In this paper, we develop a framework for an innovative perceptive mobile (i.e. cellular) network that integrates sensing with communication, and supports new applications widely in transportation, surveillance and environmental sensing. Three types of sensing methods implemented in the base-stations are proposed, using either uplink or downlink multiuser communication signals. The required changes to system hardware and major technical challenges are briefly discussed. We also demonstrate the feasibility of estimating sensing parameters via developing a compressive sensing based scheme and providing simulation results to validate its effectiveness.

Beam-steering has great potentials for joint communications and sensing, which is becoming a demanding feature on many emerging platforms such as unmanned aerial vehicles and smart cars. Although beam-steering has been extensively studied for communications and radar sensing respectively, its application in the joint system is not straightforward due to different beamforming requirements by communications and sensing. In this paper, we propose a low-cost system framework which allows seamless operation of communications and sensing, using two small- size steerable analog antenna arrays. We provide system architecture, high-level protocols, detailed signal model, novel beamforming design and advanced 1D compressive sensing algorithms for joint communications and sensing. We also provide preliminary simulation results which validate the effectiveness of the proposed technique in resolving closely located objects.

Beamforming design for millimeter-Wave hybrid array with the subarray structure is very challenging. There is neither known optimal solution that maximizes the sum rate capacity nor near-optimal solution. This paper proposes some low-complexity user-directed analog radio- frequency (RF) beamforming design schemes. The basic idea is to iteratively allocate different subarrays to different users such that users' channel correlation can be efficiently reduced via RF beamforming. Several new but less efficient schemes are also presented to shed light on RF beamforming design, and to serve as comparisons for the user-directed schemes. Simulation results are provided for these proposed schemes, existing ones in the literature and an upper-bound for hybird array with a fully-connected structure. The user-directed schemes demonstrate significantly better sum-rate and BER performance over other schemes, although the gap to the upperbound is still large.

Zero-forcing beamforming (ZFBF) is a popular pre-coding scheme for MIMO systems. Most of the studies in the literature are under total power constraints. However, the per-antenna power constraints (PAPC) are more realistic. The state-of-the-art method is interior point method which is expensive to realize in practice due to the high computational complexity. Hence, a low complexity zero-forcing precoding scheme under the per-antenna power constraints is proposed in this paper. This is achieved by introducing a regularized dual method. Simulations are carried out to show the effectiveness of the proposed method, which has a low computational complexity. In addition, the algorithm can be implemented in parallel to further reduce the computational complexity.

This letter proposes a new algorithm for solving the optimal zero-forcing beamforming problem that maximizes the user achievable rate with restriction on the per-antenna element power constraints. An accelerated gradient method with step size search is proposed for solving the problem. For each iteration of the gradient approach, a quick one dimensional search is employed to obtain the step size. The advantage of the step-size search is that it is relatively fast and requires only a few calculations of the objective function. Design examples show that the proposed algorithm converges faster than the gradient approach and the accelerated gradient approach with a constant step size while achieving a low computational complexity.

Link aggregation, or multi-band striping, has been used in wireless backhaul systems to overcome the limitations of bandwidth and transmission range. Work-conserving and in-sequence delivery are essential but conflicting requirements in multi-band striping systems. Previous schemes can achieve one or the other, but not both. The heterogeneous channel and time varying data rate in multi-band wireless backhaul systems present additional challenges. We aim at designing a multi-band striping algorithm to achieve both work-conserving and in-sequence delivery in the heterogeneous time-varying multi-band wireless backhaul system. We propose a parallel processing architecture for multi-band aggregation, and derive the necessary conditions for such system to achieve sequence preserving and work conserving. An optimum set of timing controls is developed and proved to provide the best performance. We design a work-conserving in-sequence striping algorithm and prove it to be sequence preserving and work conserving. The algorithm is then extended to practical data granularity. Performance analysis and simulation results demonstrate that our designs are able to preserve data sequence, reduce delay, and achieve 100% channel utilization.

A low-complexity and efficient sidelobe suppression with orthogonal projection (SSOP) scheme is proposed in [1] for OFDM systems. This paper provides comprehensive performance analysis for the zero-forcing receiver for the SSOP scheme. Via rigorous proof, we show the independence of the orthogonal projection matrix on the ordering of the suppression distances, and the monotonicity of the SNR with the suppression distance and the number of reserved subcarriers. We also characterized the SNR degradation of the single-side and double-side suppression schemes analytically. These analytical results match with the numerical results well.

