In this paper a high-performance copy network named PPCN is proposed for large scale ATM switching systems. The proposed copy network consists of multiple planes of the P2I Copy Networks(PCN) arranged in parallel. The PCN planes are designed based on the P2I multistage interconnection networks (MINs). A single PCN plane is itself a preliminary self-routing copy network which, however, is not a non-blocking one. A novel dispatcher is designed to dispatch input cells to the PCN planes such that no internal blocking nor output contention arises during the cell replication procedure and the offered load can be shared in an efficient way. The architecture of the PPCN provides flexibility for the maximum fanout for an input cells. In a PPCN system, the maximum fanout for an input cells is determined only by the number of interconnection stages within the PCN planes, independent of the input size of the system. The performance of the PPCN is studied under uniform traffic. It is shown that a small constant number of PCN planes are sufficient for a PPCN system to achieve an acceptable low overflow probability regardless of the system size. The hardware complexity of an N N PPCN is O(N log₂ K) and the length of the routing tag is O(log₂ K) bits, where K is the maximum fanout for an input cell. The storage complexity of the translation tables adopted in an N-inlet PPCN is O(N), which is much lower than that of the previously proposed ones.

In this paper, we investigate a call admission control (CAC) problem in a multimedia wireless ATM network that supports various multimedia applications based on micro/pico cellular architectures. Due to reduced wireless cell size (compared to conventional cellular networks), forced termination of calls in progress becomes a more serious problem in the wireless ATM network. Another problem specific to the multimedia wireless network is how to avoid an excessive delay of non-realtime applications under the presence of various realtime applications with priority over non-realtime applications. We consider two service classes; CBR for realtime applications and UBR for non-realtime applications, and then propose a new CAC scheme that addresses above two problems while minimizing a blocking probability of newly arriving calls of CBR. Through the analytical methods, we derive the blocking probabilities and forced termination probabilities of CBR calls and the average packet delay of UBR connections. We also present a method that decides the optimal CAC threshold values in our CAC scheme.

This paper presents results of subjective assessment of the media synchronization quality in the virtual-time rendering (VTR) media synchronization algorithm. For the assessment, stored voice and video streams were transmitted as two separate transport streams from a source to a destination on various traffic conditions in an experimental system. At the destination, they were output after synchronization control. We subjectively assessed the quality of media synchronization in a systematic way. This paper examines the effects of the difference between methods of recovery from asynchrony on the media synchronization quality. The paper also clarifies the relationships between the subjective and objective performance measures. Furthermore, it examines the effect of the difference in scene between media streams and that of the modification of the target output time on the media synchronization quality.

In this paper, we present a priority scheduling algorithm at ATM switches with multi-class output buffers in which the service rate of each class buffer is dynamically adjusted. The service rate is computed periodically by a control scheme. We derive the design formulas of the control scheme to ensure that each class buffer occupancy converges to its desired operating point related to QoS requirement. Moreover, through dynamic service rate control in the proposed scheduling algorithm, the available channel capacity can be estimated exactly. It may be used for rate control of ABR traffic and call admission control of the other real-time traffic (CBR, VBR, etc. ).

Rate monotonic traffic scheduling algorithm had been shown to be the optimal static priority assignment scheme. The system model studied in can be considered (although not realistic) as a preemptive multiplexer which accepts constant bit rate connections that generate packets periodically. The multiplexer adopts a service discipline such that a lower priority packet can be preempted at any stage by a higher priority one without any loss. The constraint is that every packet has to complete its service before the arrival of its succeeding packet generated by the same connection. In this paper, we study the schedulability problem of rate monotonic schedulers for a fixed-length packet switched network such as the ATM network. A necessary and sufficient condition for a set of m constant bit rate connections to be rate monotonic schedulable is first derived and then utilized to design fast admission control algorithms. One of our algorithms computes in advance the minimum period of a connection which can be accepted given a set of existing connections.

