Bursts from a number of variable bit rate sources allocated to a virtual path with a given capacity can inundate the channel. Buffers used to take care of such bursts can fill up rapidly. The buffer size limits its burst handling capability. With large bursts or a number of consecutive bursts, the buffers fill up and this leads to high cell losses. Channel reconfiguration with dynamic allocation of spare capacities is one of the methods used to alleviate such cell losses. In reconfigurable networks, spare capacity allocation can increase the channel rates for short durations, to cope with the excess loads from the bursts. The dynamic capacity allocations are adaptable to the loads and have fast response times. We propose heuristic rules for spare capacity assignments in ATM networks. By monitoring buffer occupancy, triggers which anticipate excess traffic can be used to assign spare capacities to reduce the cell loss probabilities in the network.

In the future, more and more network providers will be established through the introduction of an open telecommunications market. At this time, it is necessary to guarantee the fair competition between these network providers. In this paper, a negotiation protocol for connection establishment is proposed. This negotiation protocol is based on the concept of open, competitive bidding and can guarantee fair competition between the network providers. In this negotiation protocol, each network providers objective is to maximize its profit. Conversely, each users objective is to select a network provider which will supply as much capacity as required. Employing this negotiation protocol, the users and the network providers can select each other based on their objectives. In this paper, adaptation strategies which network providers and users can adopt under the proposed negotiation protocol framework are also discussed. A network provider which adopts this strategy can obtain enough profit even when the number of connection requests is small relative to the idle bandwidth capacity. Moreover, a user who adopts this strategy can be sure to obtain bandwidth even when the idle bandwidth capacity is small relative to the number of connection requests.

In this paper, we propose a new structure of the frequency domain adaptive filter (FDAF). The proposed structure is based on the modified DFT pair which consists of the FIR filters, so that un-delayed output signal can be obtained with stable convergence and without accumulated error which are problems for the conventional FDAFs. The convergence performance of the proposed FDAF is examined through the computer simulations in the adaptive line enhancer (ALE) comparing with the conventional FDAF and the DCT domain adaptive filter. Furthermore, in order to improve the error performance of the FDAF, we propose a composite algorithm which consists of the normalized step size algorithm for fast convergence and the variable step size one for small estimation error. The advantage of the proposed algorithm is also confirmed through simulations in the ALE. Finally, we propose a reduction method of the computational complexity of the proposed FDAF. The proposed method is to utilize a part of the FFT flow-graph, so that the computational complexity is reduced to O(N log N).

A maximal-ratio-combining (MRC) digital beamformer has been studied to attain open-loop and automatic self-beam steering towards both desired and multipath signals at the same time and diversity combining of the signals, which are made possible by spatial digital signal processing. This paper describes the performance of this beamformer under the multipath signal arrivals with various path delays using numerical simulation, aimed at application to future mobile radios with high spatial utilization efficiency. The results indicate the robustness of the MRC beamformer in a multipath environment. It features multidirectional beam steering when there is small path delay in the multipath signal and gain suppression in multipath signals when their path delay is more than about one symbol. Moreover, improvement in suppression by employing low-sidelobe amplitude distribution is discussed as a means to reduce inter-symbol interference without null-beam steering.

We have applied hot-carrier circuit-level simulation to memory peripheral circuits of a few thousand to over 12K transistors using a simple but accurate degradation model for reliability verification of actual memory products. By applying simulation to entire circuits, it was found that the location of maximum degradation depended greatly upon circuit configuration and device technology. A design curve has been developed to quickly relate device-level DC lifetime to circuit-level performance lifetime. Using these results in conjunction with a methodology that has been developed to predict hot-carrier degradation early in the design cycle before TEG fabrication, accurate total-circuit simulation is applied early in the design process, making reliability simulation a crucial design tool rather than a verification tool as technology advances into the deep sub-micron high clock rate regime.

