In video multicast, it is a challenging task to deliver video at a satisfactory level to multiple receivers with different channel quality and different battery capacities. In order to resolve such receiver heterogeneity, we propose a video multicast system that integrates scalable video coding (SVC) and flexible multicast service (FMS). The system is designed to transmit the base layer (BL) and the enhancement layer (EL) of an SVC-encoded video at different delivery traffic indication maps (DTIMs) such that each receiver can freely select either energy-intensive high quality video or energy-saving baseline quality video. Moreover, we enhance reliability by adding a functionality of intra-flow network coding (NC). As a measure of user satisfaction, we develop a multi-attribute utility model, and then, based on the utility model, we develop a Stakelberg game-based optimization framework to determine modulation and coding scheme (MCS) and the number of NC-generated packets for each layer. Through simulations, we confirm that the proposed scheme achieves higher user satisfaction by exhibiting performance gain of 16.4% to 47% over the legacy 802.11 scheme.

In this paper, we present an energy-per-bit minimized radio resource allocation scheme in heterogeneous networks equipped with multi-homing capability, simultaneously connecting to different wireless interfaces. Specifically, we formulate an optimization problem related to minimization of energy-per-bit which takes a form of nonlinear fractional programming. Then we derive a parametric optimization problem out of that fractional programming and solve the original problem by using a double-loop iteration method. In each iteration, we derive the optimal resource allocation policy by applying Lagrangian duality and an efficient dual update method. In addition, we present suboptimal resource allocation algorithms using the properties of the optimal resource allocation policy. Numerical results reveal that the optimal allocation algorithm improves energy efficiency significantly over the existing resource allocation algorithms designed for homogeneous networks and its performance is superior to suboptimal algorithms in reducing energy consumption as well as in enhancing network energy efficiency.

In this paper, we present a resource allocation algorithm for energy consumption minimization in multi-homed wireless networks, simultaneously connecting to different wireless interfaces. Specifically, we first formulate a general problem related to minimization of energy consumption while satisfying minimum rate requirements. Then, we solve the problem by relaxing integer value constraints and applying Lagrangian Lagrangian dualityduality. For a given set of dual variables, we derive optimal resource allocation policies by using an efficient dual update method. Numerical results reveal that the optimal allocation algorithm outperforms the existing resource allocation algorithms for homogeneous networks in terms of the energy consumption reduction as well as energy efficiency of the overall network.

In this paper, we propose a new feedback scheme called scheduling probability prediction-based selective feedback (SPP-SF) for channel state feedback in OFDMA downlink system. In the SPP-SF scheme, each user first calculates the scheduling probability of each subchannel (i.e., the probability that the user wins the scheduling competition for the subchannel) and then reports the state of a part of the subchannels whose “scheduling probabilities” are higher than the other subchannels. The conventional selective feedback (SF) scheme which reports a part of the subchannels whose “MCS levels” are higher than the other subchannels may be regarded as a subset of the SPP-SF scheme applicable in statistically identical subchannel environment. Numerical results reveal that the SPP-SF scheme achieves significant performance gain over the conventional SF scheme, especially when the statistical characteristics of different subchannels are non-identical.

In this paper, we present an enhanced collision arbitration protocol that operates over multiple antennas in RFID systems. An RFID system consists of a reader and tags, and all tags have to be quickly recognized by a reader. Due to a contention based tag transmission, collision arbitration is a very important issue in tag identification. However, collision always happens in the existing single antenna based collision arbitration scheme when more than one tag simultaneously transmit message. In order to reduce the number of collisions, we propose to use multiple antennas at the reader such that the reader can recognize multiple tags simultaneously by utilizing multi-user receiver at the reader and multiple orthogonal preambles used by each tag. In this scheme, only the tags that are not identified retransmit in the next time slot while the identified tags keep silent. Then the identification delay significantly decreases because the number of retransmissions decreases in comparison with the single antenna scheme. We analyze the performance of this proposed scheme and conduct simulations to confirm the validity of the mathematical modeling. According to the analytical and simulation results, the proposed scheme outperforms the existing single antenna scheme.

In this paper, we present an efficient random access-based resource request protocol that operates over multiple receive antennas. It is a hybrid form of resource request composed of four-way and two-way handshakes. In many of today's wireless systems, a resource request basically requires a four-way handshake between a mobile station (MS) and its base station (BS) because the MS can send a resource request message only after the BS allocates resources for the transmission of the message. In the multiple receive antenna environment, however, it is possible to employ a two-way handshake as well by transmitting a preamble piggybacked by a resource request message: When multiple MSs request resources, the BS first obtains the channel information out of the preambles in the received signals and then separates out the multiple resource request messages by applying multi-user receiver. If they are recovered successfully, the resource request process can be completed by a two-way handshake and, otherwise, it can be completed by a four-way handshake as in the existing resource request schemes. Completing resource request by two-way handshake contributes to decreasing the mean delay. The performance of the proposed schemes is analyzed by adopting a Markov chain model. Numerical results confirm the validity of the mathematical modeling and reveal a notable performance gain of the proposed resource request protocol.

In this paper, we present two new feedback schemes, namely the network mode selection-based feedback (NMF) and the user mode selection-based feedback (UMF), that reduce the feedback load needed for opportunistic scheduling in the OFDMA downlink system. The two schemes are designed to select either the selective feedback (SF) mode which feeds back the fine-grained condition of a part of the channels or the bit-map based feedback (BF) mode which feeds back the coarse-grained condition of all the channels using 1 bit for each channel, depending on the channel conditions of the users. Specifically, the NMF scheme makes the BS determine the more desirable feedback mode by predicting and comparing the average total throughputs of the two modes for the given statistical channel characteristics of the users. The UMF scheme makes each user determine its own feedback mode based on both its channel realization and the recommendation of the network. Numerical results reveal that the proposed NMF and UMF schemes achieve significant performance gain over the existing feedback reduction schemes with similar feedback load.

In this letter, we present an opportunistic multichannel CSMA protocol for OFDMA systems in frequency-selective fading channel. In order to opportunistically exploit the frequency selectivity, an OFDM-based single-channel CSMA protocol can employ water-filling power allocation as it allows a single station to utilize the whole frequency bandwidth at each time instant. In contrast, the proposed OFDMA-based multichannel CSMA protocol achieves the multi-user diversity gain by arranging multiple stations to transmit on their favorable sub-channels simultaneously. The proposed protocol is designed as an opportunistic version of the original multi-channel CSMA protocol, which was proposed in our previous work, and adopts a channel-adaptive backoff (or transmission probability control) algorithm, enabling each station to attempt data transmission on its high-SNR sub-channel with a high probability. The opportunistic multi-channel CSMA turns out to improve the total data rate significantly over the opportunistic single-channel CSMA, and to achieve almost full multi-user diversity gain despite the distributed operation.

In this paper, we present two new feedback reduction schemes, namely the network mode selection-based feedback (NMF) and the user mode selection-based feedback (UMF), to reduce the feedback load needed for opportunistic scheduling in the OFDMA downlink system. The two schemes are designed to select either the selective feedback (SF) mode which feeds back the fine-grained condition of a part of the channels or the bit-map based feedback (BF) mode which feeds back the coarse-grained condition of all the channels using 1-bit for each channel, depending on the channel conditions of the users. Specifically, the NMF scheme makes the BS determine the more desirable feedback mode by predicting and comparing the average total throughputs of the two modes for the given statistical channel characteristics of the users. The UMF scheme makes each user determine its own feedback mode based on both its channel realization and the recommendation of the network. Numerical results reveal that the proposed NMF and UMF schemes achieve significant performance gain over the existing feedback reduction schemes with similar feedback load.