The optimistic analytical results for performance analysis of buffered banyan networks are mainly due to certain independence assumptions used for simplifying analysis. To capture more effects of cell correlation, a refined analytical model for both single-buffered and multiple buffered banyan networks is proposed in this paper. When cell output contention occurs at a 2 2 switch element, two contention resolution schemes are used. One is based on randomly choosing the winning cell and another is to give priority to the cell which has been delayed in the current buffer for at least one stage cycle. The switch throughput, cell transfer delay and cell delay deviation for single-buffered banyan networks with and without using priority scheme are derived. Then the model is generalized to multiple buffered banyan networks where analytical expressions for throughput and delay are obtained. We show that using the priority scheme the cell delay deviation is reduced and the influence on throughput performance is insignificant. The results obtained from our analytical model are compared with the simulations and good agreement is observed. Comparisons with some proposed analytical models in the literature reveal that our model is more accurate and powerful in predicting the performance of buffered banyan networks.

We study resource allocation strategies in ATM switches, which provide quality of service (QoS) guarantees to individual connections. In order to minimize the cell loss rate over a wide range of traffic characteristics, an efficient allocation strategy is necessary. In this paper we introduce a resource allocation strategy, named TP+WRR (Threshold Pushout + Weighted Round Robin) which can fully utilize the buffer space and the bandwidth. We compare the performance of TP+WRR with two typical resource allocation strategies. An exact queueing analysis based on a Markov model is carried out under bursty traffic sources to evaluate their performance. Our results reveal that TP+WRR considerably improves the cell loss probability over the other strategies considered in this paper, especially when many connections are sharing a link.

This paper describes key technologies for establishing a large-scale public switching node system architecture for handling voice and telephony over ATM (VTOA). VTOA is one of the most promising ATM applications, which allows network operating companies to provide a less-expensive but relatively high quality telephone service, by employing voice-data compression and the efficient transmission capabilities of AAL2. We discuss several technical aspects of VTOA handling system architecture, such as the optimum basis for the node, i. e. , STM versus ATM, the appropriate network structure, and suitable signalling. These key points are evaluated from the standpoints of economy, ease of implementation, and extensibility. Our proposed methods should provide the basis for constructing an efficient and cost-effective VTOA handling network.

This paper studies the problem of testing concurrent systems considered as blackboxes and specified using asynchronous Communicating Finite State Machines. We present an approach to derive test cases for concurrent systems in a succinct and formal way. The approach addresses the state space explosion problem by introducing a causality relation model and the concept of logical time to express true concurrency and describe timing constraints on events. The conformance relation between test cases and trace observed from the real system is defined, and a new test architecture as well as a test case application is presented according to the conformance relation defined. To improve verdict capability of test cases, the approach is enhanced by relaxing the unit-time assumption to any natural number. And a computationally efficient algorithm for the enhanced approach is presented and the algorithm is evaluated in terms of computational efficiency and verdict capability. Finally the approach is generalized to describe timing constraints by any real numbers.

IEEE 802. 11 is a standard for wireless LANs. The basic access method in its MAC layer protocol is the distributed coordination function (DCF) for the ad hoc networks. It is based on the mechanism of carrier sense multiple access with collision avoidance (CSMA/CA). DCF is used to support asynchronous data transmission. However, frames in DCF do not have priorities, making it unsuitable for real-time applications. With a little bad luck, a station might have to wait arbitrarily long to send a frame. In this paper, we propose a method to modify the CSMA/CA protocol such that station priorities can be supported. The method is simple, efficient and easy to implement in comparison to point coordination function (PCF), another access method in IEEE 802. 11 based on access points (base stations). Simulations are conducted to analyze the proposed scheme. The results show that DCF is able to carry the prioritized traffic with the proposed scheme.

We propose a new method of progressive transmission of continuous tone images using multi-level error diffusion method. Assuming that the pixels are ordered and the error is diffused to later pixels, multi-level error-diffused images are resolved into a multiple number of bit planes. In an image with 8 bits per pixel, the number of the bit planes that we construct is 9, and the 2-level, 3-level, 5-level,, error-diffused images are produced by a successive use of the bit planes. The original image is finally achieved precisely.

