The authors reply to the comments by Koppelaar and Tolhuizen (see ibid. vol.46, no.9, p.573, 1998). They state that the optimum choice of the slopes used to perform the encoding process in a projection code always results in maximizing the minimum Hamming distance so that d=2/sup r/.

The operation of the projection code is introduced, and the error rate performance is determined. A new combinatorial generating function technique is used to compute the weight distribution of a cylindrical projection code, which is then employed to obtain upper and lower bounds of the bit error rate of the code. We then apply a state equation technique to compute the generating function of a convolutional projection code. This new generating function technique is needed since the standard state-diagram technique cannot realistically be used to obtain the generating function of a typical convolutional projection code which can have a constraint length of 56.

The author proposes that one can overlay an existing GSM cellular network with a broadband CDMA cellular system utilizing direct-sequence spread spectrum alone with agile notch filters at the base station. Notch filter depths of 35 dB on receive and 20 dB in transmit were found to be adequate. Capacity calculations show that the B-CDMA overlay can approximately quadruple the nominal capacity of GSM, which is 16 channels per sector, by providing an additional 51 voice channels at 16 kbps each. Also, the author has calculated that the B-CDMA mobile transmits on average with 16 dB less power than the GSM mobile, while the CDMA base transmit power per channel would be about 5.5 dB less than the GSM base due to the lower CIR required by the B-CDMA overlay system.>

Packet error probability is the key parameter in evaluating the performance of ARQ schemes, in terms of waiting time to receive a data packet, and system throughput. The authors present an analysis of the packet error probability for two modulation schemes, BFSK and BPSK, when uncoded packets are transmitted over the meteor burst channel using a fixed data rate. They show that the optimum transmission rate, minimizing the waiting time to receive a packet correctly, is independent of the initial SNR of the trail and the modulation used. The results were derived analytically, and are in excellent agreement with exact numerical calculations. They also present simple approximations to the exact calculations, and lower bounds on the packet error rate which are tight in the neighborhood of the optimum transmission rate.>

Because of the continually increasing demand for mobile communications, it has been suggested that personal communication networks (PCNs) be established in the 1850-1990 MHz range. However, that band of frequencies is currently occupied by various microwave signals transmitted by users ranging from utility companies to state and local agencies. In order to allow both sets of users to occupy these frequencies as well as improve the spectral efficiency of this band, a spread-spectrum overlay is proposed, whereby a code-division multiple-access (CDMA) PCN would share the spectral band with the existing narrowband microwave traffic. The results of several field tests which have been designed to demonstrate the feasibility of an overlay of this type are discussed.>

This work presents the results of a theoretical study of the meteor-burst channel which illustrates its time-varying bursty nature. In addition, the authors describe an innovative approach to data transmission over meteor-burst systems, i.e. the use of the feedback adaptive variable rate (FAVR) system which allows the maximum amount of information to be transmitted over each meteor-burst channel. For underdense meteor channels the FAVR performance is compared to an optimum system, i.e. a system capable of changing its bit rate instantaneously to channel conditions so as to maintain a constant SNR in each bit, and to a constant optimum rate/burst system. It is shown that when compared to an optimum rate per burst system, FAVR can result in a throughput increase exceeding a factor of 10. It is also shown that FAVR can approach the optimum system performance, although for practical purposes, it appears that 83-88% of optimum is attainable.>

Code Division Multiple Access (CDMA) is a digital multiple-access technique, whereby each signal has its own unique binary sequence, and all signals share the same spectrum. In the United States, personal communication networks using CDMA, will provide to each user, a separate code, which will change from cell-to-cell. We call this code reuse, just like the frequency reuse employed in the FDMA or TDMA systems. However, there are many more codes, than frequency bands.

