This paper presents a fast convergence block adaptive filter in which the filter weights are adjusted based on the conjugate gradient method. The proposed algorithm permits the use of the fast convolution in accordance with the overlapsave method using the FFT, so that it can reduce the computational complexity to O(N log N + N) for N taps FIR filters. Some computer simulations show the faster convergence property of the proposed method than the conventional algorithms such as well-known LMS type techniques.

The group-delay sensitivity is studied theoretically for Gray and Markel allpass lattice filter realizing complex transfer function. Recursive expressions which were derived for phase and group-delay in real lattice are rederived for the complex case. These expressions are used to obtain upper bounds on group-delay sensitivity. The minimum number of frequencies where group-delay sensitivity becomes zero is discussed. Results corresponding real allpass lattice are also shown. Phase sensitivity properties of these filters are analyzed and compared with existing results. A new bound on phase sensitivity is also obtained.

This paper describes a new method of state extimation for the energy stochastic system with decibel observation mechanism. The problem here is to get the decibel-valued estimate of the energy state variable through the decibel-valued noisy observation data, where it is usual that the stochastic system is physically driven on energy scale. The main attention is paid to the adjustment between the energy quantity at the physical countermeasure side and the decibel quantity at the human evaluation side. The basic principle of state estimation is based on the Bayes' theorem which can be applicable to any non-Gaussian and/or non-linear nature of the real stochastic system. Then, it is expanded into the suitable form adapted to successive decibel-valued observation. Thus, based on the mutual relation between energy and decibel statistics, any kinds of statistics connected with L_x evaluation at the human side can be estimated by using this decibel-valued noisy observation data (L_x is defined in the acoustics field as the (100-x)% point of the sound-level distribution and it is often used as the environmental noise assessment standard because man's sense of hearing is very sensitive to the end of the sound-level distribution form). Finally, the validity and the effectiveness of the proposed method have been confirmed by application to the actually obtained room acoustics data.

A novel digital signal processing technique fuzzy filtering is proposed for estimating nonstationary signals with ambiguous changes, which are contaminated by additive white Gaussian noises. In this filter, fuzzy clustering is utilized for classifying signal components into groups in which the signal characteristics are considered to be similar. Since the boundary between the signal groups is ambiguous, the fuzzy clustering produces a better effect than crisp clustering. Moreover, robust characteristics are obtained for various values of the parameters and types of processed signals. Computer simulations successfully demonstrate its superior capability of filtering.

In a recent paper, we proposed a subband echo canceler (SBEC) structure that has a shorter transmission delay than conventional SBEC structures and is free of distortion due to frequency gaps or aliased components. The double-talk detector we propose in this paper, takes advantages of the subband realization form and is based on adaptive echo canceler operation even during periods of double-talk. The computationally simple structure quickly and accurately detects periods of double-talk and track variations in echo path characteristics. The extended complex LMS algorithm we also propose avoids distortion during periods of double-talk. Computer simulations using white noise and actual speech signals confirm fast tracking speed, accurate double-talk detection, and other desirable features of the proposed scheme. We evaluated the proposed hardware structure using a WE DSP32C development system. Assuming a 16 kHz sampling rate, a decimation factor of 32, and a 4000-tap echo path, we found 4 DSP chips to be sufficient to implement the proposed scheme. Results of our experiments show almost the same convergence performance as that obtained in computer simulation.

This paper presents a new method for implementing modem by using a periodically time varying digital filter. Firstly, we present that a modulation and a demodulation are realized by using a periodically time varying digital filter (abbreviated to a PTV filter). Next, we present that these functions of a modem can be implemented in one PTV filter. Generally, it is very complicated to search the shifted carrier frequency of a transfered modulated signal. Since only a PTV filter are used in the proposed system, we do not need to search the shifted carrier frequency in modem. So, the proposed method is better than the conventional method in this point.

