This letter proposes a time varying step size normalized LMS (TVS-NLMS) algorithm for adaptive echo canceler structures. Proposed algorithm reduces distortion during double talk, without increasing the computational cost nor decreasing the convergence rate of the normalized LMS algorithm significantly. Simulation results using white noise and actual speech signals confirm the desirable features of the proposed scheme.

An excitation signal for a synthesis filter plays an important role in producing high quality speech at a low bit rate. This paper presents a new efficient excitation model, Adaptive Density Pulse (ADP) , for low bit-rate speech coding. This ADP is a pulse train whose density (spacing interval) is constant within a subframe but can be varied subframe by subframe. First, the ADP excitation signal is defined. A procedure for finding the optimal ADP excitation is presented. Some results on investigating the effects of the ADP parameters on the synthesized speech quality are discussed. ADP excitation is introduced to the CELP (Code Excited Linear Prediction) coding method to improve speech quality at bit rates around 4 kbps. A CELP coder with an ADP (ADP-CELP) is described. ADP excitation makes it possible for the CELP coder to follow transient portions of speech signals. Also ADP excitation can reduce computational complexity in selecting the best excitation from a codebook, which has been the primary drawback of CELP. The number of multiplications can be reduced to the order of 1/D2 by utilizing the sparseness of ADP excitation, where D is the pulse interval. The authors evaluated the speech quality of a 4 kbps ADP-CELP coder by computer simulation. ADP excitation improved the performance of conventional CELP in segmental SNR.

Let L{0,1}* be a language and let λL : {0,1}*{0,1} be the characteristic function of the language L, i.e., if x ∈ L, λL (x) = 1; otherwise,λL (x) = 0. In this paper, we consider an adaptive checker with a single program F (resp. noncommunicating multiple programs F1, F2,...) for λL that works even when an incorrect program F* (resp. incorrect noncommunicating multiple programs F*1,F*2,...) for λL adaptively behaves according to inputs previously provided to the program F* (resp. the programs F*1,F*2,...). We show that (1) for any language L, there exists an adaptive checker with a single program for λL iff L and respectively have competitive interactive proof systems; and (2) that for any language L, there exists an adaptive checker with noncommunicating multiple programs for λL iff L and respectively have function-restricted interactive proof systems. This implies that for any language L, adaptive chekers with noncommunicating multiple programs for λL are as powerful as static ones with a single program for λL.

A novel total harmonic distortion (THD) measuring technique is proposed. A modified Volterra series using harmonics in place of powers of the sinusoidal input is used to identify the nonlinear models of the source and the device under test (DUT). The least-mean square (LMS) adaptive algorithm is applied for identification. While maintaining comparable speed and accuracy this technique provides a more flexible test procedure than conventional methods, in terms of the frequency resolution, the number of samples, and the sampling rate. It outperforms conventional methods when there is a bin energy leakage, which occurs in a non-coherent system. In addition, it is real-time computing while other conventional methods post process blocks of data. Simulation results collaborate the analytical results.

In a hybrid coder which employs motion compensation and discrete cosine transform (MC-DCT coder), up to 90% of bits are used to represent the quantized DCT blocks. So it is most important to represent them with as few bits as possible. In this paper, we propose an efficient method for encoding the quantized DCT blocks of motion compensated prediction (MCP) errors, which adaptively selects one of a few scanning patterns. The scanning pattern selection of an MCP error block is based on the motion compensated images which are always available at the decoder as well as at the encoder. No overhead information for the scanning patterns needs to be transmitted. Simulation results show that the average bit rate reduction amounts to 5%.

A beam adaptive frame for finite-element beam-propagation analysis is proposed. The width of the frame can be adapted itself to either the guiding structure or the propagating beam in optical circuits, so the size of the computational window can be reduced.

This paper describes an adaptively weighted code division multiplexing (AW-CDM) system, in other words, power controlled spread-spectrum multiplexing system and describes its application to hierarchical digital broadcasting of television signals. The AW-CDM, being combined with multi-resolutional video encoder, can provide such a hierarchical transmission that allows both high quality services for fixed receivers and reduced quality services for mobile/portable receivers. The carrier and the clock are robustly regenerated by using a spread-spectrum multiplexed pseudorandom noise (PN) sounder as a reference in the receiver. The PN reference is also used for Rake combining with signals via different paths, and for adaptive equalization (EQ). In a prototype AW-CDM modem, three layers of hierarchical video signals (highs: 5.91Mbps, middles: 1.50Mbps, and lows: 0.46 Mbps) are divided into a pair of 64 orthogonal spread-spectrum subchannels, each of which can be given a different priority and therefore a different threshold. In this case, three different thresholds are given. The modem's transmission rate is 9.7Mbps in the 6MHz band. Indoor transmission tests confirm that lows (weighted power layer I), middles (averaged power layer II), and highs (lightened power layer III) are retrievable under conditions in which the desired to undesired signal ratios (DURs) are respectively 0dB, 8.5dB, and 13.5dB. If the undesired signals are multipaths, these performances are dramatically improved by Rake combining and EQ. The AW-CDM system can be used for 20-30 Mbps advanced television (ATV) transmission in the 6-MHz bandwidth simply by changing the binary inputs into quaternary or octonary inputs.