A system combining multicarrier modulation and adaptive modulation in which a suitable level of quadrature amplitude modulation (QAM) is selected for each subcarrier and time-slot, is proposed for high-bit-rate and high-quality digital land mobile communications. The advantages of the system are a mode in which information cannot be transmitted under adverse propagation conditions and a buffer memory to limit a transmission delay time. If the allowable delay time is small, such as in voice and video transmissions, the system tends to have a poor bit error rate (BER) because of the forcible QAM-level selection. Our new selection scheme improves the BER for small transmission delay time. Suitable distribution of the delay time among subcarriers is obtained by using the scheme where the QAM-level of each subcarrier is chosen collectively using the number of data bits stored in memory. Computer simulation of the systems BER performance showed that the system could provide a noticeable BER improvement over frequency-selective fading channels as well as flat Rayleigh fading channels. The QAM-level selection scheme was also effective for a low maximum Doppler frequency and a small memory size. The system could thus attain about 25-fold improvement in BER at Es/N030 dB compared to the multicarrier/16QAM system. It also attained about 60-fold and 3. 5-fold improvement in BER at fd=10Hz compared with the system with multicarrier/16QAM and without the QAM-level selection scheme, respectively.

An adaptive array has been proposed as a canceller for both inter-symbol interference (ISI) and co-channel interference (CCI). However, it has no path-diversity gain since it selects just one signal correlated to the reference signal. In this paper, a novel interference canceller having sufficient path-diversity gain is proposed. The canceller is characterized by the combined configuration of an adaptive array and an equalizer. In the proposed system, a pre-selecting adaptive array is installed first. By employing a specific training sequence and sampling timing at the receiver during the training period, the perfect correlation between the "desired signal" and "short delayed" is achieved. Therefore, the pre-selecting adaptive array can extract the desired and ISI signals simultaneously, and the cascaded adaptive equalizer can provide the path-diversity gain without degradation by interference. The proposed system achieves a simple configuration and robustness against both ISI and CCI with a sufficient path diversity gain. In computer simulations, average BER characteristics of the proposed system were evaluated in a quasi-static Rayleigh fading channel. The simulation results showed that the system can reduce both long-delayed ISI and CCI efficiently, and that the expected path diversity gain is obtained even with strong CCI. They also showed that the degradation is not so serious when the number of antenna elements is less than that of incoming signals.

In the multipath mobile channel, the received signal suffers from both the fluctuation in the received field intensity caused by fading and waveform distortion caused by the echo. Diversity reception using multiple spaced antennas is an effective method to compensate for fading, while echo cancellation with an adaptive array is good at compensating for waveform distortion. In this paper, an adaptive switching echo cancellation/diversity reception method to compensate for both waveform distortion and fading is proposed. The proposed switching reception monitors the impacts of channel conditions on received signal and then one of an echo canceller and a diversity receiver is selected accordingly to compensate the channel. The compensation performance of the proposed switching reception in terms of both average DUR (Desired to Undesired signal Ratio) and the probability of DUR below a threshold value is investigated with computer simulation. The results show that the adaptive switching echo cancellation/diversity reception has realized the advantages of both adaptive echo cancellation and diversity reception.

In this paper, we compare the performance of evolutionary digital filters (EDFs) for IIR adaptive digital filters (ADFs) in terms of convergence behavior and stability, and discuss their advantages. The authors have already proposed the EDF which is controlled by adaptive algorithm based on the evolutionary strategies of living things. This adaptive algorithm of the EDF controls and changes the coefficients of inner digital filters using the cloning method or the mating method. Thus, the adaptive algorithm of the EDF is of a non-gradient and multi-point search type. Numerical examples are given to demonstrate the effectiveness and features of the EDF such that (1) they can work as adaptive filters as expected, (2) they can adopt various error functions such as the mean square error, the absolute sum error, and the maximum error functions, and (3) the EDF using IIR filters (IIR-EDF) has a higher convergence rate and smaller adaptation noise than the LMS adaptive digital filter (LMS-ADF) and the adaptive digital filter based on the simple genetic algorithm (SGA-ADF) on a multiple-peak surface.