In this paper, we investigate the transmission power allocation for the support of broadcast/multicast services (BMSs) in multi-cell OFDM systems.We consider a cooperative architecture for distribution of the BMS data where the copies of the same data are transmitted by multiple base stations (BSs) in a synchronized manner and then aggregated at the receiving users. Under this architecture, we develop a power allocation scheme that optimizes the transmission power of the multiple BSs on each sub-channel to maximize the total data rate. The optimal power allocation turns out to take the form of iterative water-filling, particularly when heterogeneous modulation and coding schemes are allowed in each sub-channel. In addition, we study the two special cases where the BMS system has extended or limited capabilities: First, we consider the BMS system that supports a more sophisticated video coding technique, fine granularity scalability (FGS) video coding, which encodes the BMS data into base layer and enhancement layer. Taking the layered structure into account, we propose a power allocation scheme that maximizes the data rate of the enhancement layer while guaranteeing the minimum data rate requirement of the base layer. Second, we consider the BMS system that only allows average channel state information of users to be available at the resource manager. In this case, as it is not possible to guarantee successful transmission of every data due to instant deep fading, we propose a power allocation scheme that minimizes the probability of transmission failure. Simulation results reveal that the cooperative distribution system performs far better than the non-cooperative distribution system and the proposed power allocation schemes outperform the equal power allocation scheme significantly.

In this paper, we investigate the capacity region of a multiuser shared channel with time-varying transmission power, whose typical examples are the forward packet data channel (F-PDCH) of IS-2000 1xEV-DV system and the high speed downlink shared channel (HS-DSCH) in 3GPP high speed downlink packet access (HSDPA) system. The concavity of the throughput yields the property that additional power contributes more to a weaker user having a lower channel gain than to a stronger user. Noting this property, we consider a power-balanced policy which tries to balance the received SNR of each user by serving a weaker user with higher transmission power. Then, we establish the equivalence of the power-balanced policy and the Pareto optimal policy which yields a throughput vector at the boundary of the capacity region. This enables to characterize the capacity region explicitly, and also renders an easy means to maximize the total throughput while meeting each user's requirement.

In this paper, we introduce a new retransmission and combining scheme named estimation-based retransmission mode selection (E-RMS) that reinforces the error recovery capability of HARQ for MIMO systems in fast fading channels and present a framework for MIMO system equipped with HARQ capability. The E-RMS scheme is designed to select the more desirable retransmission mode between spatial multiplexing (SM) and space time code (STC) based on estimating the error recovery capability of the two retransmission modes. If the SM mode is selected, the E-RMS transmits all the erroneous streams using the SM mode and otherwise it transmits a part of the erroneous streams using the STC mode. For the E-RMS, we present a decision criterion that selects the more desirable retransmission mode using the channel realization of the previous transmission. Numerical results reveal that the proposed E-RMS achieves significant performance gain over the fixed SM or STC retransmission schemes, performing comparable to the optimal mode selection scheme.

In this paper, we present a multi-channel carrier sense multiple access with collision avoidance (CSMA/CA) protocol for orthogonal frequency division multiple access (OFDMA) systems. The CSMA/CA system in conventional single-channel operation has the advantage of not requiring the signaling for bandwidth request and allocation over the scheduled access system but it sacrifices system efficiency significantly. We overcome the system efficiency limitation by exploiting the OFDMA system features that multiple stations can transmit simultaneously on different sub-channels, and the stations can obtain the channel information of all the sub-channels at each time instant. The OFDMA-based multi-channel CSMA/CA protocol is designed to enable the stations to contend with each other for channel access both in time and frequency domains through a two-dimensional backoff scheme. The protocol takes a generalized form of the conventional single-channel CSMA/CA protocol: It allows to segment the channel bandwidth into multiple narrow-band random access channels and adjust the transmission probability according to the multi-channel activity in a flexible manner. For throughput analysis, we devise a multi-channel p-persistent CSMA protocol that can emulate the multi-channel CSMA/CA protocol and then analyze its saturated throughput. The simulation and analysis results reveal that the proposed system performs far better than the single-channel CSMA/CA system while using the same total bandwidth and even close to the scheduled access system. Further, we examine how stringent time synchronization is required among the constituent stations in implementing of the OFDMA-based multi-channel CSMA/CA system.

In this paper, we present a multi-channel carrier sense multiple access with collision avoidance (CSMA/CA) protocol for orthogonal frequency division multiple access (OFDMA) systems. The protocol exploits the OFDMA system features that multiple stations can transmit simultaneously on different subchannels and the stations can obtain the channel information of all the sub-channels at each time instant. It is designed to enable the stations to contend with each other for channel access both in time and frequency domains through a two-dimensional backoff scheme. By adopting this two-dimensional backoff, the protocol takes a generalized form of the conventional single-channel CSMA/CA protocol: It allows to segment the channel bandwidth into multiple narrow-band random access channels and adjust the transmission probability according to the multichannel activity in a flexible manner. The simulation and analysis results reveal that the proposed system performs far better than the single-channel CSMA/CA system while using the same total bandwidth.

In this paper, we investigate resource allocation in MIMO interference systems where multiple MIMO links share the same frequency band and interfere with each other. For MIMO transmission scheme, we consider a spatial multiplexing scheme that uses linear precoder and decoder. We impose the SINR constraint that the received SINR of each data sub-stream should be larger than a predetermined threshold, and the QoS constraint that the data throughput of individual link should be larger than a required level. In this system, a trade-off exists between the data throughput and the power consumption due to the co-channel interference among the multiple MIMO links. So we arrange a new resource allocation algorithm such that the data throughput and the power consumption are balanced and the QoS constraint is met for each link. We model the resource allocation problem as a noncooperative game and devise a distributed resource allocation algorithm in which the constituent links find a Nash equilibrium point without coordination. The proposed algorithm turns out to perform close to the optimal point achieved by centralized optimization.

In this paper, we analyze the performance of spacetime codes and propose a distance spectrum computation method in fast Rayleigh fading channels. We first derive a new FER upper bound using the union bound and the PEP upper bound in the fast fading environment. The derived FER upper bound is very accurate, requires only the distance spectrum of the spacetime code, and takes a closed-form expression. Then we propose a distance spectrum computation method of space-time codes in fast fading channels, which exploits the symmetric property of the error state diagram in space-time trellis coded MPSK modulation to reduce the computation complexity. Numerical results illustrate that the derived FER bound is tight enough to estimate the performance of space-time codes in fast fading channels with sufficient accuracy.

In this paper, we discuss how to find the optimal method for efficient integration of multicast and unicast service in wireless OFDM systems. The multicast capacity is limited by the user with the worst channel condition. In order to overcome this limitation, we propose channel structure for the multicast and unicast traffic. The multicast channel first encodes the multicast traffic and then demultiplexes it into the constituent subchannels, and the channel structure increases the multicast channel capacity with the help of the channel coding. Based on the channel structure, we develop the subchannel allocation that maximizes the total unicast throughput while guaranteeing the transmission rate requirement on the multicast traffic for all the multicast users. Since it is too complicated to solve the allocation problem directly, we devise a fitness metric that quantitatively measures how suitable the subchannel is for carrying the multicast traffic and present the fitness-based subchannel allocation (FSA) algorithm. Numerical results reveal that the FSA algorithm achieves almost the same performance as the performance upper bound in terms of average unicast throughput.

In this paper, we investigate the problem of minimizing the average transmission power of users while guaranteeing the average delay constraints in time-varying uplink channels. We design a scheduler that selects a user for transmission and determines the transmission rate of the selected user based on the channel and backlog information of the constituent users. Since it requires prohibitively high computation complexity to determine an optimal scheduler for multi-user systems, we propose a low-complexity scheduling scheme that can achieve near-optimal performance. In this scheme, we reduce the complexity by decomposing the multiuser problem into multiple individual user problems. We present the optimal solution to the user problem and then investigate the effects of the user selection algorithm on the performance for different channel models. The proposed user selection algorithm that utilizes both channel and backlog information turns out to be energy-efficient over wide range of delay requirement regardless of the channel model.