The full band inverse fiter of an acoustic field usually can not be approximated by causal filter because of its non-minimum phase property. However, by dividing the full band signal into many sub-bands, most of the sub-bands, that do not have zeros extremely close to or outside of the unit circle, can be inverted by causal filter. In this paper, a method of recovering the reverberated speech signals by sub-band envelope estimation is proposed. A reference signal and its reverberated version are supposed to be known at the parameter estimation phase. Both signals are divided into many sub-bands. The dereverberation filter for each sub-band is calculated by the least mean square extimation of the reference sub-band signal from the reverberated one. Reverberated signals under the same condition can be recovered using the estimated dereverberation filters. Comparison of the dereverberation error and distribution of the room transfer function zeros showed that the sub-bands that have large dereverberation errors are those with zeros extremely close to or outside of the unit circle. The performance of the dereverberation is improved with the increase of the number of sub-bands. The reverberated speech is recovered in good quality when the reverberation time is less than 0.43s, which is approximately the reverberation condition of the general small office rooms.

A method to estimate picture quality by two-dimensional perceptive power using visual modulation transfer function including its oblique function was studied. First, visual modulation transfer functions measured conventionally were reviewed and its new formulae including spatial anistropy was presented. Using these formulae, picture quality estimation method was discussed. This method was verified by picture quality with horizontal signal band limitation. As the application of this method, line-offset sub-sampling for lower level family of digital television coding below 4:2:2, progressive scanning conversion for EDTV receiver and luminance resolution expansion through field-offset sub-sampling for EDTV were discussed and good efficiencies of this method for the signal processing design was shown.

It is well known that both 1-D and 2-D half-band (HB) filters are efficient filters suitable for sampling rate converters by a factor of 2. Two kinds of HB filters have been already proposed, i.e., HB FIR filters and HB IIR filters, and their design methods have been well established. HB FIR filters show symmetric amplitude characteristics, whereas HB IIR filters do not. This paper proposes a design method of new 2-D IIR digital filters whose amplitude characteristics are similar to that of 2-D HB FIR filters. We call such filters 2-D HB IIR filters here in this paper. First a design method of 1-D HB IIR filters is explained. Then a simple design method of 2-D HB IIR filters is shown which uses 1-D HB IIR filters as prototype filters. Two design examples of 2-D HB IIR filters indicate the validity of our approach. The attenuation of the proposed filter is approximately the same as that of the filter designed by Ansari. It is also shown that the proposed filter is able to realized the complete zero-phase characteristics. As a result, it is concluded that the filter is suitable for 2-D decimators by a factor of 2.

A challenge-and-response type identification protocol consists of three moves of messages between a prover and a verifier: Move-1--The prover claims to the verifier that his/her identity is ID. Move-2--The verifier challenges the prover with a question related to the ID. Move-3--The prover responds with the answer of the question. The verifier accepts the prover if the answer is correct. The main contribution of this paper is to show that the folklore can be made provably secure under the sole assumption of the existence of one-way functions.

A technique for measuring SAW velocity on an unprocessed wafer placed on a metal pattern of a matched filter made on a glass plate is reported. The impovements are (1) the use of a matched filter with different tap spacings, (2) the reduction of the output noise at the correlation peaks, and (3) realization of close contact between the wafer and the plate. The velocity measured on lithium niobate agrees with the value measured destructively, and the relative error is 250 ppm.

In this letter, we consider the following problem: Given a weighted digraph G=[V, E_aE_b] such that subgraph G_a=[V, E_a] of G is an acyclic graph with a single source vertex and the weight of each edge in E_b is negative, if G has a positive cycle, then locate them as many as possible; otherwise, compute the longest path length from source vertex to each vertex of V. Then, we propose an algorithm to this problem and show some experimental results to demonstrate the efficiency of the proposed algorithm.

Estimation of the element values of a minimal RLC network for a specified admittance function may be accomplished by reducing the network to one which can be synthesized by routine methods. There are generally many reduced networks yielding different sets of estimates. Experience shows that the overall estimate, obtained by averaging the values of each element, is the best set of estimates for the element values.

New OTA-based analog circuits for realizing fuzzy membership functions and maximum (MAX) and minimum (MIN) operations are proposed. The synthesis of these circuits based on a bounded-difference operation and their SPICE simulations are described.

This letter proposes a simple synthesis of a class of complex digital filters whose transfer functions are obtained from real transfer functions by substitution of e^-jθz for z. Such filters are constructed by simple circuitry transformations directly applied to real circuits.

We have discovered that a second-order piecewise-linear system exhibits "Lost Solution". It implies sudden change of oscillation amplitude due to slight change of a parameter. The analysis is performed by using piecewise exact solutions.