Showing papers in "IEEE Transactions on Communications in 1985"
TL;DR: The analysis and simulation of a technique for combating the effects of multipath propagation and cochannel interference on a narrow-band digital mobile channel using the discrete Fourier transform to orthogonally frequency multiplex many narrow subchannels, each signaling at a very low rate, into one high-rate channel is discussed.
Abstract: This paper discusses the analysis and simulation of a technique for combating the effects of multipath propagation and cochannel interference on a narrow-band digital mobile channel. This system uses the discrete Fourier transform to orthogonally frequency multiplex many narrow subchannels, each signaling at a very low rate, into one high-rate channel. When this technique is used with pilot-based correction, the effects of flat Rayleigh fading can be reduced significantly. An improvement in signal-to-interference ratio of 6 dB can be obtained over the bursty Rayleigh channel. In addition, with each subchannel signaling at a low rate, this technique can provide added protection against delay spread. To enhance the behavior of the technique in a heavily frequency-selective environment, interpolated pilots are used. A frequency offset reference scheme is employed for the pilots to improve protection against cochannel interference.
2,627 citations
TL;DR: The receiver adapts to the actual jammer-to-signal(J/S)ratio which is critical when the level of interference is not known a priori, and optimizes the code rate and minimizes the delay required to decode a given packet.
Abstract: It is well known that if the data rate is chosen below the available channel capacity, error-free communication is possible. Furthermore, numerous practical error-correction coding techniques exist which can be chosen to meet the user's reliability constraints. However, a basic problem in designing a reliable digital communication system is still the choice of the actual code rate. While the popular rate-1/2 code rate is a reasonable, but not optimum, choice for additive Gaussian noise channels, its selection is far from optimum for channels where a high percentage of the transmitted bits are destroyed by interference. Code combining represents a technique of matching the code rate to the prevailing channel conditions. Information is transmitted in packet formats which are encoded with a relatively high-rate code, e.g., rate 1/2, which can be repeated to Obtain reliable communications when the redundancy in a rate-1/2 code is not sufficient to overcome the channel interference. The receiver combines noisy packets (code combining) to obtain a packet with a code rate which is low enough such that reliable communication is possible even for channels with extremely high error rates. By combining the minimum number of packets needed to overcome the channel conditions, the receiver optimizes the code rate and minimizes the delay required to decode a given packet. Thus, the receiver adapts to the actual jammer-to-signal (J/S) ratio which is critical when the level of interference J is not known a priori.
1,085 citations
TL;DR: A modulator that employs double integration and two-level quantization is easy to implement and is tolerant of parameter variation.
Abstract: Sigma delta modulation is viewed as a technique that employs integration and feedback to move quantization noise out of baseband. This technique may be iterated by placing feedback loop around feedback loop, but when three or more loops are used the circuit can latch into undesirable overloading modes. In the desired mode, a simple linear theory gives a good description of the modulation even when the quantization has only two levels. A modulator that employs double integration and two-level quantization is easy to implement and is tolerant of parameter variation. At sampling rates of 1 MHz it provides resolution equivalent to 16 bit PCM for voiceband signals. Digital filters that are suitable for converting the modulation to PCM are also described.
608 citations
TL;DR: A computationally simpler and effective method is proposed for estimating motion in a video sequence based on conjugate directions and another simpler technique called the one-at-a-time search, adopted as the basis for further research.
Abstract: A computationally simpler and effective method is proposed for estimating motion in a video sequence. The paper outlines a search technique based on conjugate directions [3], [4] and another simpler technique called the one-at-a-time search [3]. Based on the comparison of the two methods, the latter technique is adopted as the basis for further research. The adopted technique is compared with brute force search, existing 2-D logarithmic search [1], and a modified version of it [2], for motion compensated prediction [5].
602 citations
TL;DR: A broadcast channel access protocol called spatial TDMA is defined, which is designed specifically to operate in a multihop packet radio environment where the location of the nodes of the network is assumed to be fixed.
Abstract: In this paper we define a broadcast channel access protocol called spatial TDMA, which is designed specifically to operate in a multihop packet radio environment where the location of the nodes of the network is assumed to be fixed. The defined protocol assigns transmission rights to nodes in the network in a local TDMA fashion and is collisionfree. Methods for determining slot allocations are developed, and an approximate solution is given for determining the assignment of capacities for the links of the network that minimizes the average delay of messages in the system.
