This paper describes modifications to a previously proposed 8-kb/s 4-ms-delay CELP speech coding algorithm with a view to improving the speech quality while maintaining low delay and only moderately increasing complexity. The modifications are intended to improve the effectiveness of interframe pitch lag prediction and the sub-optimality level of the excitation coding to the backward adapted synthesis filter by using delayed decision and joint optimization techniques. Results of subjective listening tests using Japanese speech indicate that the coded speech quality is significantly superior to that of the 8-kb/s VSELP coder which has a 20-ms delay. A method that reduces the computational complexity of closed-loop 3-tap pitch prediction with no perceptible degradation in speech quality is proposed, based on representing the pitch-tap vector as the product of a scalar pitch gain and a normalized shape codevector.

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.

An adaptive digital filter with adaptive sampling phase is proposed. The structure of the filter makes use of an adaptive delay device at the input of the filter. The algorithm is derived to determine the value of the delay and the filter coefficients by minimizing MSE (mean square error) between the desired signal and the filter output. The computer simulation of the convergence of the proposed adaptive filter with the input of sinusoidal wave and BPSK modulated wave are shown. According to the simulation, the MSE of the proposed adaptive delay algorithm is lower than that of the conventional LMS algorithm.

This paper discusses advanced wireless communication technologies for achieving future high-speed mobile radios. Mainly, five technical fields are considered, that is, multi-level modulation for transmitting high-capacity information signal, advanced adaptive wireless system flexibly changing modulation level, symbol rate and traffic according to fading conditions, adaptive multicarrier system transmitting multimedia signals by changing the number of carrier according to the capacity of the signals, new CDMA techniques for mapping different bit rate services onto the same allocated bandwidth at the same time, and optical-linked microcellular communication system with millimeter wave air interface.

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.

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 test system with a fuzzy logic controller is proposed to assure stable outgoing quality as well as to raise throughput. The test system controls the number of items under test in accordance with fuzzy information as well as statistical information about incoming quality and outgoing quality. First, an algorithm, minimum-minimum-the center of gravity-weighted mean method, is studied with both fuzzy reasoning rules and membership functions which are used for the control. Second, characteristics of the test system are verified and examined with computer simulations so that the fuzzy logic control rules are determined to realize sufficient sensitivity to process changes. Third, the control rules are installed in the test management processor which commands test equipment for testing very large scale integrated circuits, with programming language C. The authors have obtained satisfactory results through a trial run using a series of lots of 16 bit micro controller units in an IC manufacturing factory. Finally, they study the stability condition of the fuzzy test system.

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.

This paper investigates a different method of speaker adaptation for Mandarin syllable recognition. Based on the minimum classification error (MCE) criterion, we use the generalized probabilistic decent (GPD) algorithm to adjust interatively the parameters of the hidden Markov models (HMM). The experiments on the multi-speaker Mandarin syllable database of Telecommunication Laboratories (T.L.) yield the following results: 1) Efficient speaker adaptation can be achieved through discriminative training using the MCE criterion and the GPD algorithm. 2) The computations required can be reduced through the use of the confusion sets in Mandarin base syllables. 3) For the discriminative training, the adjustment on the mean values of the Gaussian mixtures has the most prominent effect on speaker adaptation. 4) The discriminative training approach can be used to enhance the speaker adaptation capability of the maximum a posteriori (MAP) approach.

This paper describes a novel speaker-independent speech recognition method, called speaker-consistent parsing", which is based on an intra-speaker correlation called the speaker-consistency principle. We focus on the fact that a sentence or a string of words is uttered by an individual speaker even in a speaker-independent task. Thus, the proposed method searches through speaker variations in addition to the contents of utterances. As a result of the recognition process, an appropriate standard speaker is selected for speaker adaptation. This new method is experimentally compared with a conventional speaker-independent speech recognition method. Since the speaker-consistency principle best demonstrates its effect with a large number of training and test speakers, a small-scale experiment may not fully exploit this principle. Nevertheless, even the results of our small-scale experiment show that the new method significantly outperforms the conventional method. In addition, this framework's speaker selection mechanism can drastically reduce the likelihood map computation.

This paper proposes two algorithms for defining a routing domain in multiclass-of-service networks. One an off-line-based method, whose objective is to optimize dynamic routing performance by using precise knowledge on the traffic levels. The algorithm of the proposed method takes into account the random nature of the traffic flow, which is not considered in the network flow approach. The proposed method inherits the conceptual simplicity of the network flow approach and remains applicable to large and complex networks. In simulation experiments, the proposed off-line-based method performs better than the method based on the network flow approach, but has a similar the computation time requirement. The other method proposed here is an on-line-based method for application to B-ISDNs, where precise traffic data is not expected to be available. In this method, the routing domain is defined adaptively according to the network performance (call-blocking probability) measured in real-time. In simulation experiments, the performance of this method is comparable to that of the off-line-based method--especially when highly efficient dynamic routing is used. This paper also derives and describes methods for approximating the implied costs for multiclass-of-service networks. The approximations are very useful not only for off-line-based routing domain definition (RDD) methods but also for other kinds of network controls or optimal network dimensioning based on the concept of revenue optimization.