Penumbral imaging is a technique which uses the facts that spatial information can be recovered from the shadow or penumbra that an unknown source casts through a simple large circular aperture. The technique is based on a linear deconvolution. In this paper, a two-step method is proposed for decoding penumbral images. First a local-statistic filter based on adaptive windowing is applied to smooth the noise; then, followed by the conventional linear deconvolution. The simulation results show that the reconstructed image is dramatically improved in comparison to that without the noise-smoothing filtering, and the proposed method is also applied to real experimental X-ray imaging.

An adaptive infinite impulse response (IIR) filter implemented using an allpass and a minimum phase system has an advantage of its poles converging to the poles of the unknown system when the input is a white signal. However, when the input signal is colored, convergence speed deteriorates considerably, even to the point of lack of convergence for certain colored signals. Furthermore with a colored input signal, there is no guarantee that the poles of the adaptive digital filter (ADF) will converge to the poles of the unknown system. In this paper we propose a method which uses a linear predictor filter to whiten the input signal so as to improve the convergence characteristic. Computer simulation results confirm the increase in convergence speed and the convergence of the poles of the ADF to the poles of the unknown system even when the input is a colored signal.

An adaptive stability-growth (ASG) learning algorithm is proposed for improving, as much as possible, the stability of a Hopfield-type associative memory. While the ASG algorithm can be used to determine the optimal stability instead of the well-known minimum-overlap (MO) learning algorithm with sufficiently large lower bound for MO value, it converges much more quickly than the MO algorithm in real implementation. Therefore, the proposed ASG algorithm is more suitable than the MO algorithm for real-world design of an optimal Hopfield-type associative memory.

This paper presents two types of two-dimensional (2-D) adaptive beamforming algorithm which have high rate of convergence. One is a linearly constrained minimum variance (LCMV) beamforming algorithm which minimizes the average output power of a beamformer, and the other is a generalized sidelobe canceler (GSC) algorithm which generalizes the notion of a linear constraint by using the multiple linear constraints. In both algorithms, we apply a 2-D lattice filter to an adaptive filtering since the 2-D lattice filter provides excellent properties compared to a transversal filter. In order to evaluate the validity of the algorithm, we perform computer simulations. The experimental results show that the algorithm can reject interference signals while maintaining the direction of desired signal, and can improve convergent performance.

A slotted ALOHA direct sequence spread spectrum system with random signatures is considered. The system is applicable in cases where a large number of terminals transmit to a single hub station like in cellular digital radio, personal mobile systems and wireless LANs. It is shown that significant improvements in packet throughput capacity are obtained if the adaptive receiver structures are used. Systems for the comparison are the spread spectrum slotted ALOHA system and the conventional slotted ALOHA system.

Pilot symbol-assisted (PSA) decision-directed coherent adaptive array diversity (CAAD) is proposed for increasing the reverse link capacity of DS-CDMA mobile radio systems. In the proposed scheme, PSA channel estimation is applied to coherent Rake combining, and the weights of the antenna array are adaptively updated using both pilot symbols and decision-directed data symbols after Rake combining as references for minimum mean squared error (MMSE) criteria. The reverse link capacity of a 3-sectored base station is evaluated by computer simulation when fast transmit power control (TPC) based on singal-to-interference plus backgound noise power ratio (SIR) measurement is applied under nultipath Rayleigh fading environments. It is shown that a 6-element (sector) CAAD receiver can increase the capacity to about 4.2 times that with a single antenna (per sector) receiver when links are interference-limited. The link capacity achievable with the 6-element CAAD receiver is 1.2 times that with a 6-branch antenna diversity reciever with antenna spacing of 10 carrier wavelengths, while significantly reducing the strong interference from high bit rate transmission (high transmit power) users.

