This paper examines the effect of segmentation mismatch on audio-video transmission by Bluetooth. We focus on the segmentation mismatch caused by the difference between the RFCOMM Maximum Frame Size and the baseband packet payload size. By experiment, we assessed the maximum throughput and media synchronization quality for various types of ACL packets. In the experiment, a media server transferred stored video and audio streams to a single terminal with point-to-point communication; we supposed no fading environment and added white noise by which interference from DSSS systems is modeled. The experiment showed that the effect of segmentation mismatch is large especially when the total bit rate of the two streams is near the channel transmission rate. We also observed that the media synchronization control is effective in compensating for the disturbance by the segmentation mismatch in noisy environments.

The characteristics of the spreading sequence significantly affect the signal-to-interference power ratio (SIR) of the received signal in direct sequence code division multiple access (DS-CDMA) system. In this paper, we analyze the receiver performance of the forward link of a DS-CDMA system in terms of the SIR and bit error rate (BER) when pseudo noise (PN) codes and concatenated orthogonal/PN (OPN) codes are used as the spreading sequence. The use of OPN spreading codes can cancel out the intra-cell interference signals with equal path delay, but the use of PN spreading codes cannot, significantly degrading the performance. As a result, the BER performance of the OPN spreading system is better than that of the PN spreading system. The use of OPN spreading sequences can provide the system capacity at least two times larger than the use of PN spreading sequences in the single-cell environment even when the channel has a large number of multipaths. The two spreading systems also show significant difference in the user capacity even in a multi-cell environment.

In order to produce precise enclosures from a multi-dimensional interval vector, we introduce a sharp interval sub-dividing condition for optimization algorithms. By utilizing interval inclusion properties, we also enhance the sampling of an upper bound for effective use in the interval quadratic method. This has resulted in an improvement in the algorithm for the unconstrained optimization problem by Hansen in 1992.

This paper presents a new automatic-frequency control (AFC) configuration capable of removing wide range frequency offsets (up to about 0.625 fs, where fs is signal symbol rate). The new configuration consists of an AFC that removes frequency offsets between 0.125 fs and another AFC that detects the frequency offset range coarsely between 0.625 fs. This paper describes the principle of the new AFC configuration. The proposed AFC configuration employs four correlators to enhance the acquisition range. It also adopts the reverse modulation scheme to decrease the acquisition time. The performance of the new AFC configuration is confirmed via computer simulations. It is shown that the proposed configuration can accommodate wide range frequency offsets as well as reduce the acquisition time.

A prototype modem with unequal error protection (UEP) capability was developed using multiple block coded modulation (MBCM). Benefited from its unique structure, MBCM can be used to provide UEP straightforward. We propose a new method to increase the robustness of the carrier recovery process by taking advantage of the MBCM code structure. We also use a frame format to facilitate the synchronization operation. This modem was developed in preparation for the mobile satellite communication experiments using the Engineering Test Satellite of VIII-type (ETS-VIII). In addition of MBCM, some other types of modulation schemes have been implemented in the same modem to enable a range of communication experiments to be performed. The modem can operate at variable data rates. The results of laboratory measurements agreed well with computer simulation results. Typical link budgets based on the parameters of ETS-VIII are also presented.

This paper addresses the utility-based radio resource allocation problem in DS-CDMA systems carrying multimedia traffic. The proposed scheme, aiming at achieving optimal resource allocation, considers the joint power and data rate allocation. To avoid high computational complexity of nonlinear optimization, we reformulate the radio resource allocation problem as a market model, where resource is regarded as a commodity. Since the market model satisfies the incentive-compatible constraint, the optimal resource allocation can be obtained at the market equilibrium in a distributed manner. According to whether to allocate a minimal transmission data rate to each user, two algorithms, UCA and FCA, are proposed. UCA emphasize on maximizing system overall utilities, while FCA guarantees fairness to users. Simulation results show that the proposed radio resource allocation scheme and algorithms are flexible and efficient for multimedia DS-CDMA systems.

Smart or adaptive antennas promise to provide significant space-time communications against fading in wireless communication systems. In this paper, we propose multiple-input multiple-output (MIMO) beamforming for frequency-selective fading channels to maximize the Signal-to-Noise and Interference Ratio (SINR) based on an iterative update algorithm of transmit and receive weight vectors with prior knowledge of the channel state information (CSI) at both the transmitter and receiver. We derive the necessary conditions for an optimum weight vector solution and propose an iterative weight update algorithm for an optimal SINR reception. The Maximum Signal-to-Noise (MSN) method, where noise includes the additive gaussian noise and interference signals, is used as a criterion. The proposed MIMO with M N arrays allows the cancellation of M + N - 2 delayed channels. Computer simulations are presented to verify our analysis. The results show that significant improvements in performance are possible in wireless communication systems.