In this paper, we discuss how to achieve optimal subcarrier allocation for multicast services in wireless OFDM systems. The multicast capacity is limited by the user with the worst channel condition. To overcome this limitation, we exploit the fine granularity scalability (FGS) video coding which divides multicast data into base layer and enhancement layer data. Then we develop an optimal subcarrier allocation that maximizes the enhancement layer throughput while satisfying the requirement for the base layer by employing linear programming (LP). For a simple subcarrier allocation we present the suitability-based subcarrier allocation (SSA) algorithm which allocates the subcarriers having a higher metric value to the base layer. The SSA algorithm turns out to be equivalent to the optimal solution of LP problem and is easily modifiable for application to practical OFDM systems. In addition, we present the channel condition-based user selection (CCUS) algorithm which selects a proper set of users to serve when the requirement cannot be met. Numerical results reveal that the SSA algorithm achieves the optimal performance in terms of average spectral efficiency and the CCUS algorithm combined with the SSA algorithm significantly improves the performance over the original SSA algorithm.

In this paper, we present new upper and lower bounds of the pairwise error probabilities (PEP) for space-time codes over Rayleigh channels with independent fading. In support of this, we derive a new expression of the PEP using alternative form of the Gaussian tail function, which is very simple and descriptive. The new bounds are very accurate and suitable for easy estimation and application to the performance analysis based on distance spectrum. According to numerical examples, they turn out to be much closer to the exact PEP than other existing bounds. In addition, they help to envision the behavior of the space-time codes for the cases with more than one receiveing antennas.

In this paper, we present new upper and lower bounds of thepairwise error probabilities(PEPs) for space-time codes over Rayleigh channels with independent fading. In support of this, we derive a new expression of the PEP using an alternative form of the Gaussian tail function, which is very simple and descriptive. The new bounds are very accurate, and suitable for easy estimation and application to the performance analysis based on distance spectrum. According to numerical examples, they turn out to be much closer to the exact PEP than other existing bounds. In addition, they help to envision the behavior of the space-time codes for the cases of more than one receive antennas.

In this paper, we rigorously investigate the energy minimization problem for wireless packet networks with automatic repeat request (ARQ) capability. We first formulate the problem for the single-hop case under constrained packet delay and reliability and derive the necessary and sufficient condition for the optimal transmission power at each ARQ stage. We formulate a global rule of optimal transmission power control that achieves optimality regardless of the delay and reliability constraints. Then we extend it to encompass the multi-hop case by dividing the overall problem into two subproblems-energy determination for each ARQ stage and energy distribution among the constituent nodes. We show that the optimality condition established for the single-hop case is also applicable to solve the energy determination problem in the multi-hop case, rendering an optimal solution to the energy distribution problem. Numerical examples reveal that a significant amount of energy is saved by adopting the optimal transmission power and the performance gain is strongly correlated with the decreasing property of the frame error rate

In this letter, we investigate the power minimization problem while supporting the required reliability in wireless multi-hop relay networks. The novelty of the formulation is that it assumes a generalized relation between the power consumed in the transmitter and the packet reception rate in the receiver and, in addition, it takes into consideration the error detection capability of each relay node that can prevent unnecessary error propagation. We derive the optimality condition and provide a method that determines the optimal transmission power of each relay node sequentially. Numerical examples reveal that the optimal power allocation renders notable power saving effect while supporting the required level of reliability.

We present a relay path determination scheme that maximizes the network lifetime and guarantees the end-to-end reliability in the wireless sensor networks with mobile sink. The simple flooding algorithm may render an easy means for the determination but the resulting frequent collisions and profuse relay hops disable determining energy-efficient and reliable paths. To resolve this problem, we present a transmission backoff (TB)- based broadcasting algorithm that uses a transmission deferring approach to reduce the number of collisions and the number of the relay hops, while guaranteeing the end-to-end reliability. It is a distributed and stateless approach in which the relay path is determined by each node without any global and/or local exchange of network information. Each node decides whether or not to rebroadcast based on its own transmission backoff time, which is calculated using its residual battery energy and the target end-to-end reliability. Simulation results reveal that the proposed algorithm can achieve a longer network lifetime and a higher end-to-end reliability performance than other existing schemes.

In this paper, we present a distributed resource allocation algorithm for multi-cell OFDMA systems by adopting a game theoretic approach. We first define a utility function that can represent the system performance while taking into account the co-channel interference among cells. Then we model resource allocation problem as a noncooperative game. Based on the game, we devise a iterative algorithm called DRA that requires no coordination among the base stations. We prove that there exists a Nash equilibrium for the game and the equilibrium is unique in some constrained environment. The proposed DRA algorithm turns out to converge to the equilibrium within a small number of iterations.

In this paper, we investigate the energy minimization problem for wireless packet networks with automatic repeat request (ARQ) capability. We formulate the problem under constrained packet delay and reliability, and derive a necessary and sufficient condition for optimal transmission power that achieves optimality regardless of the delay and reliability constraints. Numerical examples reveal that a significant amount of energy is saved by adopting the optimal transmission power and the performance gain is strongly correlated with the decreasing property of the frame error rate.

In this paper, we provide an efficient quality-of-service (QoS)-guaranteeing scheme using the cdf-based scheduling (CS) algorithm in wireless fading channels. We first extend the CS algorithm such that it can encompass the practical environment with discrete user transmission rates. The extended CS algorithm can allocate the time fractions to users in an arbitrary manner, and render an exact estimation of user average throughput, through which it can provide differentiated QoS to each user. We also introduce the effective capacity concept to describe the delay-constrained capacity of the CS algorithm, both in time-independent and time-correlated channels. In contrast to other existing scheduling algorithms, the CS algorithm enables calculating the effective capacity analytically, rather than estimating it by measurement on the queuing behavior. Using the effective capacity, we can check the feasibility of the user-specified QoS effectively in wireless time-varying channels.

In this paper, we provide an efficient quality-of-service (QoS)-guarantee scheme using the cumulative-distribution-function-based scheduling (CS) algorithm in wireless fading channels. We first extend the CS algorithm such that it can encompass the practical environment with discrete user transmission rates. The extended CS algorithm can allocate the time fractions to users in arbitrary manner, and render an exact estimation of user average throughputs, through which it can provide differentiated QoS to each user. We also introduce the effective-capacity concept to describe the delay-constrained capacity of the CS algorithm, both in time-independent and time-correlated channels. In contrast to other existing scheduling algorithms, the CS algorithm enables calculating the effective capacity analytically, rather than estimating it by measurement on the queueing behavior. Using the effective capacity, we can check the feasibility of the user-specified QoS effectively in wireless time-varying channels

We investigate the problem of the lifetime maximization in wireless sensor networks under the constraint of end-to-end transmission success probability, by adopting a cross-layer strategy that considers physical layer (i.e., power control), MAC layer (i.e., retransmission control) and network layer (i.e., routing protocol) jointly. We decouple the problem into separate sub-problems of each layer and propose the optimal algorithm as well as an alternative heuristic algorithm with lower complexity for each sub-problem. We demonstrate through computer simulations that a trade-off relation exists between the network lifetime maximization and the reliability constraint, and the strategy that is designed by combining the proposed algorithm at each layer can significantly increase the network lifetime. We also investigate the effect of the retransmission control on the energy efficiency for different energy consumption models. Simulation results reveal that multiple retransmissions with low power yield little gain when the link distance is short and the power conversion efficiency of the amplifier increases with the transmission power

In this paper we investigate adaptive resource allocation schemes in multiuser OFDM systems for fair share of resources and efficient operation. We employ the CDF-based scheduling (CS) algorithm for the subcarrier allocation, taking advantage of its distinctive feature of analyzability and multiuser diversity. Noting that conventional power allocation schemes do not exhibit efficient and fair operations in heterogeneous user channel environment, we present a new algorithm called proportional-fair power allocation (PFPA). This algorithm is designed to allocate transmission power in such a way that the resulting relative throughput-increment is identical for all subcarriers. The PFPA algorithm is shown to be equivalent to the power allocation of the asymptotically optimal algorithm, which exhibits the largest achievable region in the asymptotic case. Numerical results reveal that the combined CS-PFPA algorithm improves the overall system capacity in terms of time-average throughput and provides efficient estimation of user performances. Further, the CS-PFPA algorithm can meet each user's requirements using a minimum amount of resources, so it renders an efficient and fair means for resource allocation in multiuser OFDM systems.