Sidelobe suppression, or out-of-band emission reduction, in multicarrier systems is conventionally achieved via time-domain windowing which is spectrum inefficient. Although some sidelobe cancellation and signal predistortion techniques have been proposed for spectrum shaping, they are generally not well balanced between complexity and suppression performance. In this paper, an efficient and low-complexity sidelobe suppression with orthogonal projection (SSOP) scheme is proposed. The SSOP scheme uses an orthogonal projection matrix for sidelobe suppression, and adopts as few as one reserved subcarrier for recovering the distorted signal in the receiver. Unlike most known approaches, the SSOP scheme requires multiplications as few as the number of subcarriers in the band, and enables straightforward selection of parameters. Analytical and simulation results show that more than 50dB sidelobe suppression can be readily achieved with only a slight degradation in receiver performance.

To enable ubiquitous end-to-end quality of service over IP networks expanding to rural areas, we examine resource allocation problems of a MAC layer for an OFDM MIMO wireless base station (BS) using multiuser beamforming. We compare two linear beamforming techniques, zero-forcing (ZF-BF) and minimum mean square error (MMSE-BF). To guarantee minimum bandwidth and low packet delay in an environment where the number of users is much larger than the number of BS antennas, one needs to partition the users into several sets and combine space division multiple access (SDMA) with time division multiple access (TDMA). We study the impact on the optimal transmission power resulting from selecting the following design parameters: (i) ZF-BF vs. MMSE-BF; (ii) the number of user sets multiplexed by TDMA; and (iii) the number of BS antennas. Two notable results are observed: (1) power wise, ZF-BF is far more superior to MMSE-ZF; (2) substantial power can be saved by increasing the number of BS antennas; however, only by large increments, e.g., from 6 to 9 and further to 21.

Modern telecommunications networks maintain synchronization and distribute accurate timing information using approaches with strict requirements on clock quality and jitter and wander at interfaces. In the future, packet networks may be used for timing distribution and synchronization e.g. using IEEE 1588v2 specifying the Precision Time Protocol (PTP). Without amelioration, there may be effects on data buffers within telecommunications networks which have been designed with particular assumptions of clock quality. Applications such as digital audio and video also have synchronization requirements and limits on jitter and wander tolerance. Attention is needed on the impact of clock quality due to clock synchronization on the synchronization and jitter and wander limits for data services over networks. A well-understood approach to the transfer of timing for a data service over a packet network is the Synchronous Residual Time Stamp (SRTS) method. We previously presented a novel approach to modelling and visualising jitter spectra in SRTS timing transfer due to jitter in the reference clocks. In this paper, we present experimental results, which validate the appropriateness of our approximations. Experimental results also show the existence of some spectral components not predicted from the modelling. In the paper we discuss the origins of these spectral components.

This letter presents a new computational procedure for the second-order gradient-based blind signal separation (BSS) problem with convolutive mixtures that has improved convergence characteristics over the steepest descent algorithm. The BSS problem is formulated as a constrained optimization problem with complex unmixing weight matrices where the constraints are formulated to overcome the permutation effects. This problem is then transformed into an unconstrained optimization problem, so that the conjugate gradient algorithm can be applied. The convergence of the proposed procedure is compared with the steepest descent algorithms in real and simulated environments.

In a direct-conversion transmitter system, the non-ideal transfer characteristics of analog subsystems in the I and Q channels adversely affect the performance of the system. The study of static errors, including gain imbalance, phase errors, and DC offsets has received considerable attention in the literature. Much less attention has been paid to the effect of frequency-dependent imperfections that arise in the analog reconstruction filters commonly employed in direct-conversion structures. This paper presents results of research into robust methods for automatic digital compensation for these non ideal frequency-dependent characteristics. The design and implementation of FIR and Eigenfilter structures to achieve compensation are presented. As a specific application the paper considers a direct-conversion system used to generate continuous-phase frequency-shift keying (CPFSK) signals. The performance achievable using the proposed digital compensation structures is presented, including measurement results taken on a DSP-based modulator platform.