The authors analyze the problem of noncoherent FM demodulation of trellis-coded continuous-phase M-ary FSK (frequency-shift keying). The FM demodulation process is divided into two parts, the first being the actual noncoherent FM demodulation and the second being trellis decoding of the data. Upper bounds on the bit error rate as well as the 99% energy bandwidth are determined for the codes under consideration. In particular, the authors consider the trellis codes with rates 1/2 and 2/3 and symmetric and asymmetric signal constellations. Upper bounds on the probability of error are obtained for the symmetric and the optimum asymmetric cases. The optimum asymmetry is one which minimizes the bit error probability. The performance of this system is compared to that of the standard continuous phase modulation techniques employing noncoherent detection.>

An analysis is presented of a fiber-optic random-access network suitable for real-time communication. It is based on pure Aloha access discipline of a hybrid, time-wavelength, code-division-multiplexed channel. The network performance is determined as a function of the signaling parameters, number of active users, and their information rates. It is shown that the network throughput can be significantly increased if users allow a few bit errors per packet. Thus, users have the freedom of trading off throughput for bit error rate and vice-versa.>

M-ary phase-shift keying and differential phase-shift keying (DPSK) on a slow fading Rayleigh channel without diversity is investigated. Expressions for the distribution of the phase angle between a vector with Rayleigh amplitude distribution and a noiseless reference, and between two vectors both with Rayleigh amplitude distribution perturbed by Gaussian noise are obtained.>

We present an efficient search technique which minimizes the computations necessary for estimating the motion in video-sequences by the block matching method. We also discuss the theoretical basis for conducting such a reduced search by our technique. We then present two algorithms which employ the proposed technique for estimating the motion typical of video-conferencing environment. Next, the results of computer simulations on a real video-sequence are included which demonstrate the effectiveness of the proposed technique. Finally, the results of a study of statistical properties of block motion-compensated frame difference signals are also summarized, to assist in future choice of a coding strategy for such signals.

In this paper, we briefly describe the physical characteristics of a meteor-burst (MB) communication channel, and then present the results of an analysis showing the expected throughput and the probability of completing a given message under two different scenarios. In the first scenario, the transmitter is presumed to have knowledge (obtained by appropriate probe signals) of the entire duration of a given MB. In the second, only the starting time of each MB is known. Numerical results based on the analysis are provided, and it is shown what types of delays must be tolerated in order to achieve a given degree of reliability of having the transmitted message correctly received at its destination.

In this paper, the results of comparing the performance of various forward error correction techniques and several modulation formats when used over a nonselective Rayleigh fading channel in the presence of a pulse-burst jammer are presented. Both binary and nonbinary codes are considered, as well as concatenated codes consisting of either a block or a convolutional inner code and a Reed-Solomon outer code. Finally, the use of side information to allow the decoding of both erasures and errors is also analyzed.

It is shown that the set of cycles of a linear feedback shift register with characteristic polynomial (1 + x)nare at most doubly adjacent.

This paper describes a rapid acquisition technique which can be used in direct sequence spread-spectrum systems. It is sequential in nature and is based on a double threshold which defines when a decision can be made. The values of the thresholds change at every examination instant. The procedure makes use of a bound to the partial crosscorrelation function of PN sequences which ensures that only the sequence length needs to be given as far as the PN sequence particulars are concerned. A decision is made only if the probability of error associated with a decision at that point is equal to or better than a prespecified value. Finally, it is shown that most of the computational difficulties associated with the bound as given by a recursive algorithm can be eliminated when using a low-order polynomial approximation to the bound. However, either by using the bound or its polynomial approximation, a significant improvement in the acquisition time is obtained.

Predictors operating on video signals sampled at six times the color subcarrier(6f_{c})have been investigated and compared with predictors operating at three times the color subcarrier(3f_{c}). When the best6f_{c}predictor is used with an adaptive delta modulator (ADM) step size generator, the resulting pictures are similar in quality to3f_{c}predictors operating with a four-level adaptive DPCM quantizer. An ADM coder was implemented in real time. The coder used between 24 and 34 IC's, depending upon the algorithm, and fits on a single 9 in × 6 in PC board. The transmission rate is 21.4 Mbits/s.

This paper describes a one-stage look-ahead algorithm for adaptive delta modulators. The algorithm does not require the calculation of two encoding paths nor does it require the decision circuitry to choose the optimum path.