We investigate the problem of lifetime maximization in a wireless sensor network under the constraint of target end-to-end transmission success probability, by adopting a cross-layer strategy that considers physical layer (i.e., power control), MAC layer (i.e., ARQ control) and network layer (i.e., routing protocol) jointly. We first present a near-optimal retry limit allocation algorithm for a given routing path. This allocation determines the per-hop success probability for each link along the path in order to minimize the total energy consumption while guaranteeing the reliability constraint. Then, we develop an optimal routing and power control algorithm that maximizes the network lifetime while keeping the network stable, and an alternative heuristic algorithm that has lower and tractable complexity. Simulation results reveal that a trade-off relation exists between the network lifetime maximization and the reliability constraints and that the network lifetime can be increased significantly by employing the proposed algorithms.

In this paper, we investigate the capacity region of a multiuser shared channel with time-varying transmission power, whose typical example is the forward packet data channel (F-PDCH) of IS-2000 IxEV-DV system. The concavity of the throughput yields the property that additional power contributes more to a weaker user having a lower channel gain than to a stronger user. Noting this property, we consider a power-balanced policy which tries to balance the received SNR of each user by serving a weaker user with higher transmission power. Then, we establish the equivalence of the power-balanced policy and the Pareto optimal policy which yields a throughput vector at the boundary of the capacity region. This enables to characterize the capacity region explicitly, and also renders an easy means to maximize the total throughput while meeting each user's requirement.

In this paper, we investigate the problem of the lifetime maximization in a wireless sensor network under the constraint of the target end-to-end transmission success probability, by adopting a cross-layer strategy that considers both physical layer (i.e., power control) and network layer (i.e., routing protocol) jointly. We first present the optimal power allocation algorithm for a given routing path. This allocation determines per-hop success probability for each link along the path in order to minimize the total energy consumption while guaranteeing the reliability constraint. Then, we develop an optimal routing algorithm that maximizes the network lifetime and a heuristic algorithm that has lower and tractable complexity. Simulation results reveal that a trade-off relation exists between the network lifetime maximization and the reliability constraint, and that the proposed routing algorithm combined with the optimal power allocation algorithm increases the network lifetime significantly.

In this paper, we present a new wireless scheduling algorithm based on the cumulative distribution function (cdf) and its simple modification that limits the maximum starving time. This cdf-based scheduling (CS) algorithm selects the user for transmission based on the cdf of user rates, in such a way that the user whose rate is high enough, but least probable to become higher, is selected first. We prove that the CS algorithm is equivalent to a scheduling algorithm that regards the user rates as independent and identically distributed, and the average throughput of a user is independent of the probability distribution of other users. So, we can evaluate the exact user throughput only if we know the user's own distribution, which is a distinctive feature of this proposed algorithm. In addition, we try a modification on the CS algorithm to limit the maximum starving time, and prove that the modification does not affect the average interservice time. This CS with starving-time limitation (CS-STL) algorithm turns out to limit the maximum starving time at the cost of a negligible throughput loss.

In this paper, we investigate how to maximize network lifetime, while meeting the constraints on data collecting time, or data convergence latency (DCL), in networks consisting of N energy-constrained sensor nodes and a central base station (BS). Noting the trade-off relation that a data packet delivered to the BS requires less transmission energy through multi-hop relays but more DCL, we first formulate an optimal routing algorithm that maximizes the network lifetime while satisfying the DCL constraint determined by latency-sensitivity of the application or network stability conditions. As this optimal solution is too complex to implement in practice, we modify it to a suboptimal latency-constrained cost minimization (LCCM) algorithm that is close to the optimal algorithm in performance but is; much lighter in complexity. The LCCM algorithm is designed to minimize the total link cost, while satisfying the DCL constraint. Simulation results show that the performance of the proposed LCCM algorithm is slightly marginal to that of the optimal algorithm in terms of the lifetime, in various network environments.

In this paper we investigate adaptive resource allocation schemes in multiuser OFDM systems for fair share of resources and efficient operation. We employ the CDF-based scheduling (CS) algorithm for the subcarrier allocation, taking advantage of its distinctive feature of analyzability and multiuser diversity. Noting that conventional power allocation schemes do not exhibit efficient and fair operations in heterogeneous user channel environments, we present a new algorithm called proportional-fair power allocation (PFPA). This algorithm is designed to allocate transmission power in such a way that the resulting relative throughput-increment is identical for all subcarriers. The PFPA algorithm is shown to be equivalent to the power allocation of the asymptotically optimal algorithm which exhibits the largest achievable region in the asymptotic case. Numerical results reveal that the combined CS-PFPA algorithm improves the overall system capacity in terms of time-average throughput and provides efficient estimation of user performance. Further, the CS-PFPA algorithm can meet each user's requirements using a minimum amount of resources, so it renders an efficient and fair means for resource allocation in multiuser OFDM systems.

In this paper, we analyze the performance of space-time codes by deriving a new approximation of the frame error probabilities for space-time trellis-coded modulations over quasi-static Rayleigh-fading channels. We take advantage of two techniques, the modified bounding technique and the limiting-before-averaging technique, to tighten the upper bound. In addition, we establish a theorem that enables us to reduce the computation and memory needed to calculate the frame error rate (FER) approximation. The newly derived approximation is very tight, requires only the distance spectrum of the space-time code, and can be computed through single numerical integration. Numerical results exhibit that the new approximation is much closer to the simulation results than other existing bounds are, especially in the case of one receive antenna.

We present a new wireless scheduling algorithm based on the cumulative distribution function (cdf) of user transmission rates and also present a simple modification of it to limit the maximum starving time. This cdf-based scheduling (CS) algorithm selects the user for transmission based on the cdf of user rates in such a way that the user whose rate is high enough but least probable to become higher is selected. It turns out that the cdf-based scheduling algorithm is equivalent to a scheduling algorithm that regards the user rates as independent identically distributed (i.i.d.) and the average throughput of a user is independent of other users' probability distribution. A distinctive feature of this proposed algorithm is that the exact user throughput can be evaluated if the user's own distribution is known. The cdf-based scheduling with starving-time limitation (CS-STL) algorithm turns out not to affect the average inter-service time but to limit the maximum starving time at the cost of a negligible throughput loss.

In this paper, we analyze the performance of ST-BICM (space-time bit-interleaved coded modulation) system employing the arbitrary constellation in quasi-static fading channels. We derive a new upper bound of the frame error probability of the system, which depends only on the product measure. In order to get the product measure, we analyze the type of error sequence, which is defined as the density spectrum of the constituent symbols that yield a certain number of bit errors. The analysis is based on the uniform interleaving, so we average the frame error probability over all possible interleavers. We also derive the statistical distribution of the product measure. Finally, we show that the new upper bound helps to predict the performance of the ST-BICM system with an arbitrary constellation.

In this paper, we analyze the performance of space-time codes and propose a distance spectrum computation method in fast Rayleigh fading channels. We first derive a new FER upper bound using the union bound and the PEP upper bound in the fast fading environment. The derived FER upper bound is very accurate, requires only the distance spectrum of the space-time code, and takes a closed-form expression. Then we propose a complexity reduction method of computing the distance spectrum of space-time codes in fast fading channels, which exploits the symmetric property of the error state diagram. Numerical results exhibit that the derived FER bound is tight enough to estimate the performance of space-time codes in fast fading channels with sufficient accuracy.

