We present a speaker adaptation method that makes it possible to determine articulatory parameters from an unknown speaker's speech spectrum using an HMM (Hidden Markov Model)-based speech production model. The model consists of HMMs of articulatory parameters for each phoneme and an articulatory-to-acoustic mapping that transforms the articulatory parameters into a speech spectrum for each HMM state. The model is statistically constructed by using actual articulatory-acoustic data. In the adaptation method, geometrical differences in the vocal tract as well as the articulatory behavior in the reference model are statistically adjusted to an unknown speaker. First, the articulatory parameters are estimated from an unknown speaker's speech spectrum using the reference model. Secondly, the articulatory-to-acoustic mapping is adjusted by maximizing the output probability of the acoustic parameters for the estimated articulatory parameters of the unknown speaker. With the adaptation method, the RMS error between the estimated articulatory parameters and the observed ones is 1.65 mm. The improvement rate over the speaker independent model is 56.1 %.

We present a novel adaptive method to improve the binarization quality of degraded word color images. The objective of this work is to solve a nonlinear problem concerning the binarization quality, that is, to achieve edge enhancement and noise reduction in images. The digitized data used in this work were extracted automatically from real world photos. The motion of objects with reference to static camera and bad environmental conditions provoked serious quality problems on those images. Conventional methods, such as the nonlinear adaptive filter method proposed by Mo, or Otsu's method cannot produce satisfactory binarization results for those types of degraded images. Among other problems, we note mainly that contrast (between shapes and backgrounds) varies greatly within every degraded image due to non-uniform illumination. The proposed method is based on the automatic extraction of background information, such as luminance distribution to adaptively control the intensity levels, that is, without the need for any manual fine-tuning of parameters. Consequently, the new method can avoid noise or inappropriate shapes in the output binary images. Otsu's method is also applied to automatic threshold selection for classifying the pixels into background and shape pixels. To demonstrate the efficiency and the feasibility of the new adaptive method, we present results obtained by the binarization system. The results were satisfactory as we expected, and we have concluded that they can be used successfully as data in further processing such as segmentation or extraction of characters. Furthermore, the method helps to increase the eventual efficiency of a recognition system for poor-quality word images, such as number plate photos with non-uniform illumination and low contrast.

This paper presents time slot assignment algorithms applicable to uplink of space division multiple access (SDMA)/time division multiple access (TDMA) systems with adaptive antennas. In the time slot assignment process for a new terminal in a cell, we consider not only the signal quality of the new terminal but also that of active terminals in the same cell. Intra-cell hand over is performed for an active terminal when its signal quality deteriorates. We evaluate the blocking and forced termination probabilities for pure TDMA systems, sectorized systems, and SDMA/TDMA systems in cellular environments by computer simulations. The simulation results show that the SDMA/TDMA systems have much better performance than the pure TDMA and sectorized systems.

This paper presents an imaging technique using the MUSIC algorithm to localize cylindrical reflectors in cross-borehole radar arrangements. Tomographic measurement, in which a transmitting and a receiving antenna are individually moved in separate boreholes, can be considered as a combination of a transmitting and a receiving array. A decorrelation technique with the transmitting array, which has been proposed for imaging point reflectors, is applied for imaging cylindrical reflectors using the MUSIC algorithm. Simulated and experimental results are shown to verify the validity of this algorithm for cylindrical targets. We analyze the evaluation error caused by the increase in the radius of the cylinder.

In this letter, the concept of cross-entropy is introduced for unsupervised polarimetric synthetic aperture radar (SAR) image classification. The difference between two scatterers is decomposed into three parts, i.e., the difference of average scattering characteristic, the difference of scattering randomness and the difference of scattering matrix span. All these three parts are expressed in cross-entropy formats. The minimum cross-entropy principle is adopted to make classification decision. It works well in unsupervised terrain classification with a NASA/JPL AIRSAR image.

Adaptive antenna array is a promising technique to increase the link capacity in mobile radio communications systems by suppressing multiple access interference (MAI). In the mobile radio, the received signal consists of discrete paths, each being a cluster of many irresolvable paths arriving from different directions. For large arrival angle spread of each cluster of irresolvable paths, antenna array cannot form a beam pattern that sufficiently suppresses MAI even in the presence of single interference signal and hence, the transmission performance may degrade. In this situation, the use of antenna diversity may be a better solution. It is an interesting question as to which can achieve a better performance, antenna diversity reception or adaptive antenna array. In this letter, we study the impact of the arrival angle spread on the DS-CDMA transmission performances achievable with adaptive antenna array and antenna diversity reception. It is pointed out that the arrival angle spread is an important parameter to determine the performances of adaptive antenna array and antenna diversity.

