Showing papers by "Codex Corporation published in 1988"

Queueing in high-performance packet switching

Abstract: The authors study the performance of four different approaches for providing the queuing necessary to smooth fluctuations in packet arrivals to a high-performance packet switch. They are (1) input queuing, where a separate buffer is provided at each input to the switch; (2) input smoothing, where a frame of b packets is stored at each of the input line to the switch and simultaneously launched into a switch fabric of size Nb*Nb; (3) output queuing, where packets are queued in a separate first-in first-out (FIFO) buffer located at each output of the switch; and (4) completely shared buffering, where all queuing is done at the outputs and all buffers are completely shared among all the output lines. Input queues saturate at an offered load that depends on the service policy and the number of inputs N, but is approximately 0.586 with FIFO buffers when N is large. Output queuing and completely shared buffering both achieve the optimal throughput-delay performance for any packet switch. However, compared to output queuing, completely shared buffering requires less buffer memory at the expense of an increase in switch fabric size. >

Reduced-state sequence estimation with set partitioning and decision feedback

Abstract: A reduced-state sequence estimator for linear intersymbol interference channels is described. The estimator uses a conventional Viterbi algorithm with decision feedback to search a reduced-state subset trellis that is constructed using set-partitioning principles. The complexity of maximum-likelihood sequence estimation (MLSE) due to the length of the channel memory and the size of the signal set is systematically reduced. An error probability analysis shows that a good performance/complexity tradeoff can be obtained. In particular, the results indicate that the required complexity to achieve the performance of MLSE is independent of the size of the signal set for large enough signal sets. Simulation results are provided for two partial-response systems. A simple technique for quadrature partial-response signaling (QPRS) is described that eliminates the quasicatastrophic nature of the ML trellis. >

Coset codes. II. Binary lattices and related codes

Abstract: For pt.I see ibid., vol.34, no.5, p.1123-51 (1988). The family of Barnes-Wall lattices (including D/sub 4/ and E/sub 8/) of lengths N=2/sup n/ and their principal sublattices, which are useful in constructing coset codes, are generated by iteration of a simple construction called the squaring construction. The closely related Reed-Muller codes are generated by the same construction. The principal properties of these codes and lattices are consequences of the general properties of iterated squaring constructions, which also exhibit the interrelationships between codes and lattices of different lengths. An extension called the cubing construction generates good codes and lattices of lengths N=3*2/sup n/, including the Golay code and Leech lattice, with the use of special bases for 8-space. Another related construction generates the Nordstrom-Robinson code and an analogous 16-dimensional nonlattice packing. These constructions are represented by trellis diagrams that display their structure and interrelationships and that lead to efficient maximum-likelihood decoding algorithms. >

Coset codes. I. Introduction and geometrical classification

Abstract: Practically all known good constructive coding techniques for bandlimited channels, including lattice codes and various trellis-coded modulation schemes, can be characterized as coset codes. A coset code is defined by a lattice partition Lambda / Lambda ' and by a binary encoder C that selects a sequence of cosets of the lattice Lambda '. The fundamental coding gain of a coset code, as well as other important parameters such as the error coefficient, the decoding complexity, and the constellation expansion factor, are purely geometric parameters determined by C Lambda / Lambda '. The known types of coset codes, as well as a number of new classes that systematize and generalize known codes, are classified and compared in terms of these parameters. >

ShuffleNet: an application of generalized perfect shuffles to multihop lightwave networks

Abstract: The authors propose a multihop wavelength-division-multiplexing (WDM) approach, referred to as ShuffleNet, for achieving concurrency in distributed lightwave networks. ShuffleNet can be configured with each user having as few as one fixed-wavelength transmitter and one fixed-wavelength receiver, avoiding both wavelength agility and pretransmission coordination problems. Still, the network can achieve at least 40% of the maximum efficiency possible with wavelength-agile transmitters and receivers. To transmit a packet from one user to another, however, may require routing the packet through intermediate users, each repeating the packet on a new wavelength, until the packet is finally transmitted on a wavelength that the destination user receives. For such a multihop lightwave network, the transmit and receive wavelengths must be assigned to users to provide both a path between all users and the efficient utilization of all wavelength channels. A class of assignment schemes is proposed which is based on a generalization of the perfect shuffle and achieves high efficiency for uniform traffic loads. Physically, the network may take on a variety of topologies, including a bus, tree, or star. >