In constant bit-rate timing transfer, the reference clocks which encode and reconstruct the service clock at origin and destination may be jittered. We present new, straightforward approaches to finding and visualizing jitter spectra in timing transfer for jittered destination and reference clocks, and confirm our results by simulation

The probability hypothesis density (PHD) recursion is a first moment approximation to the multi-target Bayes filter which propagates the posterior intensity of the random finite set of targets in time. The cardinalized PHD (CPHD) recursion is a generalization of the PHD recursion, which jointly propagates the posterior intensity and the posterior cardinality distribution. Using the recently developed closed form solutions to both the PHD and CPHD recursions for linear Gaussian multi-target models, we present a comparative study of their performances

We characterize the metastability measurement system (Dike, 1999) in which asynchronous data input and sampling clock frequencies trigger metastability. We develop the equation describing the time interval between data and clock inputs for practical frequencies and show that it takes on discrete values in the absence of jitter and that the presence of jitter perturbs these discrete values. Finally, we present experimental results supporting our characterization

In a direct-conversion transmitter system, the nonideal transfer characteristics of practical analog subsystems can have significant adverse effects on the performance of the transmitter system. While a significant body of research exists on correcting static errors (including gain imbalance, phase errors, and DC offsets) in the analog subsystems, relatively little attention has been paid to the impact of nonideal frequency-dependent characteristics in the analog reconstruction filters that are employed in direct-conversion structures. This paper presents a new computationally efficient and numerically robust method for automatic digital compensation for nonideal frequency-dependent characteristics in the signal reconstruction filters of a quadrature (IQ) modulator structure. The new compensation technique is used in conjunction with existing compensation approaches that compensate for static errors.

The letter provides an analysis of optimal finite-impulse response subband beamforming for speech enhancement in multipath environments. A modification of the direct-path standard Wiener formulation is shown to give nearly optimal performance when it comes to signal-to-noise-plus-interference ratio, by including coherent multipath propagation in the criterion.

The synchronous residual time stamp (SRTS) is widely implemented for the transport of the service clock associated with continuous bit-rate services in asynchronous transfer mode adaptation layer 1. We present a new approach to service clock regeneration, which is simple to implement and can be proven to be correct.

A flexible approach to the implementation of a radio transmitter is to synthesise base-band I and Q signals using a DSP (Digital Signal Processor) followed by a vector modulator to up-convert the base-band signals directly to RF. This approach can be limited, however, by analogue reconstruction filters that are necessary to attenuate digital image components in the base-band signal spectrum before transmission. Recently, methods have been proposed to digitally compensate for these shortcomings. While these methods have been shown to be effective in improving modulator performance they require special attention to numerical issues and they typically result in large filter structures that are computationally demanding to implement. In this paper we present the results of an investigation into reducing the computational load of a digital compensation method on the DSP. The investigation has resulted in a technique that not only significantly reduces the computational burden but also improves the numerical robustness of the solution.

In this paper, the optimum (in the minimum MSE (MMSE) sense) transfer functions of the receive filter pair in a carrierless AM/PM (CAP) transceiver system are derived under the conditions that the input noise has an arbitrary spectrum shape and the baud-rate sampler has a fixed but unknown sampling phase. An expression for the corresponding MMSE at the slicer is also given. Although similar studies have been reported in the open literature, the analysis results were not consistent with the corresponding simulation results. In this work, we show that our theoretical results are in agreement with the simulation results.

A new type of adaptive scheme was proposed recently for designing a deterministic envelope-constrained (EC) filter such that the generated sequence of filters converges to the optimum filter. Previous results at this level of generality linked convergence only to within a neighborhood of the optimum filter. Based on the adaptive scheme, two new theorems are established in a stochastic environment for which the adaptive EC filter converges in mean square sense and with probability one to the noiseless optimum filter for a fixed step-size and a decreasing sequence of step-sizes, respectively. Numerical examples involving pulse compression Barker-coded signal are studied for solving the EC filtering problem.

A fully digital and simple to implement frequency detector for use with high frequency signals is presented. The new frequency detector can detect small phase movements of much less than one cycle, even though the frequency detector is clocked at a low rate. Existing digital techniques can only detect phase movements of more than one cycle, or require clock rates much higher than the frequency of the signal. The operating limits of the new frequency detector are derived. Examples of the application of the new frequency detector are also given.

Carrierless amplitude and phase (CAP) is one of the modulation schemes that has been proposed to accomplish the task of transporting high bit-rate data over cheap copper wire pairs. In this paper, we reexamined some CAP receiver structures that have been proposed in the literature and showed that the modified quadrature amplitude modulation receiver can offer a significant advantage over the other receiver structures through its greater immunity to timing phase errors. Our recommendation is based on an analysis of the eye diagram of a received CAP signal.