Call Admission Control (CAC) is a very important issue in CDMA systems to guarantee a required quality of service (QoS) and to increase system capacity. In this paper, we proposed and analyzed the CAC scheme using multiple criterions (MCAC), which can provide a quicker processing time and better performance. One is based on the number of active users with the minimum/maximum threshold by considering the spillover ratio, and the other is based on the signal to interference ratio (SIR). If active users are lower/higher than the minimum/maximum number of users threshold (N_min )/(N_max ), we accept/reject the new call without any other considerations based on the first criterion. And if the number of active users is between the N_min and N_max, we consider the current SIR to guarantee QoS based on the second criterion. Then the system accepts the new call when the SIR satisfies the system requirements, otherwise, the call is rejected. The multiple criterions scheme is investigated and its performance is compared with the number of user based CAC and power based CAC.

In this paper, the design of a QoS scheduling scheme for the Internet traffic is proposed by considering the correlation property of the arriving traffic. The basic concept of the Weighted Fair Queuing (WFQ) is adopted in the proposed scheme, however, the correlation property of the traffic stream is applied as the heuristic to adjust the share weight factors of each traffic type dynamically. The Auto Regressive Integrated Moving Average (ARIMA) model is applied in this paper to characterize the correlation property of the Internet traffic. And the share weight factors are derived from the parameters of the AR part and MA part. Experimental simulations are performed to illustrate the effectiveness of the proposed scheme. In addition to comparing the performance of each service types, we also define a fair play parameter (FPP) to examine the fairness index among various traffic streams of the proposed scheme. The experimental results indicate that the proposed scheme demonstrates a quite good performance in scheduling the integrated services and the fairness among service classes can also be achieved, especially when the link capacity is limited.

A round-ended wide straight slot cut in the broad wall of a rectangular waveguide is analyzed by the Method of Moments (MoM) using numerical eigenmode basis functions derived by the edge-based finite element method (FEM), referred to as MoM/FEM. The frequency characteristics of the calculated transmission coefficients are compared with the measured ones, and good agreement is observed for a wide variety of antenna parameters. For simpler analysis that does not use MoM/FEM, an equivalent rectangular slot approximation for a round-ended slot is discussed. The resonant frequencies of empirically introduced "equal-area" and "equal-perimeter" slots are compared with those of round-ended slots for a wide variety of parameters such as slot width, wall thickness and dielectric constant inside the waveguide. Based upon MoM/FEM, which can be a reliable reference, it is found that the equal-area slot always gives a better approximation of the order of 1% over that of the equal-perimeter one which is of the order of 5%. For higher accuracy, a new rectangular slot approximation of a round-ended slot is proposed to be a linear combination of equal-area and equal perimeter approximation. The error is around 0.25% for a wide variety of parameters such as slot width-to-length ratio, wall thickness and dielectric constant of the filling material inside the waveguide.

This letter describes the design and performance of a new ATM Adaptation Layer (AAL-UDP) for time-critical traffic over wireless ATM networks. The key ideas in the design consist of no discard at the AAL level and header protection with sequence number mechanism. The UDP/IP header is repeated for reliability, because it contains the most important information such as address and port number. The simulation results show that the AAL-UDP provides significant improvement in throughput as well as in application-level performance compared to the conventional AAL 5 case.

In a high-rate indoor wireless personal communication system, the delay spread due to multi-path propagation results in intersymbol interference which can significantly increase the transmission bit error rate (BER). The technique most commonly used for combating the intersymbol interference and frequency-selective fading found in communications channels is the adaptive equalization. In this paper, we propose a novel neural detector based on self-organizing map (SOM) to improve the system performance of the receiver. In the proposed scheme, the SOM is used as an adaptive detector of equalizer, which updates the decision levels to follow the received faded signal. To adapt the proposed scheme to the time-varying channel, we use the Euclidean distance, which will be updated automatically according to the received faded signal, as an adaptive radius to define the neighborhood of the winning neuron of the SOM algorithm. Simulations on a 16 QAM system show that the receiver using the proposed neural detector has a significantly better BER performance than the traditional receiver.

Computer simulation of cellular process is one of the most important applications in bioinformatics. Since such simulators need huge computational resources, many biologists must use expensive PC/WS clusters. ReCSiP is an FPGA-based, reconfigurable accelerator which aims to realize economical high-performance simulation environment on desktop computers. It can exploit fine-grain parallelism in the target applications by small hardware modules in the FPGA which work in parallel manner. As the first step to implement a simulator of cellular process on ReCSiP, a solver to perform a basic simulation of metabolism was implemented. The throughput of the solver was about 29 times faster than the software on Intel's PentiumIII operating at 1.13 GHz.

