Showing papers in "IEEE Transactions on Communications in 1976"

Recoverability of Communication Protocols--Implications of a Theoretical Study

Abstract: A study is presented which permits the formal analysis and synthesis of recoverable computer communication protocols. This study is based on a formal representation of processes by a model of computation, the Petri nets (PN's). The PN model is generalized to include a representation of the possible failures, and then, the concept of "recoverability" is formally defined. A set of necessary and sufficient conditions which a process must satisfy in order to be recoverable is derived. In the PN model, the processes that satisfy these conditions are shown to have some practical limitations. A new model, the time-Petri net (TPN), is introduced to remove these limitations. This new model allows the introduction of constraints in the execution times of its part. As shown in this paper, the TPN appears to be a suitable model for the study of practical recoverable processes. Several practical communication protocols are formally designed and analyzed using this new model, and some interesting properties of these protocols are formally derived.

Timing Recovery in Digital Synchronous Data Receivers

Abstract: A new class of fast-converging timing recovery methods for synchronous digital data receivers is investigated. Starting with a worst-case timing offset, convergence with random binary data will typically occur within 10-20 symbols. The input signal is sampled at the baud rate; these samples are then processed to derive a suitable control signal to adjust the timing phase. A general method is outlined to obtain near-minimum-variance estimates of the timing offset with respect to a given steady-state sampling criterion. Although we make certain independence assumptions between successive samples and postulate ideal decisions to obtain convenient analytical results, our simulations with a decision-directed reference and baud-to-baud adjustments yield very similar results. Convergence is exponential, and for small loop gains the residual jitter is proportional and convergence time is inversely proportional to the loop gain. The proposed algorithms are simple and economic to implement. They apply to binary or multilevel PAM signals as well as to partial response signals.

Fractional Tap-Spacing Equalizer and Consequences for Clock Recovery in Data Modems

Abstract: Adaptive equalizers are usually realized in the form of a transversal filter with variable tap gains and tap spacing equal to the symbol spacing T . The performance of these equalizers depends critically on the symbol-clock phase derived in the receiver, due to the clock-phase dependent aliasing of the spectral roll-off components, upon which the conventional equalizer has no influence. In this paper we study the possibility of overcoming this difficulty by making the tap spacing of the equalizer somewhat smaller than T (fractional tap spacing). It is shown that this leads to satisfactory performance of the equalizer for a broad continuous range of clock phases, without penalizing the speed of convergence. Furthermore, it allows the application of a simple clock recovery scheme which derives a phase control signal from the equalizer tap-gain values.

Singular Value Decomposition (SVD) Image Coding

Abstract: The numerical techniques of transform image coding are well known in the image bandwidth compression literature. This concise paper presents a new transform method in which the singular values and singular vectors of an image are computed and transmitted instead of transform coefficients. The singular value decomposition (SVD) method is known to be the deterministically optimal transform for energy compaction [2]. A systems implementation is hypothesized, and a variety of coding strategies is developed. Statistical properties of the SVD are discussed and a self adaptive set of experimental results is presented, Imagery compressed to 1, 1.5, and 2.5 bits per pixel with less than 1.6, 1, and 1/3 percent, respective mean-square error is displayed. Finally, additional image coding scenarios are postulated for further consideration.

Maximum Likelihood Receiver for Multiple Channel Transmission Systems

Abstract: A maximum likelihood (ML) estimator for digital sequences disturbed by Gaussian noise, intersymbol interference (ISI) and interchannel interference (ICI) is derived It is shown that the sampled outputs of the multiple matched filter (MMF) form a set of sufficient statistics for estimating the input vector sequence Two ML vector sequence estimation algorithms are presented One makes use of the sampled output data of the multiple whitened matched filter and is called the vector Viterbi algorithm The other one is a modification of the vector Viterbi algorithm and uses directly the sampled output of the MMF It appears that, under a certain condition, the error performance is asymptotically as good as if both ISI and ICI were absent

On Improving Utilization in ALOHA Networks

Abstract: By the simple expedient of dividing users into two groups-one transmitting at high power and the other at low powerthe maximum utilization of a slotted ALOHA communication system can be increased from 36.8 percent to about 53 percent. Similar comments apply to the unslotted ALOHA case. An extension to more than two power groups also is described.

