The concept of controlled resource sharing and dynamic quality of service (QoS) on the next generation Internet has attracted much attention recently. It is suggested that, by imposing real-time revision of shared resource allocated to individual media streams or data flows according to user/application QoS demand and resource availability, more balanced and efficient multimedia services can be provided. In this paper, we present an Adaptive Control Framework (ACF), which is developed for controlled resource sharing and dynamic QoS in real-time multimedia service. We discuss main elements of ACF including 1) Control schemes applicable in the framework, and 2) Control mechanisms used in ACF. It is clearly shown in this paper that, with control schemes and mechanisms incorporated in ACF and supportive algorithms and protocols for ACF applications on the Internet, more flexible service and better overall performance in terms of packet loss, latency, signal-noise ratio and re-synchronization delay, can be offered.

Newton based adaptive algorithms are among the algorithms which are known to exhibit a higher convergence speed in comparison to the least mean square (LMS) algorithms. In this paper we propose a simplified Newton based adaptive algorithm for an adaptive infinite impulse response (IIR) filter implemented using cascades of second order allpass filters and a finite impulse response (FIR) filter. The proposed Newton based algorithm avoids the complexity that may arise in the direct differentiation of the mean square error. The analysis and simulation results presented for the algorithm, show that the property of convergence of the poles of the IIR ADF to those of the unknown system will be maintained for both white and colored input signal. Computer simulation results confirm an increase in convergence speed in comparison to the LMS algorithm.

In this paper, we propose an adaptive encoding method of fixed codebook in CELP coders and implement an adaptive fixed code-excited linear prediction (AF-CELP) speech coder as a low-bit-rate extension to the 8 kbit/s CS-ACELP. The AF-CELP can be implemented at low bit rates as well as low complexity by exploiting the fact that the fixed codebook contribution to the speech signal is periodic, as is the adaptive codebook (or pitch filter) contribution. Listening tests show that the 6.4 kbit/s AF-CELP has a comparable quality to the 8 kbit/s CS-ACELP under real environmental test conditions.

A software system has dynamic adaptability if it can adapt itself to dynamically changing runtime environments. As open-ended distributed systems and mobile computing systems have spread widely, the need for software systems with dynamic adaptability increases. We propose a software model with dynamic adaptability called DAS and its description language LEAD++. The basic mechanism for dynamic adaptability is called adaptable procedure. An adaptable procedure is a special kind of generic procedures (functions) whose methods are selected based upon the state of its runtime environment. Furthermore, control mechanisms of adaptable procedures -- including method selection strategies -- are realized using generic procedures. This sort of reflective architecture enables us to write a dynamically adaptable software system in highly flexible, extensible, readable and maintainable way. LEAD++ is an object-oriented reflective language that provides adaptable procedures and their control mechanisms as its basic language functionalities. We are currently implementing a prototype of LEAD++ as a pre-processor of Java. Using LEAD++, we can systematically describe dynamically adaptable applets, mobile objects, etc.

In this paper, quasi-Gaussian filter, quasi-median filter and locally adaptive filters are introduced. A new adaptive vector filter based on noise estimate is proposed to suppress Gaussian and/or impulse noise. To estimate the type and degree of noise corruption, a noise detector and an edge detector are introduced, and two key parameters are obtained to characterize noise in color image. After globally estimating the type and degree of noise corruption, different locally adaptive filters are properly chosen for image enhancement. All noisy images, used to test filters in experiments, are generated by PaintShopPro and Photoshop software. Experimental results show that the new adaptive filter performs better in suppressing noise and preserving details than the filter in Photoshop software and other filters.

This paper discusses the classification of targets buried in the underground by radar polarimetry. The subsurface radar is used for the detection of objects buried beneath the ground surface, such as gas pipes, cables and cavities, or in archeological exploration operation. In addition to target echo, the subsurface radar receives various other echoes, because the underground is inhomogeneous medium. Therefore, the subsurface radar needs to distinguish these echoes. In order to enhance the discrimination capability, we first applied the polarization anisotropy coefficient to distinguish echoes from isotropic targets (plate, sphere) versus anisotropic targets (wire, pipe). It is straightforward to find the man-made target buried in the underground using the polarization anisotropy coefficient. Second, we tried to classify targets using the polarimetric signature approach, in which the characteristic polarization state provides the orientation angle of an anisotropic target. All of these values contribute to the classification of a target. Field experiments using an ultra-wideband (250 MHz to 1 GHz) FM-CW polarimetric radar system were carried out to show the usefulness of radar polarimetry. In this paper, several detection and classification results are demonstrated. It is shown that these techniques improve the detection capability of buried target considerably.