The envelope-constrained filtering problem is concerned with the design of a filter such that the noise enhancement is minimized while the noiseless filter response stays within an envelope. Naturally, the optimum filter response to the prescribed input signal tends to touch the output boundaries at some points. Consequently, any disturbance to the prescribed input signal could result in the output constraints being violated. In this paper, we formulate a semi-infinite constrained optimization problem in which the margin of the constraint robustness of the filter is maximized. Using a smoothing technique, it is shown that the solution of the optimization problem can be obtained by solving a sequence of strictly convex optimization problems with integral cost.

In digital IQ modulators generating continuous phase frequency shift keying (CPFSK) signals, departures from flat-magnitude, linear phase in the pass bands of signal reconstruction filters in the I and Q channels cause ripple in the output signal envelope. Amplitude modulation (AM) in the signal envelope function produces undesirable sidelobes in the FSK signal spectrum when the signal passes through nonlinear elements in the transmission path. A structure is developed for digitally pre-compensating for the magnitude and phase characteristics of signal reconstruction filters. Optimum digital pre-compensation filters are found using least squares (LS) techniques and we propose a method by which the optimum pre-compensation filters can be estimated using test input signals. This method can be used as part of an automatic compensation process.

We present a technique for digitally precompensating for shortcomings in signal reconstruction filters in an IQ modulation system. Optimum digital precompensation filters are found using least squares (LS) techniques. The optimization problem is formulated in the frequency-domain, and the optimum LS solution is characterized using an eigenanalysis approach.

The synchronous residual time stamp (SRTS) is one approach approved for the encoding and transporting of the continuous bit rate (CBR) service clock in ATM Adaptation Layer 1 (AAL1) allowing CBR services to be transported in ATM cells over the B ISDN. It has been shown by the authors and others that the SRTS method generates waiting time jitter analogous to that produced by other synchronization processes such as pulse stuffing synchronization. Modeling of the synchronization process as it applies to the SRTS method requires a time domain approach to produce an exact expression of the jitter. In this paper, we apply a new time domain analysis technique previously developed by the authors to derive the expressions for the jitter spectrum of the synchronization process in the presence of input jitter on the service clock. Furthermore, the particular form taken by the jitter spectrum when the input jitter is sinusoidal is also found. Experiments verifying the synchronization process jitter spectrum, both with and without sinusoidal input jitter, are reported. Confirmation is also provided that it is possible to approximate timing jitter by phase jitter as long as certain frequency-amplitude limits are observed.

We consider a class of FIR filters defined by the first order difference routing digital filter (DRDF) structure and sums of two powers-of-two coefficients. A novel design method is developed for constructing high quality filters with reference to the min-max error criterion. This method is highly efficient in terms of computational time. Simulation studies show a large improvement over existing methods such as quantization. In some cases, the peak ripple magnitude over the stop and pass bands is reduced by up to 13 dB over the quantization method. These results are achieved even for cases involving small number of delays.

In an envelope-constrained filtering problem with uncertain input the set of possible inputs and the set of permissible outputs are each defined by envelopes or masks. This paper considers a continuous-time filter which in structure is comprised of an A/D converter, an FIR filter, a D/A converter and an analog post-filter. The object is to design the digital component of the filter structure so as to minimize the noise enhancement whilst satisfying the constraint that every signal in the input envelope evokes a response which stays in the output envelope.

Clock-rate adaptation circuits of network synchronizers work by controlling the output data rate by pulse stuffing. The technique of stuff threshold modulation (ST-Mod) is one method used to reduce the waiting time jitter produced by pulse stuffing. An expression for the filtered waiting time jitter in the presence of an ST-Mod sequence is present and the selection of a suitable ST-Mod dither sequence for a given set of data transmission conditions is investigated. It is shown that by using either a random ST-Mod sequence or a modified-wave ST-Mod sequence, it is possible to obtain minimal filtered jitter for given synchronizer-desynchronizer characteristics.

The synchronous residual time stamp (SRTS) method has been recommended by the ITU-T as one approach to the transfer of timing information for continuous bit rate (CBR) services being carried by the asynchronous transfer mode (ATM) adaptation layer 1 (AAL1) of the broadband integrated services digital network (B-ISDN). We give an overview of the SRTS method including a description of service clock recovery at the destination. On the basis of analytical expressions derived, we show that even in an ideal SRTS system, where all clocks are jitter-free, the SRTS method inherently generates jitter. We also show that this jitter is equivalent to the so-called waiting time jitter that is found in conventional timing justification techniques. In other systems where waiting time jitter is generated, parameters are chosen in order to minimize the amount of waiting time jitter produced and in which part of the spectrum it is located. We show that for the SRTS method, the choice of parameters to minimize waiting time jitter is quite restricted. For higher service clock frequencies and clocks with large tolerances, it may be impossible to choose parameters to influence the characteristics of the waiting time jitter produced.