MSK and Offset QPSK Modulation

Abstract: Minimum shift keying (MSK) and offset keyed quadrature phase shift keying (OK-QPSK) modulation techniques are often proposed for use on nonlinear, severely band-limited communication channels because both techniques retain low sidelobe levels on such channels, while allowing efficient detection performance. A more detailed performance comparison of the two techniques on such channels is, therefore, of interest. In this paper a Markov process representation is developed which is applicable to either the MSK or OK-QPSK waveform. This representation is employed to illustrate the similarity between the modulation processes and to obtain the autocorrelations and power spectral densities of the two waveforms. This Markov process representation may be similarly employed with other modulation waveforms of the same class. The autocorrelations and power Spectral densities of MSK and offset QPSK provide initial insight to expected performance on band-limited channels. The results of a digital computer simulation are presented. The simulation compares the bit error rates (BER's) of MSK and offset QPSK on nonlinear, band-limited double-hop links such as encountered in satellite communications. The simmulation results are presented as E_{b}/N_{0} degradation with respect to ideal detection versus channel noise bandwidth. The error probability was used as a performance metric, and equal adjacent channel interference as a constraint. For the channels simulated, MSK is found to provide superior performance when the channel noise bandwidth exceeds about 1.1 times the binary data rate. For narrower bandwidths, offset QPSK provides superior performance.

Optimal Linear Coding for Vector Channels

Abstract: This paper is concerned with the problem of obtaining the optimal linear vector coding (transformation) method that matches an r -dimensional vector signal and a k -dimensional channel under a given channel power constraint and mean-squared-error criterion. The encoder converts the r correlated random variables into r independent random variables and selects at most k independent random variables which correspond to the k largest eigenvaiues of the signal covariance matrix Q . The encoder reinserts cross correlation into the k random variables in such a way that the largest eigenvalue of Q is assigned to the smallest eigenvalue of the channel noise covariance matrix R and the second largest eigenvalue of Q to the second smallest eigenvalue of R , etc. When only the total power for all k channels is prescribed, the optimal individual channel power assignments are obtained in terms of the total power, the eigenvalues of Q , and the eigenvalues of R . When the individual channel power limits are constrained by P_{1}, ..., P_{k} and R is a diagonal matrix, the necessary conditions of an inverse eigenvalue problem must be satisfied to optimize the vector signal transmission system. An iterative numerical method has been developed for the case of correlated channel noise.

Two's Complement Pipeline Multipliers

Abstract: Digital filters and signal processors when realized in hardware often use serial transfer of data. Multipliers which are capable of accepting variable coefficients and data in sign and magnitude notation and producing serial products of the same length as the input data word have been known for some time. This concise paper addresses the design of multipliers capable of accepting data in 2's complement notation, or both data and coefficients in 2's complement notation. It also considers multiplier recoding techniques, such as the Booth algorithm. Specialized (fixed coefficient) multiplier designs are considered briefly. Finally, multiplier rounding and overflow characteristics are discussed, and a rough comparison is made between the complexity of the various designs.

A Fast Karhunen-Loeve Transform for a Class of Random Processes

Abstract: The Karhunen-Loeve transform for a class of signals is proven to be a set of periodic sine functions and this Karhunen-Loeve series expansion can be obtained via an FFT algorithm. This fast algorithm obtained could be useful in data compression and other mean-square signal processing applications.

Packet Switching in Radio Channels: Part III--Polling and (Dynamic) Split-Channel Reservation Multiple Access

Abstract: Here we continue the analytic study of packet switching in radio channels which we reported upon m our two previous papers [1], [2] Again we consider a population of terminals communicating with a central station over a packet-switched radio channel. The allocation of bandwidth among the contending terminals can be fixed [e.g., time-division multiple access (TDMA) or frequency-division multiple access (FDMA)], random [e.g., ALOHA or carrier sense multiple access (CSMA)] or centrally controlled (e.g., polling or reservation). In this paper we show that with a large population of bursty users, (as expected) random access is superior to both fixed assignment and polling. We also introduce and analyze a dynamic reservation technique which we call split-channel reservation multiple access (SRMA) which is interesting in that it is both simple and efficient over a large range of system parameters.

The Capture Effect in FM Receivers

Abstract: In this paper a theoretical explanation of the capture effect is given by calculating the instantaneous frequency of the output signal of a limiter when two frequency modulated (FM) signals are present at the limiter input. When this signal is applied to a demodulator with unlimited bandwidth, the output signal of the demodulator proves to have an extreme capture effect. When however the demodulator bandwidth is limited, the capture effect is shown not be be extreme. This phenomenon is explained and possibilities are given to minimize the capture effect. Some of the results of measurements on limiters and demodulators are given in this paper; they prove that a weak capture effect can be obtained. A method of calculating the degree of capturing is included.

