In this paper, we propose a new multicast routing algorithm for constructing the delay-constrained minimal spanning tree in the VP-based ATM networks, in which we consider the efficiency even in the case where the destination dynamically joins/departs the multicast connection. For constructing the delay-constrained spanning tree, we first generate a reduced network consisting of only VCX nodes from a given ATM network, originally consisting of VPX/VCX nodes. Then, we obtain the delay-constrained spanning tree with a minimal tree cost on the reduced network by using our proposed heuristic algorithm. Through numerical examples, we show that our dynamic multicast routing algorithm can provide an efficient usage of network resources when the membership nodes frequently changes during the lifetime of a multicast connection, while the existing multicast routing algorithm may be useful for constructing the multicast tree with a static nature of destination nodes. We also demonstrate that more cost-saving can be expected in dense networks when applying our proposed algorithm.

Recently, progress has been made in the area of electrical modeling of conductors embedded in arbitrary dielectrics using circuit oriented techniques. These models usually occur in conjunction with VLSI type circuits. Many different applications exist today for such models in the EMI, EIP (Electrical Interconnect and Package) analysis as well as for the microwave circuit area. Practical problems involve a multitude of hardware components and they demand a wide spectrum of both time as well as frequency domain solution techniques. In this paper we consider circuit oriented techniques for the solution of these problems. Specifically, we give an outline of the three dimensional Partial Element Equivalent Circuit (PEEC) full wave modeling approach and review the recent progress in this area.

We consider the upper bounds of the finite block length capacity Cn, FB (P) of the discrete time Gaussian channel with feedback. We prove the relation C2 (P) C2, FB (P) < C2 (2P), which is a partial solution of the conjecture given by Cover. In addition we prove several relations.

This paper presents a new upper bound on overall bit error rate (BER) for a concatenated code which consists of an inner convolutional code and an outer interleaved Reed-Solomon code. The upper bound on BER is derived based on a lower bound on the effective minimum distance of the concatenated code. This upper bound can be used for the cases when the interleaver size is small such that the conventional upper bound is not applicable.

This paper proposes a new speech codec based on CELP for PHS multimedia communication. PHS portable terminals should consume as little power as possible, and the codec used in them has to be robust against channel errors. Therefore, the proposed codec operates with low computational complexity while reducing the deterioration in speech quality due to channel errors. This codec uses two new schemes to reduce computational complexity. One is moving average scalar quantization for the filter coefficients of the synthesis filter. This scheme requires 90% less complexity to quantize synthesis filter coefficients compared to the widely used vector quantization. The other is pre-selection for selecting an algebraic codebook used as random excitation source. An orthogonalization scheme is used for stable pre-selection. Deterioration of speech quality is suppressed by using CRC and parameter estimation for error protection. Two types of codec are proposed: a 10-ms frame type that transmits 160 bits every 10-ms and a 15-ms frame type that transmits 160 bits every 15 ms. The computational complexity of these codecs is less than 5 MOPS. In a nochannel error environment, the speech quality is equal to that of ITU-TG.726 at 32.0 kbit/s. With 0.3% channel error, both codecs offer more comfortable conversation than G.726. Moreover, at 1.0% channel error, the 10-ms frame type still provides comfortable conversation.

A method for recovering the LPC spectrum from a microphone array input signal corrupted by less directional ambient noise is proposed. This method is based on the subspace method, in which directional signal and non-directional noise is classified in the subspace domain using eigenvalue analysis of the spatial correlation matrix. In this paper, the coherent subspace (CSS) method, a broadband extension of the subspace method, is employed. The advantage of this method is that is requires a much smaller number of averages in the time domain for estimating subspace, suitable feature for frame processing such as speech recognition. To enhance the performance of noise reduction, elimination of noise-dominant subspace using projection is further proposed, which is effective when the SNR is low and classification of noise and signals using eigenvalue analysis is difficult.

Generalized minimum-distance (GMD) decoding is well-known as a soft decision decoding technique for such linear block codes as BCH and RS codes. The GMD decoding algorithm generates a set of candidate codewords and selects as a decoded codeword that candidate with the smallest reliable distance. In this paper, for a GMD decoder of RS and BCH codes, we present a new sufficient condition for the decoded codeword to be optimal, and we show that this sufficient condition is less stringent than the one presented by Taipale and Pursely.

We propose a method to search all the zeros of a complex function in a given compact region D Cn. The function f: Cn Cn to be considered is assumed to consist of polynomial and transcendental terms and to satisfy f (x) Rn for any x Rn. Using the properties of such a complex function, we can compute the number of zeros and determine the starting points of paths on the boundary of D, which attain all the zeros of f in D without encountering a singular point. A piecewiselinear approximation of the function on a triangulation is used for both computing the number of zeros and following the paths.