We first present a new method of determining the upper bounds of maximal eigenvalue of Hermitian positive-definite matrix, which includes the Dembo's (1988) upper bound as a special case. Then we derive the kth-order upper bound /spl Lambda//sub k/ and its first-round iteration /spl Lambda//sub k//sup (1)/ which is much tighter than the Dembo's upper bound /spl Lambda//sub 0/.

We present an active buffer control based TCP agent scheme that can overcome TCP performance degradation caused by handovers in broadband wireless networks. The agent performs an active buffer control and adjusts the sending rate of the TCP server by forcing three duplicate acknowledgements while it splits an end-to-end TCP connection into two TCP connections. So, high utilization is achievable even with frequent handovers. We also propose an active buffer control algorithm deployed in the agent and a guideline on design of the buffer control parameters using bandwidth-delay product. Simulation results reveal that high TCP throughput can be achieved in comparison with the Fast Retransmit scheme and the traditional TCP for the single traffic scenario without bottleneck, especially for the scenario with bottleneck in wired links. The proposed TCP agent scheme can be easily implemented with the traditional TCP without modifications at the end terminals.

We analyze the performance of space-time codes. In particular, we derive an upper bound of the frame error probabilities for space-time trellis coded modulations over quasi-static Rayleigh fading channels. Two techniques, modified bounding and limiting before averaging, are used to derive the new bound. The newly derived upper bound is very accurate, needs only the distance spectrum of the space-time code, and can be computed through single numerical integration. Numerical results show that the new upper bound is much tighter than other existing bounds, especially in the case of one receive antenna.

In this paper we present new upper and lower bounds of the pairwise error probabilities (PEP) for space-time codes over Rayleigh channels with independent fading. We first derive a new expression of the PEP using an alternative form of the Gaussian tail function, based on which we then derive new upper and lower bounds. The new bounds are very accurate and suitable for easy estimation and application to the performance analysis based on distance spectrum. According to numerical examples, the new bounds turn out to be much closer to the exact PEP than other existing bounds. In addition, they help to envision the behavior of the space-time codes for the cases or more than one receive antennas.

We introduce a correlation-aided distributed sample acquisition (CDSA)-based intercell synchronous direct-sequence code-division multiple-access (DS/CDMA) system for enhancing the robustness of the previously proposed distributed sample acquisition (DSA)-based system. The CDSA system incorporates the state symbol correlation process in the existing DSA process to maintain the fast acquisition performance even under very poor channel environment. For its realization, each CDSA-based mobile station stores in its memory a period of the possible state symbol sequences, which are determined by the long-period scrambling PN sequences used in the system and the sampling time parameters for the state samples. In the first stage of the synchronization procedure, the mobile station (MS) attempts to acquire the psuedo noise (PN) sequence by taking the comparison-correction based synchronization approach of the original DSA scheme. However, if it cannot acquire the synchronization until it collects a predetermined number of state symbols (e.g., a period of the state symbol sequence), then the MS determines the PN sequence timing by correlating the received symbol sequence with each shift of the prestored state symbol sequences. Performance analysis and simulation results reveal strong robustness of the CDSA-based system in low-signal-to-noise ratio, fading, and large frequency-offset channels.

In this paper, we extend the previously presented QoS routing scheme called pre-computation based selective probing (PCSP) to the practical environment where the state information available for the routing decision is not precise. This extended PCSP algorithm pre-computes the QoS variations as well as the cost and QoS metrics of the least-cost and best-quality paths, taking into account the impreciseness of state information. The pre-computed information enables to strictly limit the set of neighbor nodes involved in the probing process, thereby reducing the message complexity without sacrificing cost optimality. This extended PCSP scheme encompasses the original PCSP scheme as a special case where the variation of state information reduces to zero. Computer simulation reveals that the extended PCSP scheme maintains low message complexity and high success ratio with guaranteed optimal search.

We study a gateway congestion control scheme for TCP traffic in an internetwork consisting of both rate-controlled and non-rate-controlled links. In this network scenario, TCP window-based congestion control and rate-based congestion control at a rate-controlled link may not interact in a harmonious way. So, we present a packet dropping scheme called rate adaptive dropping (RAD) as a means for reconciling the mismatch between window-based and rate-based congestion controls. The main idea of RAD is to drop arriving packets depending on the rate feedback information so that dropped packets may notify TCP sources of incipient congestion before the gateway buffer overflows. We compare the performance of RAD with that of tail drop in terms of goodput and drop ratio. Results from simulations show that RAD achieves significant improvements over tail drop, especially when the gateway buffer size is around the bandwidth delay product or less.

This paper presents an iterative decoding method in the coded MFSK (multilevel frequency shift keying)/FH-SSMA (frequency hopping-spread spectrum multiple access) system. The kernel of the system, which is a symbol APP (a posteriori probability) calculator, accepts channel values and a priori information from the channel decoder and generates reliability values of the received symbols. The channel decoder also makes reliability outputs using the reliability values from the symbol APP calculator. Iterative decoding is performed by repeating this process. Simulation results reveal that the channel coded system with iterative decoding in the FH-SSMA system can reduce the BER significantly, thereby accommodating more users.

We present a distributed dynamic reservation (DDR) scheme that supports mobility in wireless multimedia communications, noting that low handover blocking and high channel utilization are possible if channel reservation can be done dynamically. As the traditional reservation schemes do not fit to handling time-varying multiclass multimedia traffic due to the involved computational complexity, we employ an elaborate two-regional approximation scheme that can reduce the computation dramatically. We approximate the channel occupancy distribution based on the observation of arrival rates, means, and variances of total calls and handover calls, which can be easily measured locally at each base station or switch in a distributed manner. The approximation is made by arranging the distribution into two regions such that a simple distribution model can be applied in each region and then joining the two regions using legitimate boundary conditions. This approximation enables us to estimate the relevant number of reservation channels very quickly, with the computational complexity reduced to the order of O(log C) for the channel capacity C. Nevertheless, the estimation turns out very close to the exact solution determined by applying the multidimensional Markov chain approach. Simulation results reveal that the proposed DDR scheme can adapt itself well to time-varying multiclass multimedia traffic and achieve high channel utilization, yet maintain a very low handover blocking probability.

We propose a parallel distributed sample acquisition (PDSA) scheme applicable to a direct-sequence code-division multiple-access (DS/CDMA) system employing M-ary signaling for data transmission. While the primitive distributed sample acquisition scheme can be employed only for the binary signaling systems which transmit 1 bit per symbol, the proposed PDSA scheme extends its applicability to the general signaling systems allowing multibit transmission per symbol. The proposed PDSA technique can also be applied to fast acquisition of the scrambling code in multicarrier DS/CDMA cellular systems with a slight modification of the transceiver circuitry.

The paper presents a new spread spectrum communication system called orthonormal code diversity detection (OCDD)/CDMA system based on the novel concept of orthonormal-basis diversity which is a generalization of the existing spread spectrum diversity concepts such as path diversity and frequency diversity. The OCDD/CDMA system is similar to the conventional DS/CDMA system in the transmitter structure, but is different in the receiver structure as it employs the extended orthonormal basis-function set which is the union of the Walsh basis-functions multiplied by the PN sequences and, optionally, their delayed replicas. The received signal is matched to the extended basis functions, and the matched signal components are combined together after individual channel compensation. The proposed OCDD/CDMA system exhibits the bit error performance which is much improved over the conventional DS/CDMA system using maximal ratio combing. In addition, it is robust to the chip timing error, which becomes more crucial in the future DS/CDMA systems having a higher data rate and smaller chip interval. From the simulation results, we confirm that the OCDD/CDMA system is a unique spread spectrum communication technique that can effectively increase the diversity utilization in the slowly fading channel, overcoming the inherent problems in the DS/CDMA and OFDM/CDMA systems.