Consider a continuous-time filter the structure of which is comprised of an A/D converter, an FIR filter, a D/A converter and an analog post-filter. The envelope constrained (EC) filtering problem for this filter structure is to design the digital component so as to minimize the effect of the input noise whilst satisfying the constraint that the noiseless response of the filter to a specified excitation fits into a prescribed envelope. This problem is formulated as a quadratic programming (QP) problem with functional inequality constraints. Approximating this continuum of constraints by a finite set, the problem is solved by QP via an active set strategy.

Envelope-constrained filtering is concerned with the design of a time-invariant filter to process a given input signal such that the noiseless output of the filter is guaranteed to lie within a prespecified output mask. In this paper, using Laguerre filters and H/sub /spl infin// optimization techniques, the continuous-time envelope-constrained filter design problem has been reformulated and solved as a constrained H/sub /spl infin// model-matching problem. To illustrate the effectiveness of the design method, a numerical example is presented which deals with the design of an equalization filter for a digital transmission channel.

Narrow bandwidth phase-locked loops (PLLs) can have difficulty acquiring lock reliably when there is a significant difference between the input signal and the free run frequency of the PLL's voltage-controlled oscillator (VCO). The new technique presented here incorporates an accurate local reference frequency into the PLL structure. The range of frequencies to which the new PLL structure can lock can be confined to a desired small region around the accurate local reference frequency. The new PLL structure also provides other benefits such as reduction of VCO phase noise. The new technique does not require any monitoring nor any switching of the local frequency reference signal which is always acting. The key parameters of the new PLL structure are identified and the performance characterized.

This paper investigates two techniques for extending the narrowband (NB) eigenvector method for towed array shape estimation to process broadband sources. These techniques are the weighted incoherent broadband processor (WIB) and the weighted quasi-coherent broadband processor (WQB). It is demonstrated, both analytically and via computer simulations, that the two broadband processors can dramatically improve the bias and variance of the receiver position estimates, compared to the NB technique. The results of computer simulations also demonstrate that the WQB method has a lower SNR threshold than the NB and incoherent methods.

The envelope constrained (EC) filtering problem is concerned with the design of a filter with minimum gain to white input noise and with a response to a given signal that fits into a prescribed envelope. Using the discrete-time orthonormal Laguerre network, the EC filtering problem is posed as a quadratic programming (QP) problem with affine inequality constraints. An iterative algorithm for solving this QP problem is proposed which serves as a basis for the derivation of adaptive algorithms for applications where the parameters of the underlying signal model are either not known or varying with time.

A new synchronizer design is presented. Current synchronizer designs have certain disadvantages, both in characterization and in the tradeoff between settling time and sampling rate, which are overcome in the new design. Two possible implementations of the synchronizer are discussed.

The jitter generated by two different approaches to synchronization of an external timing source is analyzed. The two approaches are: the use of conventional stuffing and the use of a simpler frame sampling synchronizer. It is shown that these two approaches have the same jitter power density spectrum. The result is obtained using a new time domain method developed in the paper. Exact expressions for the jitter on the synchronized external timing source are obtained. Furthermore, the effect of quantization in the conventional stuffing synchronizer on the jitter expressions obtained is also explored. Jitter generated by the frame sampling synchronizer is also analyzed using the time domain method, which again produces an exact expression. In the case where certain parameters of the two approaches are related, the location of spectral lines in their respective jitter spectra is shown to be identical. It is also shown that both approaches to synchronization introduce the so-called waiting-time jitter.

The eigenvector method for estimating the positions of the receivers of an ocean-towed array is based on the eigendecomposition of the array signal correlation matrix to find the phase delays between the array receivers. Previous work has shown that for reasonable SNR, the bias and variance of the phase estimates is relatively independent of the number of receivers in the array. This suggests that the computational cost of the eigenvector method could be substantially improved by partitioning the array receivers into groups of smaller sub-arrays and then applying the eigenvector method to each sub-array. This paper introduces the interleaved partitioned eigenvector method, and shows that it significantly reduces the computational cost without adversely affecting the quality of the position estimates. Numerical examples substantiate the theoretical work of this paper and also demonstrate the improvement in beamforming when employing the shape estimation algorithm.