481 citations
TL;DR: A very simple method is presented for improving the efficiency of minimum distortion encoding for vector quantization by reducing the number of multiplications in a full search vector quantizer with a large number of codewords.
Abstract: In this note we present a very simple method for improving the efficiency of minimum distortion encoding for vector quantization. Simulations indicates a reduction of up to 70 percent in the number of multiplications for a full search vector quantizer with a large number of codewords, and about 25-40 percent for a tree search vector quantizer. Similar improvement can be achieved in other vector quantization systems.
455 citations
TL;DR: A polynomial time algorithm is proposed under which a channel is assigned to nodes from global, multiple-source broadcasting considerations, and it is shown that the problem of finding an optimal protocol is NP-hard.
Abstract: In this paper we develop a graph-oriented model for dealing with broadcasting in radio networks. Using this model, optimality in broadcasting protocols is defined, and it is shown that the problem of finding an optimal protocol is NP-hard. A polynomial time algorithm is proposed under which a channel is assigned to nodes from global, multiple-source broadcasting considerations. In particular, nodes participating in the broadcast do not interfere with each other's transmissions, but otherwise simultaneous channel reuse is permitted. Protocol implementations of this approach by frequency division and by time division are given. It is shown that, using these protocols, bounded delay for broadcasted messages can be guaranteed.
390 citations
TL;DR: The outphasing technique (LINC) combines two nonlinear RF power amplifiers into a linear RF power-amplifier (PA) system using saturated class-B PA's and proper selection of the shunt reactance improves efficiency.
Abstract: The outphasing technique (LINC) combines two nonlinear RF power amplifiers into a linear RF power-amplifier (PA) system. The two PA's are driven with signals of different phases, and the phases are controlled so that the addition of the PA outputs produces a system output of the desired amplitude. However, the resultant time-varying impedances presented to the PA's alter their dc power consumption and efficiency. Power and efficiency characteristics are derived for both simple (transformer-coupler) and Chireix (transmission-line-coupler with shunt reactance) outphasing systems using saturated class-B PA's. Simple outphasing systems have the efficiency characteristic of a linear class-B PA. Through proper selection of the shunt reactance, the efficiency of a Chireix outphasing system can be maximized at a specific output amplitude. The average efficiency with various amplitude-modulated signals is determined as a function of shunt reactance. Selecting the shunt reactance to fit the signal can improve efficiency by as much as a factor of 2.
389 citations
TL;DR: A new analytical solution, taking the form of a straightforward multiplicative weighting function, is developed which is readily applicable to image compression and quality assessment in conjunction with a visual model and the image cosine transform.
Abstract: Utilizing a cosine transform in image compression has several recognized performance benefits, resulting in the ability to attain large compression ratios with small quality loss. Also, incorporation of a model of the human visual system into an image compression or quality assessment technique intuitively should (and has often proven to) improve performance. Clearly, then, it should prove highly beneficial to combine the image cosine transform with a visual model. In the past, combining these two has been hindered by a fundamental problem resulting from the scene alteration that is necessary for proper cosine transform utilization. A new analytical solution to this problem, taking the form of a straightforward multiplicative weighting function, is developed in this paper. This solution is readily applicable to image compression and quality assessment in conjunction with a visual model and the image cosine transform. In the development, relevant aspects of a human visual system model are discussed, and a refined version of the mean square error quality assessment measure is given which should increase this measure's utility.
297 citations
TL;DR: This paper proposes and analyzes a waveform coding system, adaptive vector predictive coding (AVPC), in which a low-dimensional vector quantizer is used in an adaptive predictive coding scheme.
Abstract: Vector quantization, in its simplest form, may be regarded as a generalization of PCM (independent quantization of each sample of a waveform) to what might be called "vector PCM," where a block of consecutive samples, a vector, is simultaneously quantized as one unit. In theory, a performance arbitrarily close to the ultimate rate-distortion limit is achievable with waveform vector quantization if the dimension of the vector, k , is large enough. The main obstacle in effectively using vector quantization is complexity. A vector quantizer of dimension k operating at a rate of r bits/sample requires a number of computations on the order of k2^{kr} and a memory of the same order. However, a low-dimensional vector quantizer (dimensions 4-8) achieves a remarkable improvement over scalar quantization (PCM). Consequently, using the vector quantizer as a building block and imbedding it with other waveform data compression techniques may lead to the development of a new and powerful class of waveform coding systems. This paper proposes and analyzes a waveform coding system, adaptive vector predictive coding (AVPC), in which a low-dimensionality vector quantizer is used in an adaptive predictive coding scheme. In the encoding process, a locally generated prediction of the current input vector is subtracted from the current vector, and the resulting error vector is coded by a vector quantizer. Each frame consisting of many vectors is classified into one of m statistical types. This classification determines which one of m fixed predictors and of m vector quantizers will be used for encoding the current frame.
