Continuous sensing and transmission by secondary users (SUs) in cognitive radio (CR) allows efficient use of available spectrum. In this approach, the SU receiver continuously senses the spectrum while it transmits in the same frequency band. Leveraging this advantage, we study the performance of error vector magnitude (EVM) based detection to detect the reappearance of the primary user (PU) signal during ongoing SU data transmission. We exploit the pilot tones that are inherent to many OFDM based standards to measure the EVM. We consider two models of PU namely an unknown deterministic signal and a Gaussian random signal with additive white Gaussian noise (AWGN). We derive an analytical form of the probability density function (PDF) of the EVM based detector based on the exact representation of Chi-square distributions. We analyze the performance of our detector by deriving the type I (false positive) and type II (false negative) error probabilities. Simulation results show that the EVM based approach achieves better performance than energy detection.

In cognitive radio (CR) systems, secondary users (SUs) equipped with a single transceiver are unable to sense and transmit simultaneously. Due to this limitation, most of the models intended to describe spectrum sensing focus on pre‐transmission sensing and consider that primary users (PUs) only change their activity state in the beginning of each SU's operation cycle. This study characterises the more practical scenario which considers the reappearing PU during ongoing SU communication. The authors study a quickest detection scheme that uses an error vector magnitude (EVM) of the received signal. They consider two models of PUs namely an unknown deterministic signal and a Gaussian random signal with additive white Gaussian noise. The probability density function of the EVM test statistic is derived for both cases. The authors develop an exact quickest detection scheme by using a traditional cumulative sum (CUSUM) test. They show that CUSUM EVM outperforms CUSUM ED using theoretical analysis and numerical simulations.

In the communication systems that use a linear modulation scheme for transmission, the fractionally-spaced (FS) samples of the received signal constitute a wide-sense cyclostationary time series. Hence, the standard Fourier transform techniques cannot be used to study the spectral characteristics of the received FS samples or to derive the transfer function (TF) of the corresponding digital minimum mean-square error (MMSE) receiver. In this paper, an analytical expression for the TF of the FS MMSE equalizer is derived, which includes the effects of the continuous-time to discrete-time (C/D) converter used at the receiver front end. Using this TF, the sources of instability of the FS least-mean-square (LMS) algorithm and the effects of the equalizer length and sampling phase on convergence of the LMS algorithm are explained. For stabilization of the FS LMS algorithm, conditions on the front-end C/D converter are provided, such that, when satisfied the LMS algorithm becomes more stable and the learning characteristics of the modified receiver are better than the leaky-FS LMS algorithm. Theoretical results are corroborated by simulations.

The periodic sensing requirements in cognitive radios cause interruptions for secondary user (SU) communication. As a result, the throughput of the secondary system during sensing can be very low. In this paper, we propose a monitoring scheme for Orthogonal Frequency Division Multiplexing (OFDM) based cognitive radio which can monitor the spectrum during ongoing communication and detect the emergence of primary users (PUs). Our scheme permits secondary cognitive radios to perform spectrum monitoring while receiving packets. The proposed method exploits the pilot tones that are inherent to many OFDM based standards to measure the error vector magnitude (EVM) of the group of pilots (GOPs). A step change in the EVM curve signifies the emergence of the PU.

In this study, the authors design a multi‐user ultra‐wideband system for indoor wireless communication. The proposed system is useful in multipath radio reception as the full multipath diversity gain is achieved. The system has multiple transmitters and receivers. They use polyphase spreading waveforms which feature low levels of mutual interference and enable each of the channel impulse responses to be measured free from distortion continuously with each data symbol transmission. A key feature of the proposed system is that transmitted waveforms are used which produce outputs from the receiver cross correlators, which are zero on both sides of the main correlation peak, for each radio path. This zero response lasts for a duration in excess of the expected delay spread of the radio channel enabling full Rake processing of the multipath signal. The effects of the low levels of mutual interference are reduced further by using time and frequency hopping, forward error correction and soft decision decoding.