The discrete-time envelope constrained (EC) filtering problem can be formulated as a quadratic programming (QP) problem with linear inequality constraints. In this paper, the QP problem is approximated by an unconstrained minimization problem with two parameters. These parameters can be selected so that given an acceptable deviation from the norm of the optimal EC filter, the solution to the unconstrained problem satisfies both the deviation and envelope constraints. Newton's method with line search is applied to solve the unconstrained problem iteratively.

In ATM networks, one effective way to provide service guarantees is to shape individual traffic streams at various points in the network. In the paper, the authors present a method for multiplexing a large number of single-stream spacer outputs on cell emissions rather than on cell arrivals. The method not only approximates first come first served (FCFS) multiplexing of such outputs but also better suits bursty arrivals. A number of performance related issues of the proposed method are identified and examined.

This paper presents new recursive procedures for designing optimum envelope-constrained (EC) filters. Using a constraint transcription technique, the inequality constrained quadratic programming problem associated with EC filtering can be approximated as an unconstrained minimization problem. Two types of optimization methods are developed to solve this unconstrained problem in a recursive adjusting manner. Simulations of the proposed recursive procedures are presented which support the theoretical predictions.

The eigenvector method for estimating the positions of the receivers of an ocean-towed array is based on the eigendecomposition of the array signal correlation matrix to find the phase delays between the array receivers. This paper derives the mean, variance and covariance of the phase delays when the correlation matrix is estimated from receiver data. It is shown that the variance of the relative phase delays is inversely proportional to the number of snapshots and SNR but relatively independent of the size of the correlation matrix. The theoretical derivations are supported by numerical simulations.>

A method frequently employed to optimise the performance of a broadband antenna array processor is to find the set of processor weights which minimises its mean output power subject to linear constraints on the processor weights. These linear constraints determine the frequency response of the array processor in the look direction. To obtain robustness against mismatch between the array look direction and the actual signal direction, derivative constraints can be further imposed. However, derivative constraints corresponding to necessary and sufficient (NS) conditions for a maximally flat spatial power response in the look direction are in general quadratic. This limits their value for real-time processing. This paper presents several simpler derivative constraints. These constraints correspond only to sufficient conditions for maximal flatness but perform almost as well as the NS constraints and are better suited to real-time implementation.>

The problem of envelope-constrained filters with uncertain input (ECUI) was first formulated as a nonsmooth optimization problem. Previously, it was shown that this nonsmooth problem is equivalent to a standard quadratic programming problem which can be solved efficiently. The objective of the present paper is to investigate several issues relating to the sensitivity of ECUI filters. They are: (i) the output mask tolerance; (ii) the effect of the location of the pulse peak in the output response on the output mask tolerance; and (iii) the effect of the order of ECUI filters on the output mask tolerance. Theoretical analysis is supported by numerical simulation studies.>

This paper proposes a new design for a frame sampling synchronizer and gives some typical applications for the device. It also discusses metastability which arises in the system due to the existence of asynchronous inputs. The motivation for the new design, details of the design and its modelling are also covered. Finally the jitter generated by the synchronizer is analysed.>

This paper is concerned with the maximum robustness design of envelope-constrained filters with uncertain input (ECUI). It is shown that the maximum robustness ECUI filter design can be formulated as a minimax optimization problem in which the set of feasible filters is characterized by nonsmooth matrix inequalities. On the basis of the use of a newly developed transformation technique and the use of standard linear programming and quadratic programming, an efficient design algorithm is developed to solve this problem. The attractive feature of the proposed algorithm is that the ECUI filter thus designed can allow for the greatest uncertainty in the input signal while still achieving the acceptable filter performance. A numerical example is given to illustrate the effectiveness of the algorithm.>

The authors examine how to provide constant bit rate (CBR) service over a channel that can be characterized by a constant transmission delay plus a variable cell access delay. To compensate for the variability in the access delay, buffers are used at the source and the destination ends of the connection. Whenever buffers overflow or underflow the continuity of data flow over the connection breaks down. The minimum size of buffers required at the transmitter and the receiver to provide the connection is determined. The minimum source to sink delay achievable for different buffer initialization strategies and the corresponding buffer prefills are derived.>