In the future satellite broadcasting system in 21GHz band, the rainfall attenuation is a most significant problem. To solve this problem, the hierarchical transmission systems have been studied. This paper analyzes the performance of the hierarchical modulation scheme from the view point of power assignment in the presence of the rainfall attenuation. This paper shows an optimum power assignment ratio to maximize the spectral efficiency and the signal-to-noise ratio of received image, and these optimum ratio is varied with the measure of system performance.

In this letter we present an electronic circuit based on a neural net to compute the discrete Walsh transform. We show both analytically and by simulation that the circuit is guaranteed to settle into the correct values.

Synchronization has been one of the problems in M–ary spread spectrum communication systems. In this letter, we propose the frame synchronization method using the Hadamard matrix and a frame synchronization method of PCM communication systems. Moreover, we analyze the probabilities of keeping synchronous state and frame renewal rates, and we evaluate the relationship between these probabilities and the number of stages of counters.

The sidelobe canceler in radar systems is a highly computational demanding problem. It can be efficiently tackled by resorting to the QR decomposition mapped onto a systolic array processor. The paper reports several mapping strategies by using massive parallel computers available on the market. MIMD as well as SIMD machines have been used, specifically MEIKO Computing Surface, nCUBE2, Connection Machine CM-200, and MasPar MP-1. The achieved data throughput values have been measured for a number of operational situations of practical interest.

The new algorithm for VP bandwidth control described and analyzed in this paper is a revised version of the Successive Modification Method. Its operation is based only on call-level performance (call blocking probabilities) measured in real time, without explicitly taking the cell-level performance into account. This algorithm does not need to predict future traffic demand and to perform network-wide optimization according to the predicted traffic. These features are well suited for a B-ISDN environment, with the variety of ATM bearer services and the uncertainty of their traffic demand and other characteristics. This paper describes the relationship between the proposed control and other traffic controls in ATM networks, such as CAC and VP shaping/policing. It also offers a solution to the problem of the competition that arises when several VPs in the same transmission path need increased bandwidth. Evaluation of the transient behavior of the VP bandwidth occupied by VCs shows that there is a lower limit in the control cycle and that this limit can be estimated as the longest average holding time of VCs among all services. Numerical results obtained using a call-by-call simulator show that proposed control is effective in preventing the performance degradation caused by a large traffic imbalance in communications networks. Comparison of the proposed control with a dynamical alternate routing for VC reveals that the VP bandwidth control is effective in relieving only the areas showing serious performance degradation, but that it is not so effective in improving the overall network performance.

A new gradient type adaptive algorithm is proposed in this paper. It is formulated based on the least squares criteria while the conventional gradient algorithms are based on the least mean square criteria. The proposed algorithm has two variable parameters and by changing them we can adjust the characteristic of the algorithm from the RLS to the LMS depending on the environment. This capability of adjustment achieves the possibility of providing better solutions. However, not only it provides better solutions than the conventional algorithms under some conditions but also it provides a very interesting theoretical view point. It provides a unified view point of the adaptive algorithms including the conventional ones, i.e., the LMS or the RLS, as limited cases and it enables us to analyze the bounds for those algorithms.

An alias free parallel structure for adaptive digital filters (ADF's) is considered. The method utilizes the properties of the Frequency-Sampling Filter (FSF) banks to obtain alias free points in the frequency domain. We propose a new cost function for parallel ADF's. The limiting value analysis of system identification using proposed cost function is given in stochastic sense. It is also shown by simulation examples that we can carry out precise system identification. The cost function is defined in each bin; accordingly, it enables the parallel processing of ADF's.