The cable length in wired serial data communication is limited because the limited bandwidth of a long cable introduces ISI (Inter Symbol Interference). A line equalizer can be used at the receiver to extend the channel bandwidth. This paper proposes a low-power and small-area analog adaptive line equalizer for 100-Mb/s operation on UTP (Unshielded Twisted Pair) cable up to 100 m. The proposed adaptive line equalizer is fabricated with 0.35-µm CMOS process, consumes 19 mW and occupies only 0.07 mm2 Measurement results show that the prototype can operate at data rate of 100 Mb/s on a 100-m cable and 155 Mb/s on a 50-m cable.

A sequentially and linearly constrained minimum variance beamformer (SLCMV), based on a split polarity transformation (SPT) and, called an SPT-SLCMV beamformer, is proposed to minimize the degree of freedom loss, steering error, and a desired signal elimination phenomenon under coherent interferences. The SPT-SLCMV beamformer reduces the degree of freedom loss, which is inevitable in the conventional SPT-LCMV beamformer, by successively applying sub-constraint matrices. Sub-constraint matrices are derived from a complete constraint matrix to remove the correlation between the desired signal and interferences. In addition, the SPT-SLCMV beamformer is combined with the iterative steering error correction method to reduce the steering error between the look direction of the beamformer and the incident angle of the desired signal. As a result, the proposed beamformer reduces the number of array elements while maintaining the performance of an exactly steered SPT-LCMV beamformer having sufficient array elements under coherent interferences.

We investigated the accuracy of nickel-cadmium (Ni/Cd) battery degradation estimation by measuring the capacity of over 400 used cordless-telephone batteries using a discharge-current-pulse technique. The capacity is calculated from the change in battery voltage after the current pulse is applied, using an equation that we developed. Battery degradation is represented by a percentage of the capacity based on the nominal one. To estimate the accuracy of the degradation estimation, we compare capacity Qe estimated from the current pulse with the capacity Qa measured by discharging the batteries. The Qe estimated from the current pulse was within a range of 20% of error indicated by (Qe-Qa) for 47% of the tested batteries. The Qe of 51% of the batteries, however, was underestimated and exceeded lower limit (-20%) of the error. One reason for the discrepancy could be that the equation is inadequate for estimating the capacity from the current pulse. On the other hand, the capacity Qe of 1% of the batteries was overestimated and exceeded upper limit (+20%) of the error. An internal short is probably the main reason for this.

The new Internet has to provide the Quality of Services to converged multimedia services, in which each one may choose its own requirements. Managing such a dynamic network is not an easy task. A more intelligent and adaptive behavior is required from network management. We argue that agents are able to realize this task by dynamically adapting management mechanisms to the current network conditions. This article presents a Behavioral Multi-Agent-based model for QoS-enabled Internet. Based on this behavioral approach, we analyze network management mechanisms (or "elementary behaviors") in terms of performance and applicability profile. We use simulation to observe services performances when submitted to diverse QoS management elementary behaviors.

A new algorithm and architecture to eliminate redundant operations in block-matching (BM) motion estimation is proposed. The key step of this work is to use binary-matching to define image regions with the static background content and then exclude these regions from the actual motion estimation. According to experiments, the approach maintains the highest PSNR, while making as half as less computations in comparison to the adaptive BM or 1/8 of the computations required by the full-search BM. An implementation scheme is outlined.

The main issue in real time voice applications over Internet is to investigate a lossless playout without jitter while maintaining playout delay as small as possible. Existing playout algorithms estimate network delay by using timestamps and determine the playout schedule only at the beginning of each talkspurt. Also their scheduled playout time is determined based on a fixed upper playout delay bound over a talkspurt. The sliding adaptive playout algorithm we propose can estimate jitter without using timestamps and its playout time is allowed to slide to a certain extent. The aim of sliding playout schedule is to determine the scheduled playout time at the beginning of each talkspurt based on the playout delay expected under the normal condition where the degree of actual jitter is below the magnitude which is not quite large in relation to variations in the "baseline" delays. Then the proposed algorithm can be effectively applied to minimize the scheduled playout delay while improving the voice quality against a spike which may occur at the beginning of a talkspurt as well as a spike which occurs in the middle of a talkspurt. We develop an analytical model of the general adaptive playout algorithms and analyze the playout buffer performance for different degrees of jitter, different values of the scheduled playout delay, different maximum lengths of delay spikes, and arbitrary tolerable ranges of sliding. Based on the analytical results, we suggest the specific values of parameters used in the sliding algorithm.