295 citations
TL;DR: Bit error rate (BER) is analyzed theoretically for diversity reception in Nakagami fading environment using an M -branch maximal ratio combiner (MRC) and the results are extended to include coherent phase shift keying (CPSK) and differential phase shiftkeying (DPSK).
Abstract: Bit error rate (BER) is analyzed theoretically for diversity reception in Nakagami fading environment using an M -branch maximal ratio combiner (MRC). Coherent and incoherent reception of frequency shift keying (FSK) are considered, using the multiple branch diversity system for both identical and different diversity branch fading parameters. The effect of correlation is also considered for the dual diversity case. The results are extended to include coherent phase shift keying (CPSK) and differential phase shift keying (DPSK).
TL;DR: The channel throughput for a finite number of packet broadcasting users is analyzed for random access protocols, including slotted persistent carrier sense multiple access (CSMA) with and without collision detection, and can be extended to infinite population cases by taking the proper limit.
Abstract: The channel throughput for a finite number of packet broadcasting users is analyzed for random access protocols, including slotted persistent carrier sense multiple access (CSMA) with and without collision detection and unslotted persistent CSMA with and without collision detection. We consider both p - and 1-persistent CSMA. Our results can be extended to infinite population cases (by taking the proper limit), where they agree with the known throughput expressions when available.
TL;DR: It is shown, arguing from the form of the equations, that there is a conservation law in effect in this system, that the nodal mean waiting times are weighted by the relative intensity, defined here as the intensity weighted mean.
Abstract: This paper derives exact results for a token ring system with exhaustive or gated service. There are N nodes on the ring and control is passed sequentially from one to the next. Messages with random lengths arrive at each node and are placed on the ring when the control arrives at that node. Exhaustive service means that the queue at a node is empty before the token is released and gated means that only those messages in the queue at the arrival of the token are served at that cycle. Generating function recursions for the terminal service time (the total sojourn time of a token at a node) and, from this, joint cycle and intervisit times are derived. Using known results relating the marginal generating functions of the waiting time and the cycle and intervisit time, it is shown that the N mean waiting times at the nodes require the solution of N(N - 1) and N2equations for the exhaustive and gated cases, respectively. The arrival processes are assumed to be Poisson with different rates and the service processes are general and different at each node. In addition the token overhead is allowed to have an arbitrary but independent distribution at each node. Explicit, simply programmed equations are given. It is shown, arguing from the form of the equations, that there is a conservation law in effect in this system. If the nodal mean waiting times are weighted by the relative intensity, defined here as the intensity weighted mean, then the sum takes on a particularly simple form and is independent of the placement of the nodes on the ring. When the service means at each node are equal, this quantity is just the system mean waiting time.
TL;DR: The performance of the motion compensated prediction developed here is investigated for various block sizes and is compared to other techniques.
Abstract: Interframe motion estimation of subblocks based on improved search techniques is developed. These techniques are based on minimizing the mean difference between the subblock in question in the present frame and the displaced subblock in the previous frame. The performance of the motion compensated prediction developed here is investigated for various block sizes and is compared to other techniques.
IBM1
TL;DR: The objective is to permit the transmission of a given pattern of traffic, while ensuring that the number of times that the on-board switch needs to be reconfigured is minimized, using an efficient heuristic algorithm.
Abstract: In this paper, we investigate the problem of constructing a TDMA frame for a multibeam satellite system. Our objective is to permit the transmission of a given pattern of traffic, while ensuring that the number of times that the on-board switch needs to be reconfigured is minimized. We find that the underlying optimization problem is computationally intractable, but go on to suggest an efficient heuristic algorithm which we validate through experiments on randomly generated traffic patterns.
TL;DR: The performance of coherent direct-sequence spread-spectrum communications over specular multipath fading channels is investigated and the average probability of error of the correlation receiver is derived for an arbitrary number of paths with deterministic or random gain coefficients.