Bit-interleaving can enhance performance of a trellis coded modulation system over a fading channel. A combined system with decision feedback equalization is proposed. In the system, TCM decoded symbols are fed back for equalization. To avoid a bad effect of decoding delay, a deinterleaver is utilized effectively. Information sequence is divided into three subsequences and encoded by three encoders. Among the 3 code vectors from the encoders, bits are interleaved and decoding proceeds in parallel. Simulation results show that the proposed system realizes 0.6 dB more coding gain than a symbol interleaved system. A calculation method of a branch metric for decoding is proposed. Performance with the branch metric is shown to be nearly independent from the desired/undesired power ratio of a intersymbol interference channel. An approximate upper bound is analyzed for the proposed system, and the optimum code is searched.

This paper describes a vision system with dual viewing angles, i. e., wide and narrow viewing angles, and a scheme of user-friendly speech dialogue environment based on the vision system. The wide viewing angle provides a wide viewing field for wide range motion tracking, and the narrow viewing angle is capable of following a target in wide viewing field to take the image of the target with sufficient resolution. For a fast and robust motion tracking, modified motion energy (MME) and existence energy (EE) are defined to detect the motion of the target and extract the motion region at the same time. Instead of using a physical device such as a foot switch commonly used in speech dialogue systems, the begin/end of an utterance is detected from the movement of user's mouth in our system. Without recognizing the movement of lips directly, the shape variation of the region between lips is tracked for more stable recognition of the span of a dialogue. The tracking speed is about 10 frames/sec when no recognition is performed and about 5 frames/sec when both tracking and recognition are performed without using any special hardware.

In this paper, we consider conventional signaling link fault tolerance and error correction mechanisms to provide reliable services of mobile multimedia telecommunication network based on ATM (Asynchronous Transfer Mode) technology. Also, we propose an efficient signaling protocol interworking architecture and a reliable distributed interworking network architecture between SS7 based FPLMTS and ATM networks. Besides, we evaluate the performance of proposed method through computer simulation. According to the results, proposed signaling architecture shows efficient and fast fault restoration characteristics than conventional MTP-3/3b based network. Functional signaling protocol stack and network architecture of proposed fast rerouting mechanism provide reliable and efficient restoration performance in view of interworking between SS7 based FPLMTS and ATM networks.

In computer hardware there is a constant evolution towards smaller transistor sizes. At the same time, more and more transistors are placed on one chip. Both trends make the pin limitation problem worse. Scaling down chip sizes adds to the shortage of available pins while increasing the number of transistors per chip imposes a higher need for chip terminals. The use of three-dimensional systems would alleviate this pin limitation problem. In order to decide whether the benefits of such systems balance the higher processing costs, one must be able to characterize these benefits accurately. This can be done by estimating important layout properties of electronic designs, such as space requirements and interconnection length values. For a two-dimensional placement, Donath found an upper bound for the average interconnection length that follows the trends of experimentally obtained average lengths. Yet, this upper bound deviates from the experimentally obtained value by a factor of approximately 2 which is not sufficiently accurate for some applications. In this paper, we first extend Donath's technique to a three-dimensional placement. We then compute a significantly more accurate estimate by taking into account the inherent features of the optimal placement process.

In order to efficiently conduct the complex operations of network OpSs (Operation Systems), we introduce the TMN-based Integrated Management Display System (IMD). IMD is able to control the display of windows monitoring individual operations on multiple screens such as project monitors. In this paper, we describe an implementation of the IMD and experimental results. This implementation consists of the Multi-Screen Control System (MSS), Multi-Screen Agent System (MSA) and Multi-Screen Manager System (MSM). MSA controls MSS directly via proprietary protocols. MSM manages MSA via the TMN Interface. MSA is modeled using the GDMO (Guidelines for the Definition of Managed Objects). Moreover, in order to realize cooperation with other OpSs, MSM is based on TMN. We confirm that MSM interworks with ATMOS (ATM Transport Network Operation System) and develop an alarm surveillance system for ATMOS using IMD at in an OpS-Room. Experiments verify that MSA can be controlled by MSM following the alarms emitted from the real network equipment of the ATM Transport Network. As a result, the operators can comprehend the operations information more quickly than is possible with the conventional system. By using the TMN-based Integrated Management Display System (IMD), operations such as alarm surveillance can be performed more efficiently.

Distributed multimedia applications are often sensitive to the Quality of Service (QoS) provided by the communication network. They usually require guaranteed QoS service, so that real-time communication is possible. However, searching a route with multiple QoS constraints is known to be a NP-complete problem. In this paper, we propose a new simple and efficient distributed QoS routing algorithm, called "DQoSR," for supporting real-time communication in high-speed networks. It searches a route that could guarantee bandwidth, delay, and delay jitter requirements. Routing decision is based only on the modified cost, hop and delay vectors stored in the routing table at each node and its directly connected neighbors. Moreover, DQoSR is proved to construct loop-free routes. Its worst case message complexity is O(|V|2), where |V| is the number of nodes in the network. Thus DQoSR is fast and scales well to large networks. Finally, extensive simulations show that average rate of establishing successful connection of DQoSR is very near to optimum (the difference is less than 0.4%).

