Free-Space Optical Communications for Next-generation Military Networks

There is a continuing need for increased capacity for military applications, especially in network-centric operational concepts that promote the use of information as fundamental for gaining superiority on the battlefield. As an example, the access to, and distribution of, sensor data is a major tenet of network-centric warfare and yet radio frequency (RF) links will struggle to provide the needed capacity. Free-space optical communications (FSOC) has the potential to meet these emerging military needs by offering dramatic increases in capacity. However, there are many technical challenges al multiple layers of the communications protocol stack. This article describes these challenges and discusses some mitigation approaches to provide a path to realizing this capability on the battlefield     

Mobile small aperture satellite terminals for military communications

The United States Army is currently developing a satellite-based network-centric waveform. Mobile terminals that are small and compact are highly desirable for the military. This article gives an overview of the technical challenges and performance issues when mobile small aperture satellite terminals are connected to a network- centric communication system. The specific issues addressed are regulatory limits on off-axis emissions, performance of small aperture terminals in the presence of adjacent satellite interference, and the effects of motion induced antenna pointing errors on the satellite communication system.     

Dual-mode type algorithms for blind equalization

Adaptive channel equalization accomplished without resorting to a training sequence is known as blind equalization. The Godard algorithm and the generalized Sato algorithm are two widely referenced algorithms for blind equalization of a QAM system. These algorithms exhibit very slow convergence rates when compared to algorithms employed in conventional data-aided equalization schemes. In order to speed up the convergence process, these algorithms may be switched over to a decision-directed equalization scheme once the error level is reasonably low. The authors present a scheme which is capable of operating in two modes: blind equalization mode and a mode similar to the decision-directed equalization mode. In this proposed scheme, the dominant mode of operation changes from the blind equalization mode at higher error levels to the mode similar to the decision-directed equalization mode at lower error levels. Manual switch-over to the decision-directed mode from the blind equalization mode, or vice-versa, is not necessary since transitions between the two modes take place smoothly and automatically     

Effect of Time Diversity on the Forward Link of the DS-CDMA Cellular System

A direct-sequences code division multiple access system has been accepted as a digital cellular standard (IS–95) in North America [1]. This digital cellular standard employs a powerful rate 1/2, constraint length 9, convolutional code in its forward link. It is well-known that in a Rayleigh fading channel the performance of a channel code depends very heavily on the interleaving depth and the relative variations of the channel characteristics. In slow fading channels, since the input symbols to the channel decoder are highly correlated, the bit-error-rate at the output of the channel decoder may be unacceptably high. Interleavers of large dimensions can reduce the correlation of the input signal to the channel decoder at the expense of an intolerable delay. In this paper we examine the performance of the IS–95 system, at the mobile receiver, for different channel fade rates. Also, we present a simple time diversity technique which employs multiple receive antennas. The multiple receive antennas in this case generate a fast fading effect and thus improves the performance of the channel decoder significantly.     

An error-protected speech recognition system for wirelesscommunications

Future wireless multimedia terminals will have a variety of applications that require speech recognition capabilities. We consider a robust distributed speech recognition system where representative parameters of the speech signal are extracted at the wireless terminal and transmitted to a centralized automatic speech recognition (ASR) server. We propose two unequal error protection schemes for the ASR bit stream and demonstrate the satisfactory performance of these schemes for typical wireless cellular channels. In addition, a "soft-feature" error concealment strategy is introduced at the ASR server that uses "soft-outputs" from the channel decoder to compute the marginal distribution of only the reliable features during likelihood computation at the speech recognizer. This soft-feature error concealment technique reduces the ASR error rate by more than a factor of 2.5 for certain channels. Also considered is a channel decoding technique with source information that improves ASR performance     

A code division multiplexing scheme for satellite digital audiobroadcasting

Several systems are being considered for digital audio broadcasting (DAB) and some of these systems will be deployed for commercial use. In this paper, we consider a code division multiplex (CDM) scheme for satellite DAB. For diversity purposes, this system makes use of at least two satellites, and in urban areas, where the satellite signals are shadowed by large structures, a terrestrial network is employed. We present a forward error protection (FEC) scheme that is robust under a variety of channel conditions, especially in cases where the received signal is severely attenuated because of highway underpasses and tunnels. This FEC scheme makes use of time diversity, and this introduces a large delay. Such delays will not affect the performance of broadcast systems; however, the delay will be an issue during tuning when a switch to another channel occurs. We introduce a separate low-delay low-rate tuning channel to facilitate tuning. Extensive simulation results are given to examine the performance of the system, and it is shown that about 70 audio channels, each with a rate of 96 kbit/s, can be satisfactorily supported in a bandwidth of approximately 12.5 MHz in most of the channel conditions considered     

FEC scheme for a TDM-OFDM based satellite radio broadcasting system

There are two licenses, 12.5 MHz each, in the S-band for digital satellite-to-vehicle radio broadcasting in the United States. The potential advantages of such a system is that a motorist can enjoy commercial-free music, digital-quality sound and seamless coast-to-coast coverage. One proposal for the broadcast system is to have two satellites covering the continental USA at any given time. There will also be terrestrial repeaters in cities where the receivers on the vehicles cannot see the two satellites. The channels of these systems are affected by Rician, Rayleigh and flat fading caused by shadowing. This paper proposes a forward error correction (FEC) scheme that is not only robust against fading but also enables a low-delay tuning channel so that minimum tuning delay will occur when a user is switching and selecting programs. The scheme uses multiple source coded bit streams employing different interleaver depths. Large interleavers are used to ensure good decoded signal quality while small interleavers are used to minimize tuning delay. The proposed scheme also ensures that a program will not be interrupted by momentary shadowing frequently experienced by a motorist when, for example, a vehicle goes under a highway overpass. The impact of interleaver design on the real-time end-to-end delay and fading due to shadowing is analyzed. Finally, the channel code performance in Rician and Rayleigh fading channels are also presented     

Convergence analysis of an algorithm for blind equalization

The authors consider a key algorithm for blind equalization and derive expressions for the evolution of the equalizer output error trajectory. The authors develop a model to examine the convergence behavior of this algorithm. Suitable approximations are incorporated into the model to facilitate analysis. The validity of these approximations is demonstrated for a typical communication channel. It is shown that even for a channel for which the assumption of Gaussianity of the equalizer input data may not be very good, the analysis presented predicts the convergence behavior reasonably well     

Sign algorithms for blind equalization and their convergence analysis

In blind equalization a communication channel is adaptively equalized without resorting to the usual training sequence. In this paper we have introduced two new algorithms for blind equalization, which hard limit the equalizer input or the error at the output of the equalizer. These new algorithms are simple to implement and reduce the number of multiplications by approximately one-half. We show by way of simulations that the performance of the algorithm resulting from hardlimiting the error is comparable with the performance of the corresponding algorithm in which the error is not hardlimited. We formulate the new sign-error algorithm as a stochastic minimization of an error functional and demonstrate that the case of zero intersymbol interference corresponds to local minima of this error functional. We also present convergence analysis to predict the output mean square error in both these sign algorithms. Since the algorithms are highly nonlinear we incorporate several simplifying approximations and provide heuristic justifications for the validity of these approximations when the algorithms are operated in a typical practical environment. Computer simulations demonstrate the accuracy of the predicted convergence behavior.This research was supported in part by the Office of Naval Research under Contract N00014-89-J-1538.