This paper proposes a new cell station (CS) configuration for personal communication systems. The proposed CS employs a modified coherent demodulator with 4-branch maximal ratio combining diversity and a burst-by-burst automatic frequency control (AFC) to enhance the coverage. The proposed CS also employs an antenna-sharing diversity transmission to incorporate more than one transceiver block into a small unit with high power efficiency. With these techniques, the BER performance of the uplink control channel (CCH) is flattened regardless of carrier frequency errors within 12 kHz; the diversity gain of uplink traffic channel (TCH) is improved by 2 dB; the downlink transmission power is reduced by 1.9 dB.

We present a receiver structure with joint blind equalization, carrier recovery, and timing recovery. The blind equalizer employs a decomposition transversal filtering technique which can reduce the complexity of convolution to about a half. We analyze the performance surface of the equalizer cost function and show that the global minima correspond to perfect equalization. We also derive proper initial tap settings of the equalizer for convergence to the global minima. We describe the timing recovery and the carrier recovery methods employed. And we describe a startup sequence to bring the receiver into full operation. The adaptation algorithms for equalization, carrier recovery, and timing recovery are relatively independent, resulting in good operational stability of the overall receiver. Some simulation results for cable-modem type of transmission are presented.

In order to exploit fully the frequency diversity benefits of multicarrier modulation (MCM), and the very nature of the frequency selective radio channel, we investigate an erasure decoded π/4 QDPSK MCM (ED-MCM) by employing simple Hamming (block) code. We propose the threshold-free criteria, i. e. relative minimum receive power test (RMRPT) and relative maximum decision error test (RMDET) for erasure generation and evaluate ED-MCM's performance by applying these tests to average received power, average decision error, instantaneous symbol/bit decision errors. At a normalized delay spread of 1/64, computer simulation results indicate a coding gains of 6.0 - 7.0 dB with ED-MCM at a BER of 10^-3. RMDET/RMRPT based erasure decoding yields a 1.5 - 2.5 dB improvement over the conventional forward error correction (FEC) decoding at a BER of 10^-5. The simulation results at other normalized delay spreads, i. e. , 1/32, 1/16 are also obtained. The erasure criteria (RMRPT and RMDET) applied to average values of received power/decision error yield consistently better performance over error only decoding. The results indicate that the erasure decoding based on relative (threshold-free) measures clearly promises an improved performance of the MCM system.

This paper proposes a multiple open-loop frequency estimation scheme based on differential detection for M-ary phase shift keying (MPSK), which accomplishes fast initial acquisition, precise frequency estimation and wide frequency coverage at the same time. The proposed scheme, which has a good trade-off between complexity and performance, operates as follows: 1) it consists of several frequency error detectors (FEDs) based on differential detection with different delays; 2) it precisely estimates frequency in a wide range (the same range of one symbol differential detection) by open-loop according to frequency errors detected by the FEDs. For real-time symbol-by-symbol operation in order to track fast time-varying frequency, it has a smaller complexity than the other frequency estimation schemes. It is confirmed by analysis, numerical calculation and computer simulation that the frequency estimation error of the proposed scheme is close to the Cramer-Rao lower bound (CRLB) (asymptotic degradation of the proposed scheme from the CRLB is about 0. 5 dB) while keeping a wide frequency coverage and this scheme can track fast time-varying frequency.

This paper is concerned with radio resource allocation in multiple-chip-rate (MCR) DS/CDMA systems accommodating multimedia services with different information rates and quality requirements. Considering both power spectral density (PSD) over a radio frequency (RF) band and the effect of RF input filtering on the receiver in MCR-DS/CDMA systems, criteria for capacity estimation are presented and the characteristics of co-channel interference between subsystems are investigated. System performance in MCR-DS/CDMA systems is strongly affected by radio resource assignment. A minimum power-increment-based resource assignment scheme for an efficient resource assignment scheme is proposed herein. The performance of this scheme is compared with that of a random-based resource assignment scheme in terms of blocking probability and normalized throughput. The minimum power-increment-based resource assignment scheme yields a better performance than the random-based resource assignment scheme for multimedia services.