The paper concerns the study of waiting times at a packet-switched merger node for which the outgoing capacity exceeds or is equal to the sum of the capacities of the incoming channels. The case of constant service times, fixed-length packets, and general independent inputs is considered. A trivial upper bound on the waiting time is derived. Exact results on waiting time are obtained for certain specific network configurations.

Upper bounds on the worst case autocorrelation and the worst case cross correlation of linear PN sequences of the same period are obtained. The value of the worst case correlation curve at some specific points is found and using interpolation methods a smooth curve fitting to the correlation curve is formulated.

Three schemes for course acquisition of spread-spectrum signals are compared: stepped serial search, matched filter, and twolevel. Analytical models and formulas are developed which characterize performance in adverse environments. Comparisons are on the basis of miss probability, mean time to search uncertainty region, and relative complexity.

Tracking requirements for a frequency-hopped spreadspectrum system with a view to application in mobile radio are investigated. Models characterizing Rician fading and multiple access or jamming environments are used to formulate signal detection probabilities. These determine the mean time to loss of lock. Optimized tracking in these environments is then considered.

The Chernoff bound is used to upper bound the average probability of error of a direct sequence spread spectrum communication system under the conditions of multiple-tone interference. A special case is analyzed using a Gaussian approximation to the interference, and this latter result is compared to that predicted by the Chernoff bound. The limitations of the Gaussian approximation are then carefully explained.

The performance of four digital modulation techniques are compared to one another when each is received in the presence of adjacent channel interference. The interfering waveforms that each modulation format sees are identical to the modulation format under consideration. The four types of modulation are QPSK, 8-PSK, 16QASK, and QPR. Upper bounds to the probability of error are derived and used to evaluate the relative merits of the different schemes.

The Chernoff bound is used to estimate the performance of a direct sequence spread spectrum communication system operating over a frequency-selective Rician fading channel. In addition to the degradation caused by the selectivity of the channel, the system is further degraded by the presence of a tone interfering signal. A closed form expression for the bound is presented, and it is seen that at low probabilities of error, the bound appears to be within 1 dB or so of the true value. Numerical results are also presented for typical combinations of system parameters.

The performance of a binary noncoherent frequency-hopped/frequency-shift-keyed system operating in the presence of partial-band tone interference and transmitted over a frequency-selective Rician fading channel is derived. The probability of error for the system is derived both with and without the use of error-correction coding, and the two cases are compared on the basis of equal information rates and equal spread bandwidths.

Spread-spectrum communications, with its inherent interference attenuation capability, has over the years become an increasingly popular technique for use in many different systems. Applications range from antijam systems, to code division multiple access systems, to systems designed to combat multipath. It is the intention of this paper to provide a tutorial treatment of the theory of spread-spectrum communications, including a discussion on the applications referred to above, on the properties of common spreading sequences, and on techniques that can be used for acquisition and tracking.

This paper presents the results of a study in data compression of adaptive delta modulated video signals. The Song mode ADM is first investigated at a sampling rate of 16 Mbits/s and shown not to produce enough redundancy to warrant entropy encoding. We then present a modified adaptive delta modulator algorithm that operates at a sampling rate of 16 Mbits/s and does produce sufficient redundancy to yield a 40-50 percent data compression by using a simple code on 4 bit data blocks. Other techniques such as field interpolation and direct substitution are shown to increase the possible data compression further without noticeable degradation in the two input images used in this investigation. The effects of channel errors in the transmission of packet video over a computer network are considered. A leaky integrator is used to reduce the effects of channel errors in the data bits. We show that the effects of channel errors can be reduced by field interpolating those packets that can be shown to contain errors.

The probability of error of both a 16-ary QASK system and an MSK system, each operating over a Rician fading channel, is considered. The channel model has two key features: short term variations are modeled by making the channel frequency selective, and long term variations are accounted for by allowing the specular component of the received signal to fade according to a Rayleigh density. The improvement gained by using maximal-ratio diversity combining is derived, as well as the degrading effects of an adjacent channel. Finally, a comparison between MSK and 16-ary QASK operating over the same channel is presented, and it is shown that MSK results in a smaller probability of error.

Equations (10) and (16) of the above-named work (ibid., vol. COM-28, Part 11, pp. 1734-1742, Sept. 1980) are corrected.