A Model for Evaluating the Performance of an Integrated Circuit- and Packet-Switched Multiplex Structure

Abstract: Recent papers have introduced a multiplexing structure for mixing voice and data traffic in an integrated telecommunications system. This structure utilizes a master frame format of a time division statistical multiplex facility. A certain portion of the frame is allocated to voice calls, and data traffic is assigned to the remaining frame capacity. To achieve a high transmission utilization, data are allowed to use any residual voice capacity momentarily available due to statistical variations in the voice traffic. The voice traffic is treated as a loss system and data packets are buffered. In this paper we derive exact analytical expressions for the key system perfomance measures, the probability of loss for voice calls, and the expected waiting time for data packets. Actually, two cases are considered, the one discussed above, called the movable boundary case, and one where the boundary is fixed; i.e., data are not allowed to utilize the residual voice capacity. The computational aspects of calculating actual numbers are discussed in some detail, and results are presented for typical cases.

A Methodology for the Design and Implementation of Communication Protocols

Abstract: This paper presents a coherent method which permits the specification, check out, and implementation of communication protocols. Although this work is mainly directed towards telephone signaling in a programmable environment, the philosophy and most of the tools can be effectively used to handle the most general situations in which two entities interact by any kind of protocol. Since the proposed approach reduces the possibility of errors and inconsistencies, minimizes the programming needed, and improves the documentation and understandability, we believe that this methodology will facilitate the installation or modification of complex trunk protocols and other communication protocols.

Pulse and Spectrum Manipulation in the Minimum (Frequency) Shift Keying (MSK) Format

Abstract: The minimum (frequency) shift keying (MSK) format is generalized to enable further improvements in the spectrum while retaining the well-known constant envelope property and the excellent communications efficiency of MSK. A specific alternative to MSK is presented, and the spectrum is compared with that of MSK. The theoretical limitations on possible further spectrum improvements are discussed.

Properties of the Eigenvectors of Persymmetric Matrices with Applications to Communication Theory

Abstract: 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.

Average- and Median-Based Smoothing Techniques for Improving Digital Speech Quality in the Presence of Transmission Errors

Abstract: In digital speech communication, transmission errors generally introduce impulsive distortions in the received speech waveform. Smoothing of this waveform results at once in a squelching of the distortion component, and in an undesirable smearing of the speech. However, our experience with practical differential PCM (DPCM) codes (with adaptive quantizers and first-order predictors) has shown that if the error probability is fairly significant (for example, 0.025), the noise attenuation is perceptually desirable in spite of the attendant speech-muffling. Our observations are based on computer simulations and informal listening tests. The smoothing can be performed either on the received DPCM word (prediction error signal) or the reconstructed speech amplitude. (The two signals are identical in nondifferential PCM.) Smoothing algorithms can be either linear (based, for example, on running averages) or nonlinear (based, for example, on running medians). Studies with 3-bit quantizers indicate that with independently occurring transmission errors, smoothing of the prediction error signal is perceptually desirable, although the benefits decrease as a function of the predictor coefficient a, with the maximum advantage showing up for a = 0 (PCM). Running averages and running medians seem to work equally well, and suggested block lengths for their computations are three or five samples. Results with clustered transmission errors show a higher advantage due to smoothing and a preference of linear methods and longer block lengths.

The Detection of Signals in Impulsive Noise Modeled as a Mixture Process

Abstract: A first-order mixture noise density is considered as a model for impulsive noise channels It consists of a mixture of a small variance, probably Gaussian, background noise pdf and a large variance impulsive pdf Optimal nonlinear detector structures for known discretetime signals in such noise are derived Their large-sample performance compared to that of a linear detector is studied using asymptotic relative efficiency

Convolutional Code Performance in the Rician Fading Channel

Abstract: The performance of short constraint length convolutional codes in conjunction with binary phase-shift keyed (BPSK) modulation and Viterbi maximum likelihood decoding on the classical Rician fading channel is examined in detail. Primary interest is in the bit error probability performance as a function of E_{b}/N_{0} parameterized by the fading channel parameters. Fairly general upper bounds on bit error probability performance in the presence of fading are obtained and compared with simulation results in the two extremes of zero channel memory and infinite channel memory. The efficacy of simple block interleaving in combating the memory of the channel is thoroughly explored. Results include the effects of fading on tracking loop performance and the subsequent impact on overall coded system performance. The approach is analytical where possible; otherwise resort is made to digital computer simulation.