A method for supporting time-constrained and nontime-constrained integrated services on a shared dual bus network is presented. The method is an extension of the basic distributed queue protocol defined in the IEEE 802.6 distributed-queue dual-bus (DQDB) MAN standard. The extended distributed queueing protocol differs from the basic protocol in allowing more than one outstanding request in the distributed queue at any one time. Used at the high priority and in conjunction with segment stream shaping and subscribed service limiting, the extended protocol provides bearer services with guaranteed bandwidth and bounded delays. Bearer services with fair but nonguaranteed bandwidth shares, suitable for nontime-constrained applications, are provided by the same protocol and bandwidth balancing operating at the lowest priority. A number of the desirable attributes of the proposed scheme are identified including the optimality of the delay variance introduced by the access protocol. An approximate analysis of the access delay experience by time-constrained streams is presented.>

Issues arising in the implementation of receivers for asynchronous transfer mode (ATM) networks, e.g., IEEE Std 802.6 MANs (metropolitan area networks) and broadband integrated services digital networks (B-ISDNs), are examined. These include buffering strategies to account for the burstiness and spatial randomness of buffering demands, preventing reassembly deadlock, and enforcing packet reception precedences. It is shown that transnode delays critically affect protocol performance and the complexity of implementing spatial reuse.>

Fault-free digital systems can fail as a result of metastable behavior when asynchronous inputs have critical timing combinations. The problem of metastable behavior is generally considered to be unavoidable in digital systems that synchronize asynchronous inputs. This correspondence extends previous results on the unavoidability of metastable behavior. The set of inputs to the digital system is generalized to cover a wide range of possibilities encountered in practical circuits whilst still maintaining the result of unavoidability of metastable behavior. A sufficient condition for the set of inputs to contain a range of inputs that excite metastable behavior is that the set of input functions is connected and contains two input functions which drive the system to different stable states.

A new approach to the design of narrowband antenna array processors is proposed. The new approach allows a wide variety of possible errors to be incorporated in the problem formulation and leads to a robust optimum weight vector. The new approach can be used to make the optimum system robust against channel phase errors, array geometry errors and pointing errors, to name a few. Initially a general quadratic constraint on the weights is developed. However, it is then shown that the quadratic constraint can be replaced by linear constraints or at most linear constraints plus norm constraint. These latter set of constraints are no more complex than those required for designs which do not incorporate robustness features explicitly. The new optimum processor can be implemented adaptively.

A class of arbiters, known as batching arbiters, is introduced and defined. A particularly simple decentralised example of a batching arbiter is described, with motivation given for the batching arbiter model adopted. It is shown that under reasonable assumptions, batching arbiters can be described by a finite state Markov chain. The key steps in the analysis of the arbiter performance are the method of assigning states, evaluation of state transition probabilities and showing that the Markov chain is irreducible. Arbiter performance parameters are defined, such as proportion of time allocated to each requester and mean waiting time for each requester. Apart from results describing the steady state behavior of the arbiter for general system parameters, a number of limiting results are also obtained corresponding to light and heavy request loading.

Abstract A simple technique that improves a system's capability of error detection and correction, and enhances software debugging is described in this note. The technique relies on consistency checking of branching variables during branching decisions and the appropriate selection of codes for the permissible values of variables.

This paper considers the possibility of achieving improvements in the reliability of synchronizing an asynchronous signal, by exploiting redundancy and masking. Redundancy and masking techniques have been applied successfully to mask both permanent and transient hardware faults. However, it is shown in this paper that redundancy and masking techniques are ineffective against synchronization failures which arise because of metastable behavior of synchronizing elements.

This paper presents a new set of linear constraints for designing broadband time domain element space processor which can handle a variety of steering situations, namely, no pre-steering, coarse/ quantized pre-steering and exact pre-steering. The relationship that the new processor has to other broadband processors is established. Furthermore, the approach presented enables the processors to be made robust against directional errors. Analytical as well as simulation results on the new processors are presented.

Adaptive antenna array processing employing a constrained least mean square algorithm requires an unbiased estimate of the gradient of the output power with respect to the array weights. There are a number of schemes for obtaining an unbiased estimate of this gradient. As the covariance of the estimated gradient with each method is different, the transient and the steady state behaviour of the algorithm is different in each case. In this paper the transient and the steady state behaviour of the weight covariance matrix is analysed and the exact expressions for the misadjustment are derived when the required gradient is estimated using perturbation sequences. The necessary and the sufficient condition for the diagonalization of the weight covariance matrix is derived.