In this paper we discuss the correlation properties of the output of a binary convolutional encoder. We consider both random statistically independent inputs and first-order Markov inputs. It is shown that for random inputs, the ensemble average of the product of the outputs of a convolutional encoder taken two at a time is always either 0 or 1, and that for almost all good codes the ensemble average is always 0.

The gueast editors attempt to make the arguments: 1) that whereas there are many similarities between military and civil systems, there are also many differences; 2) that thes ources of the differences are iargely summed up with the terms survivability and interoperability; and 3) that broad systems approaches are necessary to achieve the desired goals. The systems approach is rarely described in technical journals. Because of this, the papers invited for this Special Issue feature systems approaches and overviews designed to give the reader some perspective as to what the main issues are in military communications. There is also a sprinkling of techniques-oriented papers, some invited and some submitted, to complete the picture. In keeping with this systems orientation, the Special Issue is divided,i nto four major sections, oriented to the principal categories into which the Defense Department divides its systems for management purposes. The first section contains two papers, both providing architectural overviews over system evolution. The second section is devoted to the category of Long-Haul Communications, with papers addressing various aspects of this category: systems, subsystems, and technology. The third section contains ten papers devoted to Tactical Communications, the name given to the category supporting command and control from the major field comnianders down to the lower echelons of command. The final section containing eight papers treats a variety of technical topics which are neither specifically long-haul nor tactical, but which, although primarily techniques-oriented, address specific military issues.

In this paper, the performances of the Song Voice Adaptive Delta Modulator (SVADM) and the Continuously Variable Slope Delta Modulator (CVSD) in terms of dynamic range, sampling rate and the channel errors are compared. The use of the SVADM and the CVSD in a packet voice system, the algorithms for digital detection of silent periods and the performance of a packet voice system using the SVADM and the CVSD as source encoders are presented. The parameters employed for subjective evaluation of the packet voice system are packet size, silence detection algorithm, bit rate and packet loss rate.

It is desired to determine the performance of a noncoherent FSK frequency-hopped spread spectrum communication system using MARK and SPACE tones which are not necessarily orthogonal, and in addition using an error-correction code to reduce the susceptibility of the system to jammer-induced errors. The overall bandwidth of the system (i.e., the spread bandwidth) will be held constant so that the processing gain of the system will always be adjusted to maintain this constant bandwidth as other factors (e.g., the bandwidth expansion due to the error correction coding) change the underlying baseband bandwidth.

A scheme for coarse acquisition of spread spectrum signals is described. It is characterized by a two-level acquisition process employing a passive correlator in tandem with a bank of active correlators. A threshold exceedance of the passive correlator initiates an interval of active correlation if any of the active correlators is idle. The scheme combines the rapid search capability of passive correlation with the decision reliability of (long) active correlation. An analytical model based on queuing and detection theory results is developed, and optimization of the scheme is treated. Performance in the presence of background noise with noise jamming and/or CW jamming is analyzed. Theoretical performance curves and a discussion of results are presented for each type of jamming.

In most realistic communication systems, be they spread spectrum or conventional systems, some overall bandwidth constraint is usually placed upon the system. In the case of spread spectrum communications, this constraint limits the amount of processing gain that the system can employ. In addition to processing gain, factors such as alphabet size affect the total bandwidth. It is the purpose of this paper to combine these two effects to determine how one should trade off the amount of processing gain one uses relative to the underlying modulation bandwidth when one is operating under the constraint of fixed total system bandwidth.

Summary form only given, as follows. On page 1398 of the above paper (ibid., vol. COM-28,Part II,pp. 1389-1398, Aig. 1980), the Acknowledgment was published incorrectly. The correct Acknowledgment is as follows: "ACKNOWLEDGMENT The authors would like to thank Prof. C. Ziegler for generating the numerical results, and J. Bartow and S. Hall for their helpful suggestions."