This paper presents a scalable and efficient quota-based admission control scheme for the 3rd generation (3G) operator's IP backbone network, where quota denotes a chunk of bandwidth. This research is motivated by the 3G operator's need for guaranteeing end-to-end IP QoS of mobile-to-mobile and mobile-to-server multimedia sessions. In the proposed scheme, the quota size of a path implies the proper amount of allocated and released resources on the path condition. Employing the quota size makes the job of allocating or releasing resources at nodes in a path simple so that it becomes scalable. Moreover, with this simple scheme, an edge node can be allowed not only to initiate the allocation/release request but also to perform admission control function. To maximize the efficiency, the path quota size varies depending on the bottleneck link condition in the path. In high offered load, the proposed scheme decreases the path quota size and retains higher utilization while it requires lower signaling cost than the fixed scheme using a fixed size aggregation. As the load lessens, it increases the path quota size and reduces the signaling cost significantly.

Online and realtime traffic summarization is a challenge as, except for the routine cases, aggregation parameters or, the flows that need to be observed are not known a priori. Dynamic adaptive aggregation algorithms adapt to the network traffic to detect the important flows. But present day algorithms are inadequate as they often produce inaccurate or meaningless aggregates. In this work we propose a Dynamic Constrained Adaptive Aggregation algorithm that does not produce the meaningless aggregates by using information about the network's configuration. We compare the performance of this algorithm with the erstwhile Dynamic (Unconstrained) Adaptive Aggregation algorithm and show its efficacy. Further we use the network map context that shows the network flows in an intuitive manner. Several applications of the algorithm and network map based visualization are discussed.

This paper presents a computationally efficient subspace-based method for partially adaptive beamforming which is based on the structure of the generalized sidelobe canceller (GSC). Its auxiliary beamformer operates in an estimated interference subspace which is obtained through simple computation. The computational burden of the proposed method in terms of complex multiplication is just on O(η2M) where η and M are the numbers of interferences and the array elements, respectively. Though the subspace obtained is different from the exact interference subspace due to the presence of noise, theoretical analysis shows that the proposed beamfomer virtually attains the optimal performance for strong or sidelobe interference. Simulation results validate its effectiveness including fast convergence, even in the presence of errors in the detected number of directional signals.

This paper presents a cost-effective CORDIC-based architecture for adaptive lattice filters. An implementation method for an ARMA lattice filter using the CORDIC algorithm has been proposed. The previously proposed method can provide a simple filter architecture; however, it has problems such as redundant structure and numerical inaccuracy. Therefore, by solving each problem we derive a new non-redundant filter architecture with improved numerical accuracy. The obtained filter architecture provides a low cost ARMA lattice filter in which high-precision data processing is feasible. In addition, the proposed architecture can be applied to AR-type lattice filters, so that it may have several applications in adaptive signal processing. The presented filter architecture is useful from a hardware point of view because it facilitates an effective VLSI design of various adaptive lattice filters.

This paper presents a method to compensate the effect of non-cancellable zeros of the plant on output by incorporating finite impulse response (FIR) filter in the feed-forward path. This FIR filter is designed using the approximate inverse system of the non-cancellable part of numerator polynomial of the plant. The proposed controller guarantees boundedness of all signals from the reference input to the plant output. Computer simulation and real-time control experiment results are given to demonstrate the effectiveness of the proposed method.

Range resolution for radar is determined by bandwidth. One of the various super-resolution techniques for improving resolution is bandwidth extrapolation (BWE). In this technique, a linear prediction model is fitted to the data, and the model is used to extrapolate the bandwidth. In this paper, we extend BWE, and propose a new algorithm that we call polarimetric bandwidth extrapolation (PBWE) applicable to polarimetric radar data. We show through numerical simulations that utilization of fully polarimetric information allows PBWE to improve the resolution beyond the conventional BWE method. Some results of a physical simulation experiment using a W-band polarimetric FMCW radar and corner reflectors are shown to confirm the advantage of PBWE.

In this letter, we present an adaptive hopping technique for a wireless personal area network (WPAN) system employing a frequency hop spread spectrum (FH/SS). Analytical results based on the closed-form solutions for the aggregate throughput show that the proposed hopping algorithm using two defined hopping criteria is more friendly towards all kinds of interferers and gives an enhanced throughput with a moderate computational complexity.

In this letter, a fixed-power variable-rate turbo coded modulation scheme for orthogonal frequency division multiplexing (OFDM) is proposed and subband adaptation algorithm based on capacity evaluation is presented. Our object is to improve the overall throughput under target bit-error-rate (BER). Simulation results show that our adaptive OFDM scheme exhibits an about 2.5-5 dB signal-noise-ratio (SNR) gain with target BER of 10-4 and subband number of 16 relative to fixed threshold adaptive turbo coded modulation. Moreover, unlike fixed threshold adaptation, with the number of subband decreased, throughput performance is not degraded due to our capacity evaluation algorithm.