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.

This paper describes a fully digitized modem designed for variable baud rate transmission systems with the aim of efficiently providing multimedia services over a wireless communication network. The concept of a variable baud rate wireless communication system is discussed focusing on the access scheme and channel allocation from the viewpoint of frequency utilization efficiency. For easy system construction, we propose a fully digitized variable baud rate modem based on multirate digital signal processing, taking into account the need for even performance and easy clock control for all transmission rates. We also discuss the operational principle of modulation, the degradation factor in the A/D converter, and the configuration of the clock recovery circuit. Steady modulation performance can be kept by generating the same frequency system clock for all transmission rates and using the sampling rate conversion technique without selecting the channel filter for each transmission rate. It is proved by the analysis of the degradation factor in the A/D converter that only the bandwidth of the channel filter in demodulator should be changed for the transmission rate. A double loop clock recovery configuration capable of both tank-limit type and baseband estimation type clock recovery is shown to be suitable for this system. The tank-limit clock recovery circuits can be constructed easily by employing a tank circuit array. Finally, we present experimental results for a modem having transmission rates of 1.544Mbps and 6.312Mbps for the digital hierarchy and information speed of video signals such as MPEG-1 and MPEG-2. The measured basic performance of the proposed modem shows it delivers superior performance without the need for precise adjustment when a QPSK modulation scheme is employed.

In this paper, the computational model of Quasi-Ecosystem that is constructed in the way of bottom up, i.e., that consists of herbivores, carnivores and plants is proposed and the simulation result is shown. The behavior pattern of the model is represented by finite state automata. Simple adaptive behavior of animals was observed in this simulation. This indicates that mutation is effective method for self adaptive behavior and the possibility that the model can be used as a framework for autonomous agents.

Circuit design techniques for linearizing adaptively biased differential pairs are described. An emitter-and source-coupled pair is adaptively biased by a squaring circuit to linearize its transconductance, one of whose inputs is divided by resistors. An input signal for a differential pair or a squaring circuit is set to an adequate amplitude by a resistive divider without sacrificing linearity. Therefore, a differential pair is biased by the output current of a squaring circuit and they are coupled directly. There are three design techniques for squaring circuits. One is the transistor-size unbalance technique. Another is the bias offset technique. A third is the multitail technique. The bipolar and MOS squaring circuits discussed in this paper were proposed by the author previously, and consist of transistor-pairs with different transistor size (i.e., the emitter areas or gate W/L values are different), transistor-pairs with the same bias offset, or a multitail cell(i.e., a triple-tail cell or quadritail cell). Several kinds of squaring circuits consisting of such transistor-pairs are applied to produce the quadratic bias currents for compensating the nonlinearity of an emitter-and source-coupled pair. Therefore, four circuits using emitter-coupled pairs with adaptive-biasing current and four circuits using source-coupled pairs with adaptive-biasing current are proposed and analyzed in depth. Furthermore, a circuit configuration for low voltage operation is also introduced and verified with bipolar transistor-arrays on a breadboard.

This paper presents an experimental result of polarimetric detection of objects buried in a sandy ground by a synthetic aperture FM-CW radar. Emphasis is placed on the reduction of surface clutter by the polarimetric radar, which takes account of full polarimetric scattering characteristics. First, the principle of full polarimetric imaging methodology is outlined based on the characteristic polarization states for a specific target together with a polarimetric enhancement factor which discriminates desired and undesired target echo. Then, the polarimetric filtering technique which minimizes a surface reflection is applied to detect a thin metallic plate embedded in a sandy ground, demonstrating the potential capability of reducing surface clutter which leads to an improvement of underground radar performance, and validating the usefulness of FM-CW radar polarimetry.

In improving channel utilization in microcellular systems, adaptive channel allocation using distributed control has been reported to be effective. We describe an analytical approximation algorithm for channel dimensioning of distributed adaptive channel allocation. We compare our analytical results with simulation results and show the characteristics of permissible load as a function of the number of base station channels based on our method. Finally we illustrate traffic design and administration based on our algorithm.

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.

In this paper, we propose the media scheduler employing an adaptive estimator, which uses a posteriori information of data traffic characteristics to facilitate scheduling, when available, to provide on-line scheduling of dynamic scene change based on its statistical characteristics. Especially, a new adaptive scheduling scheme showed good persistent to the arrival message with bursty characteristics. And we confirmed the performance through the computer simulation when QOS requirements are given.

The effects of Gaussian electromagnetic noise and non-Gaussian one on TV picture degradation are studied by using a composite noise generator which can control noise parameters. Three kinds of still pictures and four kinds of motion pictures are tested, and the picture degradation is subjectively evaluated with five-grade impairment scale. The tendency of the picture degradation against the every picture is almost the same. But MOS (Mean Opinion Score) between still picture and motion picture degradation is different in some measure when the power of burst noise is small.