We consider single-h continuous phase modulation (CPM) systems that use linear prediction receivers with coefficient look up table (LUT) for mitigating the effect of multipath fading. This type of receiver shows superior performance over conventional ones; however, high channel delay spread and fast fading can still lead to a high error floor in its performance. A sub-optimal soft/iterative detection based on the soft output Viterbi algorithm (SOVA) is presented for reducing such error floors while maintaining low complexity. The design is extended with a simple spatial diversity combining scheme and the utilization of multiple coefficient LUTs. The proposed approach allows a significant performance gain to be achieved.

We propose an iterative soft multiuser detection technique for detecting multiple BPSK modulated cochannel signals in Rayleigh flat fading environments. Ant colony optimization is used to produce soft estimates of the transmitted signals. Our attention is focused on the challenging case of overloaded transmission, where there are fewer receive antennas than transmitted cochannel signals. Simulation results show our approach can achieve near maximum likelihood (ML) performance at significantly reduced complexity for critically loaded systems, and it continues to provide good performance under overloaded scenarios.

In this paper, the non-selective Decode-and-Forward (DF) scheme proposed in is analyzed using approximate Gaussian density evolution (DE). Based on the DE analysis, performance is shown to be improved by adaptively changing the transmit code at the relay node in response to the channel conditions. Simulation results show that the proposed adaptive scheme improves performance, particularly when the relay is closer to the destination than to the source. In addition, the adaptive scheme does not increase network overhead.

We investigate the use of the minimum mean-squared- error decision-feedback-equalizer (MMSE-DFE) as a preprocessor to reduce the computational complexity of symbol detection in frequency-selective multiple- input multiple-output (MIMO) channels. The preprocessor delays and focuses the received signal energy which reduces the effects of inter-symbol- interference and multiple-access-interference. This property of the MMSE-DFE preprocessor is exploited to order and to reduce the number of symbol candidates considered by an iterative maximum- likelihood (ML) detector. Simulations show that the proposed detection scheme can achieve lower bit- error-rates than existing reduced-complexity algorithms in overdetermined and underdetermined receive scenarios. Near-optimum performance is achieved when the number of symbol candidates is increased.

In this paper, we propose two solutions to the problem of joint transmit-receive antenna selection in a multiple-input multiple-output (MIMO) cognitive radio (CR) system. Our objective is to maximize CR data rates and satisfy interference constraints at the primary user (PU) receiver(s). In the first we approximate the original non-convex optimization problem using an iterative approach solving a series of smaller convex problems. Second we present a novel, norm-based transmit receive antenna selection technique that simultaneously improves throughput while maintaining the PU interference constraints. We show that this approach yields near optimal results with massive complexity reductions. We make a performance comparison between the proposed approaches and the optimal exhaustive search approach. We provide an analysis of the exhaustive search and relate selection gains to system parameters such as the shadow fading standard deviation, the path loss exponent and the number of PUs per square kilometer. Our results establish that antenna selection is a promising option for future MIMO CR devices in sparse PU environments.

Co-channel interference (CCI) occurs in a wireless receiver when multiple signals are present. The receiver experiences excessive CCI under overload which occurs when there are more signals than receive antennas. This makes separation and estimation of the transmitted data signals difficult. We develop a novel reduced-complexity receiver structure for the separation and symbol detection of multiple co-channel signals in frequency-flat Rayleigh fading channels. Moreover, we show that list-based group-wise processing can achieve better performance at low and moderate signal-to-noise ratios than existing group-wise soft information processing schemes. The receiver is equipped with multiple antennas and developed to work under overload. Its structure consists of a linear preprocessor followed by a nonlinear reduced-complexity symbol detector. The proposed list group-search detection (LGSD) algorithm relies on list feedback and reduces complexity by iteratively searching over groups of transmitted symbols. It outputs a list of likely data symbols which is well suited to further processing using soft input error control decoders. Simulation shows that LGSD can achieve near optimum joint maximum likelihood performance under overload at significantly lower complexity.