We present DPSK-based distributed sample acquisition (D/sup 2/SA) as a novel variant of the distributed sample acquisition family for application to the M-ary phase-shift keying signaling direct-sequence code-division multiple-access systems. The D/sup 2/SA employs differentially coherent phase-shift keying (DPSK) to convey the shift register generator state samples and the prerotation of the data constellation to realize the coherent demodulation of the data stream. The use of DPSK instead of orthogonal modulation as the means of the state sample conveyance improves the system performance in terms of both system complexity and acquisition time.

We propose a new decorrelating detector for the asynchronous code-division multiple-access channels, called the extended-window decorrelating detector (EW-DD). The EW-DD helps to avoid the noncausality problem of the filter-type decorrelating detector and achieve a simplified detection by converting the asynchronous system into the synchronous one. It also helps to enhance the bit-error rate (BER) performances by utilizing one whole period of information of all users and by eliminating the multiple-access interference from the past data through feedback. Nevertheless, the computational complexity is kept lower than that of other existing decorrelating detectors. We carry out BER analysis of the EW-DD and demonstrate performance improvements of the EW-DD over other decorrelating detectors through computer simulations.

We propose a new distributed QoS routing scheme called PCSP that uses pre-computation based selective probing for routing decisions. This algorithm pre-computes the cost and QoS metrics respectively for the least cost and the best QoS paths. The pre-computed information enables to strictly limit the set of neighboring nodes involved in the probing process, thereby reducing the message complexity without sacrificing cost optimality. Computer simulation reveals that the PCSP scheme indeed has low message complexity and high success ratio with guaranteed optimal search.

We introduce a fast fell-search scheme based on the distributed sample acquisition (DSA) technique for inter-cell asynchronous DS/CDMA systems. The proposed DSA scheme enables the long-code acquisition extremely fast even without employing any passive matched filter which used to dominate synchronization circuit complexity in asynchronous DS/CDMA systems. In the DSA-based asynchronous DS/CDMA system, each base station spreads its DQPSK-modulated long-code generator information with an assigned short-period igniter sequence, and broadcasts it as a common pilot signal. A mobile station first identifies and acquires the igniter sequence of the cell group currently located, then detects the conveyed long-code generator information by despreading the acquired igniter sequence, thereby identifying and synchronizing the long-code of the current cell. The mean acquisition time of this proposed DSA-based inter-cell asynchronous DS/CDMA system, evaluated under the assumption that each of 512 cell-specific long-codes is a 10-ms segment (38400-chips for the chip rates of 3.84 Mchips/s) of complex Gold codes of period 2/sup 18/-1 and the igniter sequence (or, group code) set is composed of 7 complex orthogonal Gold codes of period 256, turned out even shorter than that of the inter-cell synchronous DS/CDMA systems employing the conventional serial search method. Furthermore, simulation results showed that the DSA scheme employing a passive matched filter is much superior in terms of acquisition time and robustness, to the 3GPP W-CDMA synchronization scheme having comparable complexity.

We consider the issue of source clock frequency recovery in packet networks and propose a new adaptive clock method based on the Kalman filter (KF) with low-pass prefiltering-KALP (Kalman filter-based Adaptive clock method with Low-pass Prefiltering) in short. Noting that because of the difficulty in modeling as well as the nonwhite Gaussian nature of the packet jitter most existing adaptive clock methods could not successfully adopt the Kalman filter, we take a new approach to packet jitter modeling for the KALP. We model the packet jitter not directly but after shaping its characteristics by low-pass prefiltering. This low-pass prefiltering is an important arrangement as it helps to convert the packet jitter into a low-pass signal regardless of its original characteristics, thus enabling to model the prefiltered packet jitter using a simple first-order autoregressive [AR(1)] process. The low-pass prefilter should be carefully selected not to lose the timing information while prefiltering, and the moving averager employed in this paper satisfies this requirement. The AR(1)-modeled jitter component is amenable to the KF-based processing, which in this case becomes an optimal estimator. The design parameters including the initial conditions of the KF and AR(1) parameters can be determined based on the service clock specification and packet interarrival times during the delay smoothing process. We carry out various simulations to compare the performance of the KALP with the existing buffer-based adaptive clock method and demonstrate that the KALP can significantly reduce the fluctuation in the level of receiving buffer as well as the time to recover the source clock frequency.

We present distributed dynamic reservation (DDR) scheme that supports mobility in the wireless multimedia communications. As the traditional reservation schemes do not fit to handling time-varying multi-class multimedia traffic due to the involved computational complexity, we employ an elaborate two-regional approximation scheme that can reduce the computation dramatically. We approximate the channel occupancy distribution based on the observation of arrival rates, means, and variances of total calls and handover calls, which can be easily measured at each base station or node in a distributed manner. This approximation enables us to estimate the relevant number of reservation channels very fast, with the computational complexity reduced to the order of O(logC) for the channel capacity C. Nevertheless, the estimation turns out very close to the exact solution determined by applying the multi-dimensional Markov chain approach. Simulation results reveal that the proposed DDR scheme can adapt itself well to time-varying multi-class multimedia traffic and achieve high channel utilization, yet maintaining a very low handover blocking probability.

We propose a new acquisition scheme called distributed sample acquisition (DSA) which is applicable to the direct-sequence code-division multiple-access (DS/CDMA) system with a long-period pseudonoise (PN) sequence. The proposed scheme lays its foundations on an auxiliary PN sequence called the igniter sequence, which has relatively short period, and the distributed sample scrambling (DSS) technique previously developed for the asynchronous transfer mode cell transmission. In the DSA scheme, acquisition is done in three steps. First, the igniter sequence is synchronized, which normally takes a very short time due to its short period. Second, the state samples of the long-period PN sequence generator in the transmitter, which are conveyed to the receiver over the igniter sequence stream, are determined out of the synchronized igniter sequence. Third, the long-period PN sequence is synchronized by applying the state samples to the DSS receiver. Since the operation of each step takes a very short time, the overall acquisition is done very quickly. We analyze the acquisition time performance of the proposed scheme by taking the transform domain approach, confirming that the resulting mean acquisition time is dramatically reduced. If compared with the existing serial-search scheme, the DSA scheme can perform acquisition about 100 times faster when the period of the PN sequence is 2/sup 15/-1. None the less, the additional circuit complexity for its implementation is very small.

We analyze the acquisition time performance of the distributed sample acquisition (DSA) scheme proposed for a fast DS/CDMA signal acquisition. To analyze the composite acquisition mechanism, we first define several acquisition-related probability parameters which can describe all the essential events in the DSA synchronization, along with the basic states that constitute the eventual DSA state transition diagram. Then, we construct the state transition diagram employing these basic states and the moment generating functions which are entirely based on the above defined probability parameters. Applying this state transition diagram and determining the acquisition-related parameters, we finally analyze the mean acquisition time performance of the DSA.

We rigorously examine how to construct a large-scale distributor by taking a logical and systematic approach. We first consider requirements for realization, which include the independent operation of unit modules and the packet sequence integrity, and consider what kind of existing devices have large-scale growth potential. In this process, we take the Batcher sorting network as the candidate device and show that the distribution function can be incorporated to it by furnishing each constituent sorting element with the state inversion capability. Then we show that this is equivalent to replacing each sorting element with a 2/spl times/2 distributor. As a consequence, the large-scale distributor design problem turns into a simple substitution problem for which we substitute a unit distributor for each sorting element in the prototype Batcher sorting network. In addition, we show that this substitution approach can be extended to larger scale unit distributor modules, thus enabling a large-scale distributor realization. The resulting large-scale distributor design procedure can be summarized as follows: given a desired size L=MN for the design of a large-scale distributor, we first design an N/spl times/N Batcher sorter (BS) and then replace each constituent 2/spl times/2 sorting element with a 2M/spl times/2M distributor module.