A rigorous radar cross section (RCS) analysis of a two-dimensional parallel-plate waveguide cavity with three-layer material loading is carried out for the E- and H-polarized planc wave incidence using the Wiener-Hopf technique. Introducing the Fourier transform for the scattered field and applying boundary conditions in the transform domain, the problem is formulated in terms of the simultaneous Wiener-Hopf equations satisfied by the unknown spectral functions. The Wiener-Hopf equations are solved via the factorization and decomposition procedure together with rigorous asymptotics, leading to the efficient approximate solution. The scattered field in the real space is evaluated by taking the inverse Fourier transform and applying the saddle point method. Representative numerical examples on the RCS are given for various physical parameters. It is shown that the three-layer lossy material loading inside the cavity results in significant RCS reduction over broad frequency range.

A new steepest descent linear adaptive algorithm, called the proportion-sign algorithm (PSA), is introduced and its performance analysis is presented when the signals are from zero-mean jointly stationary Gaussian processes. The PSA improves the convergence speed over the least mean square (LMS) algorithm without overly degrading the steady-state error performance and has the robustness to impulsive interference occurring in the desired response by adding a minimal amount of computational complexity. Computer simulations are presented that show these advantages of the PSA over the LMS algorithm and demonstrate a close match between theoretical and empirical results to verify our analysis.

For system identification problems, such as noise and echo cancellation, FIR adaptive filters are mainly used for their simple adaptation and numerical stability. When the unknown system is a high-Q resonant system, having a very long impulse response, IIR adaptive filters are more efficient for reduction in the order of a transfer function. One way to realize the IIR adaptive filter is a separate form, in which the numerator and the denominator are separately realized and adjusted. In the actual applications, the order of the unknown system is not known. In this case, it is very important to estimate the total order and the order assignment on the numerator and the denominator. In this paper, effects of the order estimation error on the residual error are investigated. In this form, indirect error evaluation called "equation error" is used. Through theoretical and numerical investigation, the following results are obtained. First, under estimation of the order of the denominator causes large degradation. Second, over estimation can improve the performance. However, this improvement is saturated to some extent due to cancellation of the redundant poles and zeros. Third, the system identification error is proportional to the equation error as the adaptive filter approaching the optimum. Finally, there is possibility of recovering from the unstable state as the order assignment approaches to the optimum in an adaptive process using the equation error. Computer solutions are provided to aid in gaining insight of the order assignment and stability problem.

This paper proposes an adaptive modulation/TDMA scheme to achieve high capacity personal multi-media communication systems. TDMA is employed to cope with various bit rate for multi-media services. The modulation scheme is selected from 1/4-rate QPSK, 1/2-rate QPSK, QPSK, 16QAM and 64QAM according to the received C/IC (power ratio of the desired signal to the co-channel interference) and the delay spread. The spectral efficiency is evaluated by using the simulated bit error rate (BER) performance as well as the cumulative distribution of the C/IC with parameters of cell configurations. The results show that the spectral efficiency of the proposed scheme is 3.5 times higher than that of the conventional QPSK systems at the outage probability of 10%, and the effect is more remarkable at lower outage probability. The results also show that the proposed adaptive modulation is effective in improving delay spread immunity.

Radar signals fluctuate because of the incoherent scattering of raindrops. Dual-polarization radar estimates rainfall rates from differential reflectivity (ZDR) and horizontal reflectivity (ZH). Here, ZDR and ZH are extracted from fluctuating radar signals by averaging. Therefore, instrumentally measured ZDR and ZH always have errors, so that estimated rainfall rates also have errors. This paper evaluates rainfall rate errors caused by signal fluctuation. Computer simulation based on a physical raindrop model is used to investigate the standard deviation of rainfall rate. The simulation considers acquisition time, and uses both simultaneous and alternate sampling of horizontal and vertical polarizations for square law and logarithmic estimators at various rainfall rates and elevation angles. When measuring rainfall rates that range from 1.0 to 10.0mm/h with the alternate sampling method, using a logarithmic estimator at a relatively large elevation angle, the estimated rainfall rates have significant errors. The simultaneous sampling method is effective in reducing these errors.

A method for evaluating the degradation of subband adaptive digital filters (ADF) is presented. The performance of a simple ADF that uses critical sampling is mainly influenced by the subband filter bank's characteristics and the finite precision arithmetic operations used. This paper considers a two-channel mirror filter bank and a normalized least mean square algorithm with floating point arithmetic. The theoretical ERLE (Echo Return Loss Enhancement) and the theoretical relationships between the output error of the ADF and the circuit parameters considering finite precision A/D conversion and finite word length effects in floating point arithmetic operation are obtained using an equivalent noise model. Simulation results are found to be in good agreement to analytical values; the difference is only 3 to 5 dB.