A new motion analysis method and an image restoration process for removing motion blur are proposed. Motion analysis includes the motion estimation and motion-based segmentation. Based on the analysis, we can obtain an image divided into multiple segments with different point spread functions. For removing motion blur, we propose an image degradation model for the motion with an arbitrary direction and a regularized iterative restoration method. By using the proposed degradation model and the restoration method, we can efficiently remove the space-variant motion blur.

The current Internet does not offer any quality of service guarantees or support to Internet multimedia applications such as Internet telephony and video-conferencing, due to the best-effort nature of the Internet. Their performance may be adversely affected by network congestion. Also, since these applications commonly employ the UDP transport protocol, which lacks congestion control mechanisms, they may severely overload the network and starve other applications. We present an overview of recent research efforts in developing adaptive delivery models for Internet multimedia applications, which dynamically adjust the transmission rate according to network conditions. We classify the approaches used to develop adaptive delivery models with brief descriptions of representative research work. We then evaluate the approaches based on important design issues and performance criteria, such as the scalability of the control mechanism, responsiveness in detecting and reacting to congestion, and ability to accommodate receiver heterogeniety. Some conclusions are developed regarding the suitability of particular design choices under various conditions.

In this paper, we propose a new adaptive control system design using internal model principle (IMP) for a bounded polynomial parameters. In this method, we regard time varying parameters as variable disturbance and design an estimating law used the internal model of the disturbance so that the law is able to rejected the effectness of the disturbance. Our method has the features that the tracking error can converge to zero. Furthermore, we give a sufficient condition for the stability based on a small-gain theorem. The condition shows that our proposed method relax the stability condition more than the conventional methods based on a passivity theorem. Finally, we contain a numerical simulation to show an effect of our system.

An efficient ARQ scheme based on the packet combining technique is investigated for multi-carrier modulation systems. In multi-carrier modulation systems, several sub-carriers are used for high data rate transmission and their individual received signal quality becomes different from one sub-carrier to others in a frequency selective fading channel. Therefore by changing the assignment of data to the sub-carriers in the retransmission packets, the distortion between the previous transmitted packet and the newly retransmitted one will be different. This is the principle of the proposed adaptive data order rearrangement for a packet combining ARQ scheme, which can achieve more diversity gain in packet combining and improve the ARQ performance. From the results of the theoretical analysis and the computer simulation, it is confirmed that the proposed packet combining ARQ with the proposed operation can achieve the better performance in terms of the average packet transmission success probability. In addition, this proposed scheme is also compared with the conventional multi-carrier modulation ARQ scheme based on the partial retransmission of a packet. The computer simulation results demonstrate that the proposed scheme has also advantage against the latter one, and it is considered to be as a more efficient ARQ scheme for multi-carrier modulation systems.

This paper proposes a phase-rotating phase-shift keying (PSK) modulation and shows that its narrow-band version is suitable for Viterbi equalization. The proposed PSK has the following features: 1) a spectrum shaping of the transmit/receive filters does not need to be restricted to the Nyquist criterion; 2) the transmitted data sequence is rotated for every symbol in order to reduce noise-correlation at the receiver. First, this paper discusses a performance degradation of bit error rate of Viterbi equalizers in the presence of the sampling timing offset or under time-dispersive frequency selective fading. Next, computer simulation confirms that π/2-shifted binary PSK with narrow-band spectrum shaping filter, which includes offset QPSK for its special case, solves the above mentioned performance degradation, keeping good spectrum efficiency equal to M-ary PSK.

In this paper, an adaptive method for active control of vibration intensity in the frequency domain is proposed. In this method, vibration intensity is observed with the 4-sensor method, and the coefficients of an adaptive FIR filter for the active control is renewed with the Block Filtered-X LMS algorithm in the frequency domain. An experiment with the proposed method is performed on a simple model. As a result, the proposed method gives larger attenuation of vibration intensity than the conventional method in the high frequency region. The overall attenuation in vibration intensity in that frequency region is 14.1 dB with the proposed method, while it is 7.0 dB with the conventional method. In the lower frequency region, the reduction in vibration intensity by the proposed method is roughly equivalent to that obtained by the conventional method. An improvement may also be achieved there by setting the intervals between error sensors properly.