The purpose of this paper is to show that signals which are adaptive delta modulation (ADM) encoded can be arithmetically processed directly, without first decoding or converting to pulse code modulation (PCM) format. By operating on the serial DM bit streams, the sum, difference and product can be obtained in PCM and ADM format. Employing a four-term, non-recursive, averaging filter after the processors, we show that, for constant inputs, the signal-to-noise ratio (SNR) of the DM devices is exactly the same as that of their PCM counterparts. Although we have used the Song audio mode ADM [1] in the realization of our arithmetic processors, the designs are general enough to be applied to a large class of digital ADMs. To keep all systems practically realizable, we only employ operations which can be constructed with standard digital hardware, that is, adder, delays, hard-wired scalars and common logic circuits. Consequently, all our ADM devices can be manufactured with large scale integration where the distinct advantage is low cost and high reliability.

Two algorithms for voice encoding are described. One, the modified Abate, is a simplified version that is the type designed for the Space Shuttle. Waveforms of the response of the delta modulators to channel errors are given and performance data that shows the relationship between channel errors and word intelligibility are included. An analytic derivation yielding a comparison between PCM and adaptive delta modulation with respect to channel errors is also given.

This paper describes several adaptive delta modulators designed to meet NASA's potential need to digitally encode video signals. The adaptive delta modulators described are: (1) One-dimensional ADM to encode black-and-white pictures and also transmit digitally encoded voice with no increase in bandwidth (2) An ADM to encode field-sequential color; (3) An ADM to encode color video signals. The units described operate at bit rates that can be adjusted from 6 Mbits/s to 24 Mbits/s. The effects of the bit rate and channel errors on the response of the picture are discussed.

Two networks consisting of single server queues, each with a constant service time, are considered. The external inputs to each network are assumed to follow some general probability distribution. Several interesting equivaleneies that exist between the two networks considered are derived. This leads to the introduction of an important concept in delay decomposition. It is shown that the waiting time experienced by a customer can be decomposed into two basic components called self delay and interference delay.

This paper describes several adaptive delta modulators designed to encode video signals. One- and two-dimensional ADM algorithms are discussed and compared. Results are shown for bit rates of 2 bits/pixel, 1 bit/pixel and 0.5 bits/pixel. Pictures showing the difference between the encodeddecoded pictures and the original pictures are presented. Results are also presented to illustrate the effect of channel errors on the reconstructed picture. A two-dimensional ADM using interframe encoding is also presented. This system operates at the rate of 2 bits/pixel and produces excellant quality pictures when there is little motion. We also describe and illustrate the effect of large amounts of motion on the reconstructed picture.

We investigated the effects of channel errors on an adaptive delta modulator used to encode video signals. The investigation revealed that channel errors caused a permanent shift in the dc level of the delta modulator's estimate. Errors in the step size, on the other hand, were transitory and had no noticeable effect on the received pictures. We then presented three error correction schemes to minimize the visibility of the channel errors in the received picture. The first scheme required the transmitter to periodically send the correct dc level of the estimate to the receiver. The second method employed a leaky integrator, and the third method used line-to-line interpolation.

In February 1975 the Editor-in Cheif announced a new policy, that of restricting papers to be less than 20 pages in length. He is pleased to say that authors have cooperated, and as a result, this Transactions has been able to significantly reduce the number of pages published. Whereas in 1974 this Transactions published 1900 pages, in 1975 we published 1530 pages. The Editorial Board intends to continue to limit papers to 20 pages. By so doing, we hope to continue our policy of publishing every accepted manuscript as it is received, without a backlog.

The author is leaving the Editorship of this Transactions because it is "time for a change." He has been the Editor since 1968. In that time he has expanded the scope of the Transactions into the fields of computer communication and communication electronics. He also encouraged the formation of the two Technical Committees which now represent these areas. He also would like to take this opportunity to thank the large number of people who worked so hard to make his Editorship a success.

An overshoot suppression scheme to improve the performance of the digital Song adaptive delta modulator (DM) for picture transmission is described. This scheme allows for a faster increase in the step sizes than permitted by overshoot and settling time considerations, thus improving rise-time performance. The overshoot suppression (OSS) algorithm used has been verified using computer simulation on a PDP-8. Furthermore, experimental results using computer generated test pictures as well as pictures from a flying-spot scanner show the improvements due to the overshoot suppression scheme. It is also shown that the additional hardware required for the actual implementation of the algorithm is simpler than those encountered in the literature, and gives better signal-tracking accuracy. Upper bounds for the settling time, with and without OSS, are derived showing an improvement with the suppression scheme. Stability conditions are also derived permitting the proper selection of DM parameters.