It is well known that M-ary/spread spectrum (M-ary/SS) system is superior to direct-sequence spread spectrum system under AWGN, and can achieve high spectral efficiency. On the other hand, however, the main drawback of this system is that the synchronization acquisition is difficult. In this paper, we propose a new synchronization acquisition method of M-ary/SS system. This method acquires the code synchronization by introducing a symmetrical property in spreading sequences, and detecting this property with the differential decoding technique. As spreading sequences, a set of orthogonal sequences and a set of non-orthogonal sequences are considered. The strong features of proposed systems are that the systems can acquire the code synchronization in carrier band and can reduce the complexity of calculation greatly. Among the comparison results of the systems with newly proposed orthogonal and some specific non-orthogonal spreading sequences, it is especially noted that the latter can reduce the mean acquisition time and calculation complexity much greater than the former.

This paper propeses advanced multi-stage interference canceller systems (MSICS) wihch can estimate radio channels with precision in the direct sequence code division multiple access (DS-CDMA) systems. For the accurate channel estimations, we propose a novel radio channel estimation method specified by the following two signal processing methods. One is the radio channel estimation using both pilot and information signals. The other is the correction of estimated radio channels using adaptation algorithm based on the least mean square method (LMS). The results of our computer simulation indicate that the cell capacity of the advanced MSICS in serial and parallel structure can be increased by about 1.8 and 1.3 times over that of a receiver which does not has a canceller, respectively. Moreover, the advanced MSICS in serial and parallel structure can reduce the required Eb/No by about 1.2 dB and 1.6 dB at a BER of 10-3 compared to the Eb/No of a basic MSICS, respectively.

This paper proposes a novel synchronous acquisition method with an adaptive filter in asynchronous direct sequence/code division multiple access (DS/CDMA) communication systems. An adaptive filter is used in a single-user receiver, in complete synchronization of desired user's signal, the tap coefficients of the filter are controlled to orthogonalize to all other user's spreading sequences without knowledge of the sequences, amplitude and time delays of the signals. While, in the proposed system for synchronous acquisition, the tap coefficients are controlled to orthogonalize to all user's sequences including desired user's signal. The synchronous acquisition can be achieved by using the difference of cross-correlation function value between desired user's sequence of inphase and the tap coefficients for each phase. The principle and performance evaluation for the proposed method are shown. As a result, compared to an acquisition method of conventional sliding correlator, considerable improvement of the average acquisition time can be achieved in large power multiple access interference environment.

In this paper, we present orthogonal multi-code CDMA systems with the constant amplitude transmission. In general, the dynamic range of the amplitude of the transmitting signal is very large in the case of orthogonal multi-code CDMA systems. In order to realize the constant amplitude transmission for orthogonal multi-code CDMA systems, we propose a constant amplitude coding. First, we show the basic concept of the constant amplitude coding. And then, we show that the constant amplitude transmission can be realized by the combination of the conventional orthogonal multi-code CDMA and the constant amplitude coding. Finally, the effectiveness of the proposed method is evaluated in terms of the bit error rate performance and it is shown that the proposed method is robust to the non-linear distortion caused by a high power amplifier (HPA).

In addition to additive thermal noise, a received direct-sequence spread spectrum (DS/SS) signal may suffer from intersymbol interference (ISI) and interference caused by cochannel narrowband users or other narrowband radio frequency interference (RFI). This paper presents a workable solution for removing narrowband interference (NBI) and reducing ISI or inter-chip interference (ICI) when the communication channel can be modeled as an FIR filter and the NBI comes from multiple CW tones, an AR-modeled Gaussian process, or a BPSK signal. Unlike earlier solutions, the proposed scheme is capable of performing the functions of NBI-rejection, ISI/ICI suppression and data detection (code despreading) simultaneously. It is easy to implement and, more importantly, it yields lower bit error rate (BER) and smaller mean squared error (MSE).