Using Triangularly Weighted Interpolation to Get 13-Bit PCM from a Sigma-Delta Modulator

Abstract: We present and analyze a method of interpolation that improves the amplitude resolution of an analog-to-digital converter. The technique requires feedback around a quantizer that operates at high speed and digital accumulation of its quantized values to provide a PCM output. We show that use of appropriate weights in the accumulation has important advantages for providing finer resoution, less spectral distortion, and white quantization noise. The theoretical discussion is supplemented by the report of a practical converter designed especially to show up the strengths and weaknesses of the technique. This converter comprises a sigma-delta modulator operating at 8 MHz and an accumulation of the 1-bit code with triangularly distributed weights. 13-bit resolution at 8 kwords/s is realized by periodically dumping the accumulation to the output. We present a practical method for overcoming a thresholding action that distorts low-amplitude input signals.

A Generalization of Minimum-Shift-Keying (MSK)-Type Signaling Based Upon Input Data Symbol Pulse Shaping

Abstract: In recent years, minimum-shift-keying (MSK) has gained increasing popularity as a modulation technique because of its desirable spectral properties. Quite often, the spectral concentration provided by MSK is not sufficient to meet requirements on out-of-band energy spillover. In these situations, one might apply additional input pulse shaping m such a way as to still maintain constant envelope signals. The properties of such MSK-type signals are the subject of this paper. Specific examples are included as illustrations of the theory both for the binary and M -ary cases.

A New Digital Echo Canceler for Two-Wire Full-Duplex Data Transmission

Abstract: A new approach to echo canceling for two-wire fullduplex data transmission is proposed. The canceling signal is directly synthesized from the binary data, using a transversal filter approach, and the usual multiplications are replaced by additions and subtractions, thus allowing efficient operation of a large number of taps as required for the canceling of distant echoes. As a specific application, a system processing one sample per baud is discussed where timing signals at both communicating stations are assumed to be synchronized. A stochastic adjustment gradient-type algorithm is used for both training and adaptive tracking of the canceler. It is shown that convergence does not depend on intersymbol interference, timing phase, carrier phase, or the energy ratio of the local to the received signal, but is a function only of the number of taps. Convergence time is proportional to that number, and the optimum step size for fastest convergence is equal to the reciprocal of the number of taps. The residual fluctuation noise is proportional to that part of the mean-square (MS) error which cannot be reduced by the canceler and is a simple function of the product of the tap signal and the step size. The predicted convergence properties are verified by simulation results. Finally, it is shown how such an echo canceler might be used to allow two-wire full-duplex transmission for data rates as high as 4800 bit/s.

Behavior of Tandem Buffers with Geometric Input and Markovian Output

Abstract: A discrete-time system of infinite-capacity buffers in tandem is studied. The input process to the first buffer consists of individual arrivals characterized by a geometric distribution of the time between arrivals; and the probability that the output channel of each buffer, except the last, is transmitting at any epoch depends only on the number of digits in the buffer at that epoch. The transmitting state of the last buffer may depend more generally on the history of that buffer. The analysis shows that in equilibrium, the input processes to the subsequent buffers in the system are geometric with the same parameter as the input process to the first buffer. Therefore, each buffer in the system can be analyzed separately. Furthermore, the equilibrium state probabilities for a given buffer at a given epoch are independent of those for any other buffer in the system at the same epoch.

Delay-Lock Tracking of Stochastic Signals

Abstract: The measurement and tracking of the delay between two versions of a stochastic signal by cross-correlation techniques is considered. Such techniques have broad applications, e.g., interferometry, noncontact speed and distance measurement, etc. The paper begins by discussing the functional diagram of the tracking system. From this diagram a mathematically equivalent model of the system is derived and its similarities to the well known baseband model of the phase-locked loop are discussed. Using Fokker-Planck (F-P) techniques the performance of the system, as a function of fundamental system parameters, is computed and graphically illustrated. These results are then compared with experimental results obtained by computer simulation.