In this paper, we study the use of ABR connections for transporting TCP traffic over paths where ATM is only a part of the whole path, the subnet ATM model. We show that when ABR connections are used, both throughput and fairness may be improved by using intelligent gateway congestion control mechanisms. Specifically, we propose the ABR+FB (fair buffering) gateway congestion control scheme based on the observation that the poor performance of TCP over ABR connections is mainly due to the mismatch between the ABR rate control mechanism and the TCP window based congestion control mechanism. We show that the proposed ABR+FB mechanism improves the performance of TCP over ABR connections with respect to both throughput and fairness.

This paper introduces a new acquisition scheme called distributed sample acquisition (DSA), which is applicable to the DS/CDMA system with a long-period PN sequence. The DSA scheme employs two new concepts, igniter sequence and distributed samples, for signal acquisition. The igniter sequence is a short-period sequence that conveys the main-sequence samples of the transmitter to receiver in a distributed manner, which are thus called distributed samples. In synchronizing the igniter sequence, we employ normal serial search schemes, but for synchronization of the main long-period PN sequence we employ the novel distributed sample scrambling (DSS) technique. The resulting mean acquisition time of the DSA scheme is dramatically reduced for the long period PN sequence, and the additional circuit complexity for its implementation is very small.

In this letter we consider two efficient external control input (ECI) value generation methods for the controlled switching element (CSE)-based distributor, which is a new distributor structure that does not require the dummy address generation and extraction operations of the conventional distributors. The first method relies on the active packet counter and the ECI generator for ECI value generation, and the second method utilizes the intrinsic ECI generation property of the CSE-based reverse banyan network (RBN). The first method brings forth a flexible distributor structure, while the second method renders a very simple structure. In fact, the second method yields the so-called autonomous distributor, consisting only of a CSE-based RBN and a set of delays, which is perceived as the simplest among all available distributors. In addition, we introduce a modified version of the autonomous distributor which can speed up the connection-state setup process.

We consider how to improve the performance of the TCP connections over personal communication networks by introducing the auxiliary timeout and selective packet discard schemes. Because TCP is a reliable transport protocol tuned to reliable wired networks, we need a data link protocol to supplement the TCP to overcome unreliable error environment in wireless networks. In this case, however, the data link protocol can interact with the TCP and thus influence the TCP performance. According to computer simulations, when the selective repeat ARQ is used as the data link protocol in the fading channel, the throughput decreases as the fading frequency increases; and when the fading frequency is low, TCP retransmission occurs spuriously and the throughput decreases due to delay variation and the misunderstanding of congestion. In order to solve these problems we introduce the auxiliary timeout and selective packet discard schemes. The auxiliary timeout keeps the rule of the congestion control in traditional networks and avoids spurious retransmissions and misbehaved congestion controls by acknowledging the delayed ACK. The selective packet discard is designed to avoid retransmitting data packets that the mobile station has acknowledged or will acknowledge soon at the base station, thus saving the wireless bandwidth. Therefore the proposed schemes can help to improve the TCP performance in wireless networks by increasing throughput and reducing spurious retransmissions.

In this paper, we consider the behaviors and performances of the three-state-based synchronization mechanisms of DSS in the errored environment. We establish a mathematical model to describe the synchronization mechanisms by employing the sequence space theory, and discuss the behaviors of the two different three-state synchronization schemes-the ITU-T recommended thresholded-counting scheme and the newly proposed windowed-observation scheme. We consider the design guidelines for the synchronization schemes, and finally compare their performances in the cell-based ATM.

In this paper, we consider how to apply the sequence space and the shift register generator (SRG) theories to distributed sample scrambling (DSS), which exhibits the best performance in scrambling of small frame-sized signals. We first consider how to predict scrambling sequences using their samples, which is a basic problem for a proper synchronization within the DSS systems. Then for DSS scramblers, we consider how to determine the scrambler SRGs and the sampling times for conveying information on the scrambling sequence; and for DSS descramblers, we consider how to determine the descrambler SRGs, the correction times, and the correction vectors for a proper synchronization. We further examine how to realize the scramblers and the descramblers with minimized circuit complexity. Finally, we demonstrate how to apply the results to DSS scramblers and descramblers for application in cell-based asynchronous transfer mode (ATM) transmission.

We consider how to realize parallel shift register generators (PSRG) and multibit PSRGs, which can be directly used for parallel frame synchronous scrambling (FSS) in the bit- and multibit-interleaved multiplexing environments. We first describe the structure of PSRGs in terms of three parameters-the state transition matrix, the initial state vector, and the generating vectors. Then we discuss how to determine the three parameters of PSRGs that generate the desired parallel sequences in general. We further develop a method for the realization of minimum length PSRGs, and for the realization of PSRGs with minimized circuit complexity. Finally, we consider how to realize minimal PSRGs for use in multibit-parallel scrambling. The results are summarized in four sets of theorems, and are demonstrated through four examples.

In this paper, a theory is developed in support of parallel realizations of the distributed sample scramblers (DSS) in the cell-based ATM transmission environment. For the relevant parallel realizations of the SRG (shift register generator) engines and the SRG sequence-generating circuits, the PSRG theory is applied which is readily available for parallel FSS (frame synchronous scrambler). For the synchronization of the parallel DSS, the double-sampling and double-correction schemes are employed as they efficiently match the two adjacently located sample conveyance channels within the ATM cells. The resulting synchronization conditions are described in terms of three theorems-one regarding the sampling time and vector's condition, and the other two regarding the correction time and vector's conditions for even- and odd-lengthened SRGs. Demonstration is given at the end on the use of the developed theory for parallel realizations of DSS in scrambling and descrambling the ITU-T recommended cell-based ATM signals.

In this paper, a theory is developed to support the synchronization of shift register generators (SRGs) in the general class of distributed sample scramblers (DSSs). In the general DSS, the samples transmitted to the descrambler for its synchronization are freely generated out of the scrambler SRG, a special case of which is to take samples directly from the SRG sequence. To describe this general sampling method, the concept of sampling vectors is introduced. The delayed sample-transmission problem, which is a real problem in practical transmission systems, can be resolved by adopting the sampling vectors. This paper provides a systematic solution to the synchronization problem of the general DSS based on mathematical modeling. It first considers the sampling problem of the SRG state information, and then discusses the correction problem of the descrambler SRG state using the transmitted samples. Applications are attached at the end to demonstrate the developed theory for simple realizations of the DSS for use in the cell-based ATM transmission environment.

This paper presents an acoustic model-based lossy pole-zero modeling for speech signals, which overcomes the limitation in the existing lossless pole-zero model that forced the numerator part of the pole-zero transfer function to be symmetric. We derive the lossy pole-zero model and its transfer function by employing the wave digital filter (WDF) adaptor formulas and by converting the fixed termination value -1 to a loss factor /spl mu//sub 0//sup c//spl isin/(-1, 1). Then we discuss how to determine the reflection coefficients of the lossy pole-zero model. For this we first employ a well-performing ARMA modeling algorithm for a pole-zero type estimation of the given speech signal and then fit the transfer function of the lossy pole-zero model to that of the ARMA model under Euclidean cost function. This procedure is demonstrated by an example using the Steiglitz-McBride ARMA estimation method with a synthetic speech signal. The lossy pole-zero modeling yields a new filter structure-namely, three-branch lattice structure-which consists of three lattice branches with the third branch terminated by the loss factor /spl mu//sub 0//sup c//spl isin/[-1, 1] and with the three branches connected by a three-port wave adaptor characterized by the area ratio /spl sigma//spl isin/ [0,1]. The three-branch lattice structure is a general filter structure which becomes the lossless pole-zero structure when /spl mu//sub 0//sup c/=-1 and becomes the existing all-pole lattice structure when /spl sigma/=0.