Detection of coded modulation signals on linear, severely distorted channels using decision-feedback noise prediction with interleaving

Abstract: On linear bandlimited Gaussian noise channels with sufficiently high SNR, channel capacity can be approached by combining powerful coded modulation schemes designed for Nyquist channels with the equalization power of decision-feedback equalization (DFE). However, this combination may not be realized in a straightforward manner, since, in general, DFE requires delay-free decisions for feedback, and in a coded system such decisions are not sufficiently reliable. A technique is proposed that combines periodic interleaving with noise-predictive DFE, so that delayed reliable decisions can be used for feedback. When sufficient delay in the interleavers can be tolerated, this technique can attain the DFE performance. On severely distorted channels, modest delays can be sufficient to obtain respectable gains over linear equalization. >

Communication system for sending an identical routing tree to all connected nodes to establish a shortest route and transmitting messages thereafter

Abstract: The communications network includes a plurality of interconnected nodes and communication links between nodes. Computing apparatus in provided for determining a shortest path from a starting node to a destination node. The computing apparatus is adapted so that each node forms a routing tree having nodes with indentities, branches with weights, and a distinguished node called a root. The routing tree is the estimated shortest path to all of the nodes and each node communicates its routing tree to each adjacent node. Upon receipt of a routing tree by a reference node from an adjacent node, the reference node stores the routing tree and produces a large tree having roots and branches by placing the reference node as the root of the large tree and creating branches from the reference node to the roots of the routing trees of the adjacent nodes. The lengths of the branches are equal to the lengths of the links from the reference node to the adjacent nodes. A breadth first search of the large tree is performed to determine a connected subset of the large tree where each node identity appears only once. The connected subset forms the new routing tree for the reference node. If the new routing tree differs from the previous routing tree, the new routing tree is broadcast to all adjacent nodes and the procedure is repeated until no new tree differs from a previous tree, thereby defining a final routing tree. The final routing tree includes the shortest path from the reference node to all connected nodes.

Testing the performance of a communication line between two modems using modem processor

Abstract: Apparatus for testing the performance of a communication line that carries customer data between two modems, including a programmable processor (for example, a digital signal processor) in each modem that either performs modulation or demodulation of customer data or tests the line, and a controller (for example, a programmable host processor) that switches the programmable signal processor selectively between processing customer data and line testing. The communication line carries both customer data and network control commands that include a test command for initiating testing, and each host processor responds to the test command by switching the signal processor to the test mode. There are a plurality of modems forming a data network and a central network control system connected to the network for providing the test command. The test command names two of the modems to participate in the testing and the controller identifies whether a given modem is one of the participants. The controller causes all of the modems to cease processing customer data while the testing is in progress. The network carries progress signals while the testing is in progress. The controller causes all modems other than the two modems to monitor the network for the progress signals and to again process customer data when the progress signals cease to appear.

Interconnection network for multiple processors

Abstract: A large number (e.g., 20 or more) independent processors are connected by a network that provides high data throughput (e.g., 64 megabits/second). The network has redundant data, arbitration, and status/control buses. The buses extend across a plurality of microstripline backplanes connected together by impedance-matched twisted-pair cables. Data is transmitted in packets, along with status signals that provide error detection information.

Queueing in space-division packet switching

Abstract: The authors study the performance of four different approaches for providing the queuing necessary to smooth fluctuations in packet arrivals to a space-division packet switch. They are (1) input queueing, where a separate first-in, first-out (FIFO) buffer is provided at each input to switch; (2) input smoothing, where a frame of b packets is stored at each of the N input lines to the switch and simultaneously launched into a switch fabric of size Nb*Nb; (3) output queuing, where packets are queued in a separate FIFO buffer located at each output of the switch; and (4) completely shared buffering, where all queuing is done at the outputs and all buffers are completely shared among all the output lines. >

Performance comparison of error control schemes in high-speed computer communication networks