This paper describes a new approach to the design of broadband element space antenna array processor which can handle a variety of steering situations. The approach is applicable to array processor without pre-steering, with coarse pre-steering and exact pre-steering. Furthermore, the approach presented enables mismatch between signal model and actual signal scenario to be incorporated in the problem formulation. Analytical as well as simulation results on the new class of antenna array processors are presented.

This concise paper investigates various aspects of the application of unbiased linear minimum mean-square error (ULMMSE) estimators to the equalization of channels used for digital data transmission. One application is to the equalization of the channel to a response free from intersymbol interference. Previous results regarding this application are clarified and extended. The other application is to the equalization of the channel to a finite short memory response. This stems from an interest in reducing the complexity of the Viterbi algorithm. Various new results on the optimum ULMMSE equalizer and its performance are presented. In both applications, the optimum ULMMSE equalizer is a stable, realizable, finite dimensional recursive filter.

Companion work on the design of envelope-constrained filters is extended and shown to provide an easily implementable adaptive filter with a structure quite similar to that of other adaptive filters based on least-squares techniques. Behavior of the new filter in noise is examined, and a variety of other extensions are discussed. An application to TV channel equalization is explored in some detail.

In the envelope-constrained filtering problem, one attempts to optimize a filter's impulse response subject to the constraint that its response to a specified input lies within a prescribed envelope. This constrained optimization problem is reduced to an unconstrained dual problem with a nondifferentiable cost and then solved using a steepest ascent algorithm based on directional differentials. The resulting technique provides a versatile alternative to the least-squares methods that are commonly applied to such problems. Applications to a wide variety of signal processing problems in areas such as communication channel equalization, television transmission, radar and sonar detection, filter and antenna design, seismology, and ultrasonic imaging are discussed, and some numerical results are presented.

This paper describes an interpretive function generator, COMPIL, for use with the PDP-11 FORTRAN COMPILER and Object Time System. The implementation uses a ‘mixed code’ approach, which enables the exchange of run time for program space. Other benefits include improved debugging facilities and extended interactive facilities such as the ability to define and link functions at run time. This latter feature is especially useful on the PDP-11 as compilation and linking of function subprograms take an appreciable time.

The matrices which are symmetric about both diagonals (persymmetric matrices) occur often in communication and information theory problems. There have been a number of problems whose solution involves the eigenvectors of persymmetric matrices. This paper summarizes a number of properties of the eigenvectors of persymmetric matrices and demonstrates how they can be applied to characterize and simplify the solution to a number of communication problems.

The method of inversion of linear discrete time systems utilizing quantized feedback is examined, and a stability concept is developed. The application in mind is decision feedback equalization (DFE). Sets of necessary conditions and sufficient conditions for stability are derived. A lower bound for the probability of recovery from error, and an upper bound for the mean recovery time are obtained. It is suggested that a stability index for each system could be defined. The bounds derived do not depend on the nature of the noise or the channel impulse response except through its length.

This paper considers the convergence properties and numerical behavior of an adaptive equalization algorithm. Particular attention is paid to the behavior of the algorithm when the computations are carried out with limited precision arithmetic as would be the case in a real implementation. It is shown that a previous conjecture regarding the best ordering of the step size parameters does not hold when numerical errors are significant. An alternative stable ordering is given in the paper.

An equalizer based on the minimum mean-square error (MSE) criterion, but with direct solution of the resulting equations, is described. The direct solution is based on the algorithms devised by Levinson and Trench. With the availability of large-scale integration (LSI) bipolar computing elements, these algorithms are competitive with iterative procedures. A method is considered for the estimation of the required parameters, and for automatic adaption to a changing channel. It is shown that the delay can be adjusted to achieve minimum MSE with respect to that parameter. Simulations undertaken show the robust performance of the algorithm, and that the equalizer performance is not adversely affected by operation in decision-directed mode.

A receiver structure recently proposed for the detection of pulse-amplitude modulation (PAM) signals is considered. The receiver is designed to reduce the complexity of the Viterbi decoder by equalizing the channel to an optimum desired short memory channel. Expressions for the optimum prefilter and short memory model are given. It is also shown that the minimum mean-square error achievable is a monotonic decreasing function of the model memory length. Numerical results for a specific channel are presented verifying this. A numerical investigation of the effect of shifting the reference tap is also included.

A method is described for determining an optimal straight-line segment approximation to specified functions for constrained and unconstrained endpoints.