A Generalized Symbol-rate-increased (GSRI) Pragmatic Trellis coded Type-I Hybrid ARQ based on a Selective-Repeat (SR) ARQ with multicopy (MC) retransmission (SR+MC scheme) for high speed mobile satellite communication system is analyzed. The SR+MC ARQ is a suitable scheme for mobile satellite systems and further improvement of the throughput performance can be expected by an additional combination of an error control coding. In this paper, we investigate the performance of the SR+MC scheme employing GSRI Pragmatic TCM. GSRI TC-MPSK can arbitrarily set the bandwidth expansion ratio keeping higher coding gain than conventional TCM scheme. Also Pragmatic TCM has an advantage in that the modulation level can be easily changeable. By changing the modulation level and the bandwidth expansion ratio, this scheme can optimize the performance according to the channel conditions. Numerical and simulation results show that the GSRI Trellis Coded Type-I Hybrid ARQ presents better performance than conventional Pragmatic Trellis Coded Type-I Hybrid ARQ.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

Multi-Input Multi-Output (MIMO) systems, which utilize multiple antennas at both the receiver and transmitter, promise very high data rates in a rich scattering environment. It was proven in literature that with optimal power allocation, MIMO eigenmode transmission system (EMTS) is optimal because MIMO capacity is maximized. However, the performance of MIMO EMTS is very sensitive to the accuracy of channel state information and thus it is of practical importance to analyze its performance when channel state information is corrupted under realistic system and propagation conditions. In this paper, we lower bound the mutual information of MIMO EMTS with imperfect channel estimation and delayed quantized feedback in a spatially correlated continuous fading channel. Our results showed that this lower bound is tight and can serve as a comprehensive guide to the actual performance of MIMO EMTS under practical operating conditions.

This paper describes the design of a programmable QAM transceiver for VDSL applications. A 12-b DAC with 64-dB spurious-free dynamic range (SFDR) at 75-MS/s and an 11-b ADC with 72.3-dB SFDR at 70-MS/s are integrated in this complete physical layer IC. A digital IIR notch filter is included in order to not interrupt existing amateur radio bands. The proposed dual loop AGC adjusts the gain of a variable gain amplifier (VGA) to obtain maximum SNR while avoiding saturation. Using several low power techniques, the total power consumption is reduced to 300-mW at 1.8-V core and 3.3-V I/O supplies. The transceiver is fabricated in a 0.18-µm CMOS process and the chip size is 5-mm 5-mm. This VDSL transceiver supports 13-Mbps data rate over a 9000-ft channel with a BER < 10-7.

Independent shadowing losses are often assumed for computing the frequency reuse distance of cellular mobile communication systems. However, shadowing losses may be partially correlated since the obstacles surrounding a mobile station block similarly the desired signal and interfering signals. We investigate, by computer simulation, how the shadowing correlation impacts the frequency reuse distance of a power controlled cellular system. It is pointed out that the shadowing correlation impacts the frequency reuse distance differently for the uplink and downlink.

A new scheme for watermarking based on vector quantization (VQ) over a binary symmetric channel is proposed. By optimizing VQ indices with genetic algorithm, simulation results not only demonstrate effective transmission of watermarked image, but also reveal the robustness of the extracted watermark.

We have developed Multi-modal Neural Networks (MNN) to improve the accuracy of symbolic sequence pattern classification. The basic structure of the MNN is composed of several sub-classifiers using neural networks and a decision unit. Two types of the MNN are proposed: a primary MNN and a twofold MNN. In the primary MNN, the sub-classifier is composed of a conventional three-layer neural network. The decision unit uses the majority decision to produce the final decisions from the outputs of the sub-classifiers. In the twofold MNN, the sub-classifier is composed of the primary MNN for partial classification. The decision unit uses a three-layer neural network to produce the final decisions. In the latter type of the MNN, since the structure of the primary MNN is folded into the sub-classifier, the basic structure of the MNN is used twice, which is the reason why we call the method twofold MNN. The MNN is validated with two benchmark tests: EPR (English Pronunciation Reasoning) and prediction of protein secondary structure. The reasoning accuracy of EPR is improved from 85.4% by using a three-layer neural network to 87.7% by using the primary MNN. In the prediction of protein secondary structure, the average accuracy is improved from 69.1% of a three-layer neural network to 74.6% by the primary MNN and 75.6% by the twofold MNN. The prediction test is based on a database of 126 non-homologous protein sequences.