An all digital phase-locked loop (DPLL) is designed, analyzed, and tested. Three specific configurations are considered, generating first, second, and third order DPLL's; and it is found, using a computer simulation of a noise spike, and verified experimentally, that of these configurations the second-order system is optimum from the standpoint of threshold extension. This substantiates results obtained for analog PLL's.

The theory of operation and the design of an all-digital FM discriminator is presented in this paper. It is shown that the digital discriminator is equivalent to the Foster-Seely demodulator, i.e., a discriminator that converts the FM signal into an AM signal and then recovers the AM by using envelope detection. The output signal-to-noise ratio is found analytically above and below threshold and the results are verified experimentally. The theoretical analysis includes the effect of quantization noise, frequency jitter in the internal oscillator, and thermal noise. The system can be integrated and with the use of readily available emitter-coupled logic (ECL) can operate at intermediate frequencies suitable for many space, industrial, and commercial applications.

The response of a frequency demodulator using feedback (FMFB) to a frequency modulated signal is analyzed. Canonical equations of operation are obtained. Harmonic distortion is calculated for the case of a sinusoidal modulating signal. Intermodulation distortion is calculated assuming a noise-like modulation. Design curves are presented. The special case of harmonic and intermodulation distortion in a discriminator is also presented. It is shown that the results obtained in this paper by treating the discriminator as a degenerate FMFB compare favorably with those obtained by other authors. However, the results presented here do not require digital computation.

A study of source-encoding techniques that afford a reduction of data-transmission rates is made with particular emphasis on the compression of transmission bandwidth requirements of band-limited functions. The feasibility of bandwidth compression through analog signal rooting is investigated. It is found that theNth roots of elements of a certain class of entire functions of exponential type possess contour integrals resembling Fourier transforms, the Cauchy principal values of which are compactly supported on an interval oneNth the size of that of the original function. Exploring this theoretical result it is found that synthetic roots can be generated, which closely approximate theNth roots of a certain class of band-limited signals and possess spectra that are essentially confined to a bandwidth oneNth that of the signal subjected to the rooting operation. A source-encoding algorithm based on this principle is developed that allows the compression of data-transmission requirements for a certain class of band-limited signals. The utility of this algorithm is illustrated by a digital computer simulation of a facsimile telegraphy system. The scanning and recovery of several alphanumeric character images is simulated. Recognizable replicas of the original pattern are retrieved after 6-to-1 bandwidth compression.

An FM/FM system is one in which the signal information frequency modulates a subcarrier, which, in turn, frequency modulates a main carrier for transmission. The detection process then involves two frequency demodulators in sequence. Using Rice's click model for both FM demodulators, we obtain the average click rate and an approximate output spectrum that include the effect of the impulses generated by the carrier demodulator on the subcarrier demodulation at and below threshold. We then focus on the threshold behavior of these systems and the proportioning of their modulation parameters so as to provide certain desired characteristics. We find that due to the filtering action of the second demodulator, the number of output clicks from carrier generated spikes can be orders of magnitude smaller, yet their contribution to the output noise can be much greater than that produced by the first demodulation. In fact, this term is dominant in the thresholding of the subcarrier demodulators except for the most fully loaded multiplex. In the latter case, there is so little deviation available for each channel that the self-clicks produced by the subcarrier demodulation in the absence of carrier spikes become severe. We also find that each channel has a critical carrier modulation index below which the output clicks exceed the carrier generated clicks.

The use of extreme-value theory (EVT) in the detection of a binary signal in additive, but statistically unknown, noise is considered. It is shown that the optimum threshold and the probability of error of the system can be accurately estimated by using EVT to obtain properties of the initial probability density functions on their "tails." Both constant signals and slowly fading signals are considered. In the case of a fading signal, the detector becomes adaptive. Detection of the constant signal, both with and without an initial learning period, is studied by computer simulation.