As a unified approach to the description of various shift register generators, the concept of sequence space is introduced and its properties are examined. A sequence space refers to a vector space whose elements are sequences satisfying the relation specified by a characteristic polynomial. In support of the sequence space, two bases-the elementary basis and the primary basis-are defined, and the polynomial expression of the sequence is defined as a tool for mathematical manipulations within the sequence space. Based on these definitions, various properties of sequence spaces such as sequence subspaces and minimal sequence spaces are investigated and summarized in terms of properties and theorems. The developed sequence theory is then applied to the description of the behaviors of shift register generators (SRGs). An SRG is represented by the state transition matrix, and the relevant SRG sequences are uniquely determined by this state transition matrix and the initial state vector. For an SRG, it is shown how to identify the sequence space generated by the SRG sequences with a fixed initial state vector (or the SRG space), and further, how to find the largest-dimensional sequence space that can be obtained by varying the initial state vectors (or the SRG maximal space). Conversely, for a given sequence space, it is shown how to find the minimum-sized SRGs that can generate the sequence space (or the basic SRGs). Finally, it is shown that the two typical SRGs-simple SRG and modular SRG-are special cases of basic SRGs that can generate the primary and the elementary bases, respectively.

This paper discusses recent advances in the theory and applications of scrambling techniques for digital lightwave transmission. It introduces the theories of sequence space and shift register generator (SRG) space which enable systematic analysis and mathematical manipulation of the behavior of sequences in general and the related SRG's. It discusses the behavior and realization of frame synchronous scrambling (FSS) and distributed sample scrambling (DSS) with emphasis on parallel sequences and the related parallel SRG's (PSRG). In addition, it describes self synchronous scrambling (SSS). Then the paper applies the theories to today's lightwave transmission systems by demonstrating practical parallel designs of FSS for SDH/SONET transmission, DSS for cell-based ATM transmission, and SSS for SDH-based ATM transmission. It finally considers how DSS can be used for scrambling of mixed isochronous and nonisochronous data in future high-speed data networks. The paper employs various new concepts and terminology, such as PSRG engine, generating vector discrimination matrix, (M,N) PSRG, sampling vector, correction vector, correction matrix, predictable scrambling concurrent sampling, and immediate correction.>

A theory is developed on parallel shift register generators (PSRG) and its parallel scrambling applications are discussed in the multibit-interleaved multiplexing environments. The PSRG is a shift register generator (SRG) which generates multiple sequences in such a way that when these multiple sequences are multibit-interleaved the resulting sequence becomes identical to some prespecified sequence. The properties of PSRGs are discussed and efficient realizations of PSRGs are examined with stress put on a special class of PSRGs for which the generating polynomials are irreducible. Finally demonstrations are given on the application of the PSRG theory to parallel scrambling of the SDH-based lightwave transmission signals STM-4 and STM-16.>

Studies the traffic shaping issues for VBR video services in ATM networks with stress put on the source clock frequency recovery aspect. The authors propose a "three-level traffic shaper" (TLTS) which is able to limit the minimum as well as the maximum cell rate of input VBR traffic. For a VBR video source modeled after a two-state Markov modulated Poisson process, they determine, as performance measures, the cell loss probability, the underflow probability, and the maximum delay that a cell can experience in the proposed TLTS. Then they apply the TLTS for source clock frequency recovery in VBR video services. In extending the existing CBR-based synchronous residual time stamp (SRTS) method for VBR video services, it is necessary to restrict the minimum, as well as the maximum, number of cells generated in a fixed period of time to ensure the transmission of timing information, and the proposed TLTS is tailored to meet this necessity by adopting an optimal dimensioning procedure based on the genetic algorithm (GA). Finally the authors provide some numerical examples to demonstrate the optimal dimensioning procedure of TLTS and the related performances of statistical multiplexing and source clock frequency recovery.>

The paper presents a new distribution network which is capable of concentrating and shifting the incoming active packets simultaneously, without requiring dummy destination address generation and extraction processes. It has the structure of a reverse banyan network (RBN) and consists of controlled switching elements (CSEs) which is obtained by extending the passive iterative-cells introduced by Narasimha [1994]. The CSE-based RBN has a set of external control inputs (ECIs) in addition to the data input and output lines and can generate different output patterns according to the ECI values. It is shown through four properties that the CSE-based RBN can perform the distribution function of the conventional distributor. In addition, it is rigorously described in the properties how to determine the set of ECI values to achieve the desired distribution function, which includes the distribution in the normal mode, in the reversed mode, and in alternation of these two modes. The proposed CSE-based distributor can be applied to a variety of occasions by modifying the use of the counter, the numbers to write on the registers, and the table to store the ECI values. Some of useful examples are demonstrated through applications to shift-sequence permutation, N/spl times/R concentration, nonblocking point-to-point switching, and virtual FIFO queueing.>

In this paper we present a new digital filter structure which enables a practical realization of finite-wordlength low-sensitivity digital filters. It is a cascade structure consisting of Type I and Type 2 LBR two-pairs interlacing each other, and is called a two-type-interlaced structure. In addition, an LBR test is introduced as a means to determine the LBRness of the quantized two-type-interlaced structure. The proposed structure and the LBR test are applied to the conventional digital filters. The results show that perfect LBRness can always be maintained with the proposed structure.>

In this paper a theory is developed to support the synchronization of shift register generators (SRG) in the distributed sample scramblers (DSS). DSSs, recently introduced for the cell-based ATM scrambling, are identical to the frame synchronous scramblers (FSS) in scrambling and descrambling processes, but are different in synchronizing the descrambler SRG states. The paper provides a systematic solution to the DSS synchronization problem based on mathematical modeling. It first considers how to sample the SRG state information of scrambler for transmission. Then it discusses how to use the received SRG state samples for the synchronization of the SRG state of descrambler. Examples are attached at the end to demonstrate the developed theory in the cell-based ATM scrambling environment.>

The authors establish a theory for lossless pole-zero modeling of speech signals for the description of nasal sounds. The theory is based on a generalized vocal tract tube model which consists of the main vocal tract, the oral tract, and the nasal tract. A pole-zero type transfer function, which turns out to be a generalized version of the existing all-pole type transfer function, is derived. Fundamental properties of the generalized vocal tract model are investigated, employing the concept of discrete-time reactance. A procedure for evaluating the reflection coefficients for the model is outlined. The assumption of losslessness in the modeling leads to the following two properties. First, the combination of two lattice structures representing the oral and the nasal tracts form one larger lattice structure when viewed at their joint point called the branch boundary. Second, the oral and the nasal tract render respective discrete-time reactances whose convex combination generates the discrete-time reactance at the branch boundary. The first property enables the combined reactance to be computed, and the second helps separate it into its components.>

A theory to support signal alignment in rolling-based lightwave transmission systems is developed. The theory originates from the transmission system environment, which consists of three basic building blocks-multiplexer, scrambler, and permuter. These building blocks are represented by mathematical operators, on the basis of which mathematical expressions of the whole transmission system become possible. The mathematical operators are directly translated into number operators which help detect and identify the received signals in the form of a table-the signal-detection table. The signal-detection tables play a central role in the signal-alignment theory, since any transmission system can be represented by its corresponding signal-detection table. The entries in the signal-detection tables are closely correlated so that the tables can be uniquely characterized by their corresponding characteristic expressions. Various types of signal-detection tables are examined, and solutions to align transmitted signals and to reduce signal-alignment time in rolling-based lightwave transmission systems are exemplified through scrambler-permuter configurations.>

The authors consider a generalized acoustic tube model of the vocal tract, relating it to the pole-zero type linear prediction. The generalization is done by including the nasal cavity for the modeling, thus forming a three-branched model. The transfer function is obtained from the generalized model by conglomerating one of the three branches to the branch section at the junction of the three branches. It is also discussed how find coefficients for the pole-zero type linear prediction from the voiced sounds. Also discussed is how to evaluate the reflection coefficients by connecting the pole-zero type linear prediction algorithm to the transfer function of the generalized model.>

A new discrete cosine transform algorithm, named Fast Cosine Transform (FCT), is introduced for the 2m-point discrete cosine transform. This algorithm reduces the number of multiplications to about half of earlier results, and furthermore, it renders a simple and systematic structure for implementation.