comparing single and multiple output structures. The objective is to examine how these characteristics influence network performance and whether this depends on the class of degradation. Experimental results show the importance of matching the enhancer to the class of degradation. In the case of the CELP coder the standard MLP with its inherently non-linear characteristics is shown to be consistently better than any equivalent linear structure (up to 3.2 dB compared with 1.6 dB SNR improvement). In contrast, when the degradation is from additive noise, a linear enhancer is always, superior.

This paper presents a new Adaptive Convergence Factor (ACF) algorithm without the damping parameter adjustment acoording to the input signal and/or the composition of the filter system. The damping parameter in the ACF algorithms has great influence on the convergence characteristics. In order to examine the relation between the damping parameter and the convergence characteristics, the normalization which is realized by the related signal terms divided by each maximum value is introduced into the ACF algorithm. The normalized algorithm is applied to the modeling of unknown time-variable systems which makes it possible to examine the relation between the parameters and the misadjustment in the adaptive algorithms. Considering the experimental and theoretical results, the optimum value of the damping parameter can be defined as the minimum value where the total misadjustment becomes minimum. To keep the damping parameter optimum in any conditions, the new ACF algorithm is proposed by improving the invariability of the damping parameter in the normalized algorithm. The algorithm is investigated by the computer simulations in the modeling of unknown time-variable systems and the system indentification. The results of simulations show that the proposed algorithm needs no adjustment of the optimum damping parameter and brings the stable convergence characteristics even if the filter system is changed.

Wide bandwidth of 30MHz was achieved by 3-IDT SAW resonator filters using 36Y-X LiTaO3 substrates at 800MHz. The filter, which has fractional bandwidth of 3.6%, can be applied for several mobile communication systems. Low insertion loss of 1.7 dB was obtained by taking parallel arranged configuration, which proved to be 0.4 dB better than simple cascade-connected configuration.

A noise cancelling sonar-ranging system based on the adaptive filtering technique, which can automatically adapt itself to the changes in environmental noise-field and improve the passive sonar-ranging/goniometric precision, was introduced by this paper. In the meantime, the software and hardware design principle of the system using high speed VLSI (Very Large Scale Integrated) DSP (Digital Signal Processing) chips, and the practical test results were also presented. In comparison with the traditional ranging system, the system not only enhanced obviously the ranging precision but also possessed some more characteristics such as simple structure, rapid operation, large data-storage volume, easy programming, high reliability and so on.

A new gloss-extracting method is proposed in this study. A spatial filter with variable resolution is used for the extraction of glossiness. Various spheres and cylinders with curvature radii from 4 to mm are used as the specimens. In all samples, a strong correlation, with a correlation coefficient of more than 0.98, has been observed between psychological glossiness Gph perceived by the human eye and glossiness Gfm extracted by this method. This method is useful for plane specimens as well as spherical and cylindrical ones.

In this report, we propose a robust block adaptive digital filter (BADF) which can improve the accuracy of the estimated weights by averaging the adaptive weight vectors. We show that the improvement of the estimated weights is independent of the input signal correlation.

The discrete-time short-time Fourier transform (STFT) is known as a useful tool for analyzing and synthesizing signals. This paper introduces an extention of the well-known STFT to a general form which is more suitable for high resolutional signal analysis. A channel frequency division scheme is developed for realizing arbitrary bandwidth and center frequency so as to improve resolution performance. It is based on a nonuniform filter bank structure with integer decimation and interpolation factors. A design example of the generalized STFT using symmetric windows is given.