Abstract: The performance of coherent direct-sequence spread-spectrum communications over specular multipath fading channels is investigated. The average probability of error of the correlation receiver is derived for an arbitrary number of paths with deterministic or random gain coefficients. The gain coefficients, delays, and phase angles of any two distinct paths are modeled as mutually independent random variables. Numerical results for several values of the system and channel parameters are presented.
TL;DR: A combination of a subgradient optimization procedure and an augmented Lagrangean-based procedure is used to generate tight lower bounds for capacitated spanning tree problems.
Abstract: Capacitated spanning tree problems appear frequently as fundamental problems in many communication network design problems. An integer programming formulation and a new set of valid inequalities are presented for the linear characterization of the problem. A combination of a subgradient optimization procedure and an augmented Lagrangean-based procedure is used to generate tight lower bounds. The procedure begins with an explicit representation of a subset of the constraints, and the corresponding Lagrangean problem is solved. The solution is examined in order to identify implicit constraints that are violated. Those are added to the Lagrangean problem, forming an expanded problem, and an efficient dual ascent procedure is then applied. When no further improvement is possible through this procedure, a subgradient optimization procedure is invoked in order to further tighten the lower bound value. An exchange heuristic is applied to the nonfeasible Lagrangean solution, in an attempt to generate good feasible solutions to the problem. The procedure has been tested and has generated bounds that are significantly better than ones obtained through previously published procedures.
IBM1
TL;DR: This work investigates flow-control issues in local-area networks consisting of multiple token rings interconnected through bridges and suggests an enhancement to this protocol in the form of a dynamic flow- control algorithm that guarantees close-to-optimal network performance under both normal traffic load and overload conditions.
Abstract: We investigate flow-control issues in local-area networks consisting of multiple token rings interconnected through bridges. To achieve high throughput, bridges perform only a very simple routing and store-and-forward function, but are not involved in error- or flowcontrol. In case of congestion, bridges discard arriving frames, which will be recovered through an appropriate end-to-end protocol between the communicating stations. The end-to-end protocol considered is the IEEE 802.2 type-2 logical-link-control (LLC) protocol. Extensive simulations show that performance can be severely degraded if, in such a network, the LLC protocol is employed as defined today. Therefore, we suggest an enhancement to this protocol in the form of a dynamic flow-control algorithm. As our results demonstrate, this enhancement guarantees close-to-optimal network performance under both normal traffic load and overload conditions.
TL;DR: An efficient encoding scheme for arbitrary curves, based on the chain code representation, has been proposed, whose code amount is about 50-60 percent of that required for the conventional chain encoding scheme.
Abstract: An efficient encoding scheme for arbitrary curves, based on the chain code representation, has been proposed. The encoding scheme takes advantage of the property that a curve with gentle curvature is divided into somewhat long curve sections (segments), each of which is represented by a sequence of two adjacent-direction chain codes. The coding efficiency of the scheme becomes higher as the segments become longer, while a variable-length coding technique makes it possible to encode short segments without an extreme increase in code amount. An experiment with complicated curves obtained from geographic maps has shown a high data compression rate of the proposed scheme, whose code amount is about 50-60 percent of that required for the conventional chain encoding scheme.
TL;DR: This paper describes a technique for increasing the efficiency of an ARQ communication system utilizing error detection codes, in which the blocks containing errors are not discarded at the receiver, as in the conventional ARQ systems.
Abstract: This paper describes a technique for increasing the efficiency of an ARQ communication system utilizing error detection codes. In this technique the blocks containing errors are not discarded at the receiver, as in the conventional ARQ systems. Such blocks are retained in order to make use of the information contained in them. A reliability value is associated to each demodulated symbol; this reliability value is updated every time a new retransmission is made, even if the vector received is in error. The process of updating the symbols often eliminates automatically the errors introduced by the channel. The results obtained by means of a simulation show that this method offers a considerable reduction in the average number of retransmissions, in comparison with conventional ARQ systems.
TL;DR: Examination of speech impairments likely to arise in dynamically managed voice (DMV) systems, which utilize speech activity detection to exploit speech idle time and variable bit rate coding to exploit nonstationary speech statistics, finds two impairments not commonly found in traditional communication systems variable Speech burst delay and speech clipping.