Optical Pulse Formats for Fiber Optic Digital Communications

Abstract: Some new optical pulse formats are investigated for solving practical problems in fiber optic communication systems. These pulse formats provide many advantageous features such as error monitoring capability, abundant timing information, uniform optical power utilization, stable detection of optical input, and so forth. It is shown that a modification of Personick's receiver design theory can be used for comparison of various optical pulse formats. The comparison suggests that for state-of-the-art fiber systems with moderate fiber loss and moderate repeater spacing, where no pulse equalization is required, some new classes of 1 binary digit converted to 2 binary digits (1B2B) or 2B3B formats will permit the realization of very simple and reliable repeaters for fiber optic digital transmission. A future low-loss fiber system may permit a very long repeater spacing with the help of equalization. In this case, application of the correlative signal-processing technique is shown to be very promising. Experimental 6.3 Mbit/s and 100 Mbit/s transmissions demonstrate some advantageous features of these optical pulse formats.

Analysis of Digital Errors in Nonlinear PCM Systems

Abstract: The effect of digital errors in linear pulse code modulation (PCM) systems has been considered by several authors for channels with independent errors. In this paper we present a general approach of analyzing digital errors in linear and nonlinear PCM systems with arbitrary channels. The examples given in this paper cover the different standard nonlinear PCM systems and independent channel errors. The technique used for the analysis leads to the conclusion that there are PCM codes less sensitive to digital errors than the standard binary folded PCM code. An example of such a PCM code is given in this paper.

Symbol Error Probabilities for M-ary CPFSK: Coherent and Noncoherent Detection

Abstract: Continuous-phase frequency shift keying (CPFSK) is discussed and theoretical predictions for symbol error probabilities are derived, where the memory inherent in the phase continuity is used to improve performance. Previously known results concluded that binary CPFSK can outperform coherently detected PSK at high SNR. New results presented here show that M -ary CPFSK outperforms more tranditionally used M -ary modulation systems. Specifically, coherently detected quaternary CPFSK with a five-symbol interval decision can outperform coherent QPSK by 3.5 dB, and octal coherent CPFSK with a three- symbol decision can outperform octal orthogonal signaling by 2.6 dB at high SNR. Results for coherently detected and noncoherently detected CPFSK are derived. These performance improvements are estimates derived from symbol error probability upper bounds. Monte Carlo simulation was performed which then verified the results.

On Routing and "Delta Routing": A Taxonomy and Performance Comparison of Techniques for Packet-Switched Networks

Abstract: An inherent capability of packet-switched networks is the speed at which they can be reconfigured; various dynamic or adaptive routing techniques have been conceived to exploit this capability. In this study, existing techniques are described and an "ultra-dynamic" technique, delta routing, is invented. Several promising techniques are then selected for comparison with one another and with a network's ultimate carrying capacity. The goal is to shed light on the questions if and when one should use which kind of adaptive routing. First, a taxonomy of routing strategies is presented. In addition to delta routing, random, proportional, shortest path, and shortest path fixed-for-session duration are selected for comparison and their mechanisms described. The delay and efficiency performance of the five techniques are then compared with one another and with ideal behavior via simulations. These have been carried out concentrating on four very small networks, each with very different characteristics with the intention of gaining insight into the strengths and weaknesses of the various techniques. A ten-node network has also been simulated. The results favor delta routing which is most effective in highly interconnected network enviroments.

A New Technique for Displaying Continuous Tone Images on a Bilevel Display

Abstract: A new technique for the display of continuous tone images on a bilevel graphical output device is described. This technique belongs to the general class of algorithms that convey intensity information by the spatial arrangement of lit and unlit cells which are in a one-to-one correspondence with the picture elements (PEL's) in the original image. Besides the rendition of a gray scale in the processed image, the new algorithm also incorporates edge emphasis to increase the legibility of textual information and other areas of high detail. A simple theory of the algorithm, derived from an intensity distribution function, is given, along with the details of an implementation and results obtained using the new technique. The technique is interesting as it requires a nonzero noise component in the image signal being processed in order to generate a gray scale.

Store-and-Forward Buffer Requirements in a Packet Switching Network

Abstract: Previous analytic models for packet switching networks have always assumed infinite storage capacity in store-store-and-forward (S/F) nodes. In this paper, we relax this assumption and present a model for a packet switching network in which each node has a finite pool of S/F buffers. A packet arriving at a node in which all S/F buffers are temporarily filled is discarded. The channel transmission control mechanisms of positive acknowledgment and time-out of packets are included in this model. Individual S/F nodes are analyzed separately as queueing networks with different classes of packets. The single node results are interfaced by imposing a continuity of flow constraint. A heuristic algorithm for determining a balanced assignment of nodal S/F buffer capacities is proposed. Numerical results for the performance of a 19 node network are illustrated.