This study shows the results of evaluating the flux noises at low frequency when the alternating current(AC) bias direct offset integrated technique(DOIT) with additional positive feedback (APF) is used in a high-Tc dc superconducting quantum interference device (SQUID). The AC-bias DOIT can reduce low-frequency noise without increasing the level of white noise because each operating point in the two voltage-flux characteristics with AC bias can always be optimum on the magnetometer in the high-Tc dc-SQUID. APF can improve the effective flux-to-voltage transfer function so that it can reduce the equivalent flux noise due to the voltage noise of the preamplifier in the magnetometer. The use of APF combined with the AC-bias DOIT reduced the noise of the magnetometer by factors of 1.5 (33µΦ0/Hz vs. 50 µΦ0/Hz) at100 Hz, 3.5 (43 µΦ0/Hz vs. 150 µΦ0/Hz) at 10 Hz, and 5.2 (67 µΦ0/Hz vs. 351 µΦ0/Hz) at 1 Hz as compared with the noise levels that were obtained with the static-current-bias DOIT. The contribution of the factors at 1 Hz is about 2 by APF and 2.6 by AC bias. The performance of improving the flux noise in the AC -bias DOIT with APF is almost equal to that of the flux locked loop (FLL) circuits in which the flux modulation uses a coupling system with a transformer and with the AC bias.

This paper presents an exact minimization algorithm for AND-OR-EXOR three-level networks, where a single two-input exclusive-OR (EXOR) gate is used. The network realizes an EXOR of two sum-of-products expressions (EX-SOP), where the two sum-of-products expressions (SOP) can share products. The objective is to minimize the total number of different products in the two SOPs. An algorithm for the exact minimization of EX-SOPs with up to five variables are shown. Up to five variables, EX-SOPs for all the representative functions of NP-equivalence classes were minimized. For five-variable functions, we confirmed that minimum EX-SOPs require up to 9 products. For n-variable functions, minimum EX-SOPs require at most 92n-5 (n6) products.

A fully differential folded cascode CMOS OP amp is combined with an adaptive bias OTA to increase the slew rate, and a continuous-time CMFB circuit with a push-pull type combination of a NMOS input and a PMOS input differential amplifiers is used to maximize the output voltage swing. The fabricated OP amp using a 0.8 µm digital CMOS process gives more than three times improvement in slew rate with a 15% increase in DC power consumption and a 7.5% increase in chip area compared to the conventional OP amp fabricated on the same die. The output voltage swing was measured to be -0.75 V -0.7 V at the supply voltage of +/-1.2 V.

The present study investigated the human ability to selectively process pictures and words in free recall. We explored whether successful bias towards a subset of priority items occurs at the expense of the remaining items-i.e., whether successful priority item bias necessitates the dumping of information related to non-priority items. It has been shown that an increase in the percentage of correct recalls to items given priority in the pre-test instructions induces a decrease in the percentage of correct recalls for non-priority items. Even in a free recall experimental paradigm, the information dumping phenomenon was observed. However, there were no effects of stimulus presentation time and stimulus modality (picture vs. word) on the percentage of correct recalls detected.

Methods of automatically adjusting delay time and feedback gain in controlling chaos by delayed feedback control are proposed. These methods are based on a gradient-descent procedure minimizing the squared error between the current state and the delayed state. The method of adjusting delay time and that of adjusting feedback gain are applied to controlling chaos in numerical calculations of Rossler Equation and Duffing equation, respectively. Both methods are confirmed to be successful.

This paper proposes a novel FEC (forward error correction) scheme for high-speed wireless systems aiming at mobile computing applications. The proposed scheme combines inner nonredundant error correction with outer parallel encoding random FEC for differentially detected QPSK (quadrature phase shift keying) signals. This paper, first, examines error patterns after the differential detection with nonredundant error correction and reveals that particular double symbol errors occur with relatively high probability. To improve the outer FEC performance degradation due to the double symbol errors, the proposed scheme uses I and Q channel serial to parallel conversion in the transmission side and parallel to serial conversion in the receiving side. As a result, it enables to use simple FEC for the outer parallel encoding random FEC without interleaving. Computer simulation results show the proposed scheme employing one bit correction BCH coding obtains a required Eb/No improvement of 1.2 dB at a Pe of 10-5 compared to that with the same memory size interleaving in an AWGN environment. Moreover, in a Rician fading environment where directional beam antennas are assumed to be used to improve the degradation due to severe multipath signals, an overall Eb/No improvement at Pe of 10-5 of 3.0 dB is achieved compared to simple differential detection when the condition of delay spread of 5 nsec, carrier to multipath signal power ratio of 20 dB and Doppler frequency at 20 GHz band of 150 Hz.