Abstract: The purpose of this paper is to examine speech impairments likely to arise in dynamically managed voice (DMV) systems. DMV systems utilize speech activity detection to exploit speech idle time and variable bit rate coding to exploit nonstationary speech statistics. The emphasis here is on systems using speech detection. This processing introduces two impairments not commonly found in traditional communication systems variable Speech burst delay and speech clipping. Simulations of these impairments were implemented, and formal subjective testing was performed to assess subjects' reactions to a range of impairment levels. Emphasis was on formal subjective listening tests and customer opinion of speech quality as defined by a rating scale. The test conditions are applicable to general telephony, where relatively high speech quality is required. Results on variable speech burst delay and front-end and midspeech burst clipping are presented. These results serve as input to the design process and to the establishment of performance guidelines for DMV systems.
TL;DR: The main advantages of the lattice DFE's are their numerical stability, their computational efficiency, the flexibility to change their length, and their excellent capabilities for tracking rapidly time-variant channels.
Abstract: This paper presents two types of adaptive lattice decisionfeedback equalizers (DFE), the least squares (LS) lattice DFE and the gradient lattice DFE. Their performance has been investigated on both time-invariant and time-variant channels through computer simulations and compared to other kinds of equalizers. An analysis of the self-noise and tracking characteristics of the LS DFE and the DFE employing the Widrow-Hoff least mean square adaptive algorithm (LMS DFE) are also given. The analysis and simulation results show that the LS lattice DFE has the faster initial convergence rate, while the gradient lattice DFE is computationally more efficient. The main advantages of the lattice DFE's are their numerical stability, their computational efficiency, the flexibility to change their length, and their excellent capabilities for tracking rapidly time-variant channels.
TL;DR: It is shown that the use of ReedSolomon coding with a parallel errors and erasures decoding algorithm accomplishes the goals of providing good performance in partial-band Gaussian noise interference by use of coding and diversity with an efficient error-correction algorithm.
Abstract: This paper is concerned with the performance of a Communications system which utilizes frequency-hop spread spectrum, diversity transmission, Reed-Solomon coding, and parallel error-correction and erasure-correction decoding. Both binary signaling and M -ary orthogonal signaling are considered. The goals are twofold. First, it is desirable to provide good performance in partial-band Gaussian noise interference by use of coding and diversity with an efficient error-correction algorithm. Second, it is necessary to totally neutralize narrow-band interference (regardless of the power level or statistical distribution of the interference) in order to have an effective spread-spectrum system. Through an analysis of the effects of partial-band interference on a frequency-hop spread-spectrum system with diversity, it is shown that the use of ReedSolomon coding with a parallel errors and erasures decoding algorithm accomplishes these goals. The paper also investigates the accuracy of the Chernoff bound as an approximation to the true performance of a frequency-hop spreadspectrum communication system with diversity; side information, M -ary orthogonal signaling, and Reed-Solomon coding. The performance results presented in the paper are based on analysis and computer evaluation. Approximate results based on the Chernoff bound are also given. It is shown that the Chernoff bound for M -ary orthogonal signaling gives a very poor approximation for many cases of interest. This is largely due to the looseness of the union bound.
TL;DR: In this article, a discrete-time multiaccess channel is considered where the outcome of a transmission is either "idle", "success", or "collision", depending on the number of users transmitting simultaneously.
Abstract: A discrete time multiaccess channel is considered where the outcome of a transmission is either "idle," "success," or "collision," depending on the number of users transmitting simultaneously. Messages involved in a "collision" must be retransmitted. An efficient access allocation policy is developed for the case where infinitely many sources generate traffic in a Poisson manner and can all observe the outcomes of the previous transmissions. Its rate of success is 0.48776. Modifications are presented for the cases where the transmission times depend on the transmission outcomes and where observations are noisy.
TL;DR: It is shown that the best combined performance is obtained when the two strategies for managing the access of two types of traffic, a blockable wide-band (WB) type of traffic and a queueable narrow- band (NB) typeof traffic, are adaptively combined according to the offered load.
Abstract: A common digital transmission facility provides service to a community of heterogeneous users generating traffic with differing intensity, message length, and bit rate. In order for this type of integrated communication system to handle its traffic demands with high efficiency and flexibility, close control of access and switching at the input node is required. We propose, analyze, and compare two different strategies for managing the access of two types of traffic, a blockable wide-band (WB) type of traffic and a queueable narrow-band (NB) type of traffic, sharing the transmission resource dynamically. The first strategy assigns preemptive priority to the WB traffic over the NB traffic, whereas the second strategy employs a wide-band to narrow-band bit rate compression mechanism. Exact analytic models are developed, and solution methods are presented and implemented. It is shown that the best combined performance is obtained when the two strategies are adaptively combined according to the offered load.