Adaptive maximum likelihood differential detection implemented by a reduced state-transition Viterbi algorithm (called adaptive 3-state RSTVDD) is presented for adaptive reception of M-ary differential phase shift keying (DPSK) signals transmitted over additive white Gaussian noise (AWGN) and frequency-nonselective Rayleigh fading channels. The adaptive 3-state RSTVDD comprises 1DD, a differential encoder, and reverse modulator, followed by reduced-state (3-state) Viterbi DD (RSVDD) with adaptive phase reference estimation. The adaptive 3-state RSVDD detector estimates the sequence of phase errors of the 1DD output. The phase reference estimator is an adaptive least mean square (LMS) filter with a step-size that adapts to changing channel conditions. The final detected symbol sequence is the modulo-2π sum of the 1DD output phase sequence and the detected phase error sequence. The bit error rate (BER) performance of M-ary DPSK, M=4, 8, and 16, in the AWGN and Rayleigh fading channels is evaluated by computer simulation to show that adaptive 3-state RSTVDD can achieve almost the same BER performance as the previously developed adaptive M-state RSVDD. Since the number of trellis states is reduced to three irrespective of M, the adaptive 3-state RSTVDD has lower computation complexity and it is particularly useful for M-ary DPSK with M8.

Extremely small aperture radial line slot antennas (RLSAs) are analyzed by method of moments. At first, the analysis model of cylindrical waveguide in terms of rectangular cavity modes is confirmed for a RLSA with a spiral slot arrangement. The overall VSWR as well as rotational symmetry of the actual structure of RLSAs is predicted for the first time and is confirmed experimentally. Secondly, the minimum diameter of the concentric array RLSA is estimated for which the conventional analysis model of a rectangular waveguide is valid for the design of matching slot pairs at the shorted periphery of the radial waveguide. It is found that the curvature and cylindrical short wall at aperture periphery must be considered in the design and analysis of small RLSAs with the gain lower than about 25 dBi.

This paper proposes two numerical methods to solve the optimal problem of contrast enhancement in the cross-pol and co-pol channels. For the cross-pol channel case, the contrast (power ratio) is expressed in a homogeneous form, which leads the polarimetric contrast optimization to a distinctive eigenvalue problem. For the co-pol channel case, this paper proposes a cross iterative method for optimization, based on the formula used in the matched-pol channel. Both these numerical methods can be proved as convergent algorithms, and they are effective for obtaining the optimum polarization state. Besides, one of the proposed methods is applied to solve the optimal problem of contrast enhancement for the time-independent targets case. To verify the proposed methods, this paper provides two numerical examples. The results of calculation are completely identical with other authors', showing the validity of the proposed methods.

A packet-based leaky-bucket algorithm functions like the early packet discard (EPD), and accepts a newly arriving packet if the probability that all the cells of the packet are accepted is high. We derive some performance characteristics of the cell and packet arrival processes that are accepted by the leaky-bucket algorithm. From these analyses, a method to determine the values of the parameters of the leaky-bucket algorithm and certain relations between this leaky-bucket algorithm and the generic cell rate algorithm (GCRA) are obtained.

This paper presents a novel approach to pixel decimation for motion estimation in video coding. Early techniques of pixel decimation use regular pixel patterns to evaluate matching criterion. Recent techniques use adaptive pixel patterns and have achieved better efficiency. However, these adaptive techniques require an initial division of a block into a set of uniform regions and therefore are only locally-adaptive in essence. In this paper, we present a globally-adaptive scheme for pixel decimation, in which no regions are fixed at the beginning and pixels are selected only if they have features important to the determination of a match. The experiment results show that when no more than 40 pixels are selected out of a 1616 block, this approach achieves a better search accuracy by 13-22% than the previous locally-adaptive methods which also use features.

Digital logic frequency detector whose operation is based on the analog quadricorrelator is presented. Proposed circuit consists of conventional digital logic devices without an alog elements. Therefore, it has superior reliabilities over component drifts or aging effects. Frequency linear discrimination range is 100% of the reference clock rate.

This letter addresses how to set SIR targets higher than normally required, in order to mitigate the effect of signal-to-interference ratio (SIR) measurement error included in power control steps. We find that scaling up the SIR targets by 1 dB is conformable to resisting SIR measurement error for code division multiple access (CDMA) mobile systems.