This paper proposes a new robust adaptive beamformer applicable to microphone arrays. The proposed beamformer is a generalized sidelobe canceller (GSC) with a variable blocking matrix using coefficient-constrained adaptive filters (CCAFs). The CCAFs, whose common input signal is the output of a fixed beamformer, minimize leakage of the target signal into the interference path of the GSC. Each coefficient of the CCAFs is constrained to avoid mistracking. In the multiple-input canceller, leaky adaptive filters are used to decrease undesirable target-signal cancellation. The proposed beamformer can allow large look-direction error with almost no degradation in interference-reduction performance and can be implemented with a small number of microphones. The maximum allowable look-direction error can be specified by the user. Simulation results show that the proposed beamformer, when designed to allow about 20of look-direction error, can suppress interference by more than 17 dB.

A new method of designing a microphone array with two outputs preserving binaural information is proposed in this paper. This system employs adaptive beamforming using multiple constraints. The binaural cues may be preserved in the two outputs by use of these multiple constraints with simultaneous beamforming to enhance target signals is also available. A computer simulation was conducted to examine the performance of the beamforming. The results showed that the proposed array can perform both the generation of the binaural cues and the beamforming as intended. In particular, beamforming with double-constraints exhibits the best performance; DI is around 7 dB and good interchannel (interaural) time/phase and level differences are generated within a target region in front. With triple-constraints, however, the performance of the beamforming becomes poorer while the binaural information is better realized. Setting of the desired responses to give proper binaural information seems to become critical as the number of the constraints increases.

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.

Template learning has potential application in several areas of Cellular Neural Network research, including texture recognition, pattern detection and so on. In this letter, a recently-developed algorithm called Adaptive Simulated Annealing is investigated for learning CNN templates, as a superior alternative to the Genetic Algorithm.

Lots of self-tuning control schemes have been proposed for tuning the parameters of control systems. Among them, pole-assignment schemes have been widely used for tuning the parameters of control systems with unknown time delays. They are usually classified into two methods, the implicit and the explicit methods according to how to identify the parameters. The latter has an advantage to design a control scheme by taking account of the stability margin and control performance. However, it involves a considerably computational burden to solve a Diophantine equation. A simple scheme is proposed in this paper, which can construct a multivariable self-tuning pole-assignment control system, while taking account of the stability margin and control performance without solving a Diophantine equation.

Wavelength Division Multiplex (WDM) photonic networks are expected as key for global communication infrastructure. The accurate measurement methods for AWG-MUX/DMUX are desirable for WDM network design. We measured a transfer function matrix of an AWG-MUX to find that polarization mode dispersion (PMD) and polarization dependent loss (PDL) shows the bandpass characteristics, which may limit the maximum size and the bit rate of the system. These bandpass characteristics of PMD and PDL are reproduced by a simple AWG-MUX model: The phase constant difference of 0.5% between orthogonal modes in arrayed waveguides is sufficient to obtain the measured passband characteristics of PMD and PDL. We find phase distribution difference between two orthogonal modes in the arrayed waveguide grating gives arise to complex PMD.

Wavelength Division Multiplex (WDM) photonic networks are expected as key for global communication infrastructure. The accurate measurement methods for AWG-MUX/DMUX are desirable for WDM network design. We measured a transfer function matrix of an AWG-MUX to find that polarization mode dispersion (PMD) and polarization dependent loss (PDL) shows the bandpass characteristics, which may limit the maximum size and the bit rate of the system. These bandpass characteristics of PMD and PDL are reproduced by a simple AWG-MUX model: The phase constant difference of 0.5% between orthogonal modes in arrayed waveguides is sufficient to obtain the measured passband characteristics of PMD and PDL. We find phase distribution difference between two orthogonal modes in the arrayed waveguide grating gives arise to complex PMD.

A pipelined architecture is proposed for the normalized least mean square (NLMS) adaptive digital filter (ADF). Pipelined implementation of the NLMS has not yet been proposed. The proposed architecture is the first attempt to implement the NLMS ADF in the pipelined fashion. The architecture is based on an equivalent expression of the NLMS derived in this study. It is shown that the proposed architecture achieves a constant and a short critical path without producing output latency. In addition, it retains the advantage of the NLMS, i. e. , that the step size that assures the convergence is determined automatically. Computer simulation results that confirm that the proposed architecture achieves convergence characteristics identical to those of the NLMS.