TL;DR: Simulations on real images show significant improvement over the conventional DPCM and tree codes using these new techniques, and the strong robustness property of these coding schemes is also experimentally demonstrated.
Abstract: The purpose of this paper is to present new image coding schemes based on a predictive vector quantization (PVQ) approach. The predictive part of the encoder is used to partially remove redundancy, and the VQ part further removes the residual redundancy and selects good quantization levels for the global waveform. Two implementations of this coding approach have been devised, namely, sliding block PVQ and block tree PVQ. Simulations on real images show significant improvement over the conventional DPCM and tree codes using these new techniques. The strong robustness property of these coding schemes is also experimentally demonstrated.
TL;DR: A new carrier sense multiple access (CSMA) algorithm, called virtual time CSMA, is described and analyzed, which uses a novel approach to granting access to the shared broadcast channel based on variable-rate clocks.
Abstract: A new carrier sense multiple access (CSMA) algorithm, called virtual time CSMA, is described and analyzed. This algorithm uses a novel approach to granting access to the shared broadcast channel based on variable-rate clocks. Unlike other CSMA algorithms, the operation of virtual time CSMA reduces to the ideal case in the zero propagation time limit: a work-conserving, first-come first-served M/G/1 queueing system. The algorithm does not appear to be difficult to implement, but offers better throughput-delay performance than existing CSMA algorithms. A simple closed form technique for estimating the mean message delay is presented. This technique is of independent interest because of its applicability to certain "sliding window" tree conflict resolution algorithms. Extensive numerical results for the algorithm are presented, including comparisons with simulation and with other CSMA algorithms.
TL;DR: Details are provided of the design and performance of the frequency-difference detector that lies at the heart of the loop, which is employed in frequency-tracking loops in digital-data receivers.
Abstract: Among other applications, frequency-tracking loops are employed in digital-data receivers, either as a frequency-acquisition aid for phase-locked coherent reception, or as the sole carrier-frequency control for noncoherent reception. This article provides details of design and performance of the frequency-difference detector that lies at the heart of the loop.
TL;DR: In this paper the case of r = 1/2 coding onto a 4-ary CPM is emphasized, with short-constraint length codes presented for continuous-phase FSK, double-raised-cosine, and triple- raised-Cosine modulation.
Abstract: Background theory and specific coding designs for combined coding/modulation schemes utilizing convolutional codes and continuous-phase modulation (CPM) are presented. In this paper the case of r = 1/2 coding onto a 4-ary CPM is emphasized, with short-constraint length codes presented for continuous-phase FSK, double-raised-cosine, and triple-raised-cosine modulation. Coding buys several decibels of coding gain over the Gaussian channel, with an attendant increase of bandwidth. Performance comparisons in the power-bandwidth tradeoff with other approaches are made.
TL;DR: The algorithm for FSVQ design for waveform coders is extended to FSVZ design of linear predictive coded (LPC) speech parameter vectors using an Itakura-Saito distortion measure and a new technique for the iterative improvement of the next-state function based on an algorithm from adaptive stochastic automata theory is introduced.
Abstract: A finite-state vector quantizer (FSVQ) is a switched vector quantizer where the sequence of quantizers selected by the encoder can be tracked by the decoder. It can be viewed as an adaptive vector quantizer with backward estimation, a vector generalization of an AQB system. Recently a family of algorithms for the design of FSVQ's for waveform coding application has been introduced. These algorithms first design an initial set of vector quantizers together with a next-state function giving the rule by which the next quantizer is selected. The codebooks of this initial FSVQ are then iteratively improved by a natural extension of the usual memoryless vector quantizer design algorithm. The next-state function, however, is not modified from its initial form. In this paper we present two extensions of the FSVQ design algorithms. First, the algorithm for FSVQ design for waveform coders is extended to FSVQ design of linear predictive coded (LPC) speech parameter vectors using an Itakura-Saito distortion measure. Second, we introduce a new technique for the iterative improvement of the next-state function based on an algorithm from adaptive stochastic automata theory. The design algorithms are simulated for an LPC FSVQ and the results are compared with each other and to ordinary memoryless vector quantization. Several open problems suggested by the simulation results are presented.