Abstract: The authors examine the performance of two different approaches for handling the loss and/or corruption of messages as they are transmitted between two end users in a high-speed network. In the link-by-link approach, two adjacent nodes in an end-to-end path locally detect and recover from message loss or corruption along their joining link. In the end-to-end approach, recovery is done solely on the basis of a single end-to-end protocol. The authors develop analytic performance models, validated with simulation, for comparing the performance of these two approaches. The authors find that for the range of network parameters of practical interest, an end-to-end approach towards error control is superior to a link-by-link approach, even under assumptions that would overly favor the link-by-link approach, while at the same time requiring fewer network resources (e.g. buffers, computation time) than the link-by-link approach. The performance differences arise primarily from the increased buffer requirements of the link-by-link approach. >

Reduced-state sequence estimation for trellis-coded modulation on intersymbol interference channels

Abstract: The authors consider the detection of trellis codes designed for ideal channels when they operate in the presence of intersymbol interference (ISI). A well-structured reduced-state sequence-estimation (RSSE) algorithm is described which can achieve the performance of maximum-likelihood sequence estimation (MLSE) at drastically reduced complexity. The authors start by constructing proper reduced-state trellises by merging the states of the ML supertrellis using set-partitioning principles. Then the Viterbi algorithm (VA) searches these trellises using a modified ML metric. The authors examine the performance and complexity of RSSE and compare it to that of MLSE and DFE (decision-feedback equalization). They also discuss the application of RSSE to higher dimensional trellis codes and to trellis coding with interleaving. >

Sample rate converter

Abstract: A sample rate converter receives an input stream of samples representative of a time-varying signal and generates a corresponding output stream of samples which is also representative of the time-varying signal and in which each output stream sample appears at a desired time which may differ from the time when a sample appears in the input stream; filter circuitry receives the input stream and generates at least two intermediate samples which appear at different times, and interpolation circuitry combines selected intermediate samples to form a sample of the output stream at the desired time. In another aspect, an improvement is provided for a modem of the kind in which a line signal is sampled at times governed by an unstable clock to generate a first sample stream in which the samples appear with possibly changing sampling phases and a converter converts the first sample stream to a second stream of samples appearing at times that are possibly different from the times of appearance of the samples in the first sample stream; the improvement is means for providing to the converter signals representative of the changing phases of the samples in the first sample stream to cause the conversion to occur in a manner that reduces the contribution of the changing sampling phases to the times of appearance of the samples in the second sample stream.

Network data flow control technique

Abstract: Data flow in a communications network is controlled by a downstream node specifying the maximum average speeds of a plurality of data streams sent from an upstream node on a plurality of data paths.

Convergence characteristics of LMS and LS adaptive algorithms for signals with rank-deficient correlation matrices

Abstract: The author investigates the convergence characteristics of the last mean square (LMS) and the recursive least squares (RLS) adaptive algorithms when the correlation matrix of the input signal does not have a full rank. It is shown that the initial convergence rate of the LMS algorithm is inversely proportional to the rank of correlation matrix, or equivalently, the number of nonzero eigenvalues. The same conclusion holds for the RLS algorithms if the minimum norm solution (MNS) is used in each iteration. A simple time-recursive method to obtain approximate MNSs in each iteration is presented and proven. The effect of additive noise is discussed. >

Systolic arrays for implementation of order-recursive least-squares adaptive filtering algorithms

Abstract: Three systolic arrays suitable for implementation of order-recursive least-squares (ORLS) adaptive algorithms are considered. It is shown that they can be constructed using two types of elementary cells. A classification of systolic implementations of the ORLS adaptive algorithms is given by exploiting the possible variations of the elementary cells. The investigation of the array structures and variations of the elementary cells leads to a systematic approach to designing reconfigurable systolic arrays for implementation of ORLS algorithms. As an application of this approach, a novel least-squares (LS) lattice algorithm based on Givens rotations is derived. >

Convergence and steady state behavior of a phase-splitting fractionally spaced equalizer

Abstract: The eigenstructure, initial convergence, and steady-state behavior of a phase-splitting fractionally spaced equalizer (PS-FSE) are analyzed. It is shown that the initial convergence rate of a T/3, or in general a T/M, PS-FSE using the least-mean-square (LMS) stochastic gradient adaptive algorithm is half that of a symbol rate equalizer (SRE) or a complex fractionally spaced equalizer (CFSE) with the same time span. The authors also show that the LMS adaptive PS-FSE with symbol rate update converges to a Hilbert transformer followed by a matched filter in cascade with an optimal SRE and thus forms an optimal receiver structure. >