We propose a cooperative transmission scheme for two users with one common relay using superposition modulation. It uses distributed Turbo codes (DTCs) for each user. The relay always decodes, then interleaves and re-encodes the decoded data. A new packet is formed by combining the re-encoded packets from both users using superposition modulation. This packet is forwarded to the destination with the average receive SNRs of each user. The destination uses cooperative Turbo decoders to improve performance. Simulation results show the proposed scheme outperforms XOR based schemes and is simple to implement.

We consider a wireless communication system where multiple co-channel users transmit data via a synchronous, frequency-flat Rayleigh fading channel. The receiver employs an antenna array. The paper develops an iterative list multiuser detector for overloaded applications where the number of transmitted signals exceeds the number of receive antennas. The receiver uses a linear preprocessor to reduce co-channel interference followed by a multiuser detector with an iterative groupwise symbol detection algorithm that extracts a list of the most likely user symbols. Simulation results show that the proposed detector provides good complexity-performance trade offs.

This paper investigates the design and performance of a two-user satellite communication system. Each user is independently encoded using a structured Turbo code with identical symbol interleavers. This permits Turbo decoding to be performed using the combined component code trellises, which provides significant gains over independent decoding. The design of the code-matched symbol interleavers is described. Both ideal and imperfect knowledge of the relative phase shift between users is considered. Performance is compared to the single user case and the performance degradation due to a third unknown user is considered. These investigations show that the proposed system performs well in a variety of conditions.

In this comment we position the work of Z. Chi, et al, 2004,2001 in terms of the prior literature and note that many of its results are subsumed by previous papers.

Space-time trellis coding is an established diversity technique that reduces the effects of multipath fading over wireless communication channels. Here, we consider high rate space-time trellis codes (STTC) with continuous phase frequency shift keying (CPFSK). We present optimized rate-2/3 STTC implemented with 3 transmit antennas. These codes provide system throughputs of 4 and 6 bits per channel use with 4-ary and 8-ary CPFSK respectively. Simulated error rate performance of the optimized codes with receive diversity is presented. We show that although the schemes do not achieve full transmit diversity, they provide excellent coding gains compared to full rank schemes that have equivalent throughput, but higher order modulations and greater complexity.

An iterative approach is developed for decoding space time trellis codes (STTCs) in frequency selective multiple input multiple output (MIMO) channels. The method uses the partitioned Viterbi algorithm (PVA) as an equaliser for the MIMO system. It is used iteratively with a STTC decoder that provides soft outputs using the soft output Viterbi algorithm (SOVA). It is shown through simulation that the iterative procedure achieves a performance within 2.0 dB of maximum likelihood (ML) decoding, at a FER of 10-2, but suffers a small diversity loss. It is also shown that the complexity of the iterative approach is lower than that of ML decoding.

In this paper, a frequency domain (FD) receiver architecture implemented using estimated channel parameters is derived for broadband single carrier modulations. Co-channel and inter-symbol interferences are compensated by a minimum mean squared error based integrated space-frequency-equalizer (SFE) using the estimated parameters. The integrated SFE in the FD consists of coupled FIR structures, that are jointly optimized by maximizing the desired signal to interference plus noise ratio. We develop analytical expressions and present simulation results for the integrated SFE. Simulation results demonstrate that excellent receiver performance is achieved even for channels with large ISI spans. The proposed SC-SFE outperforms previous layered space frequency (LSF) receivers with imperfect channel knowledge. Further, the parallel receiver architecture yields equal diversity gains to all data streams without the error propagation that is common to most LSF schemes.

To date most frequency-domain (FD) turbo equalization schemes assume ideal channel state information (CSI) is available. In this paper, a system combining FD turbo linear equalization with time-domain channel estimation is developed and evaluated for single-carrier modulation formats. The effect of estimated CSI on the equalizer form is shown. Performance results employing convolutionally encoded QPSK and 16-QAM transmissions show the efficacy of the proposed system and its capability to operate in different wireless scenarios.

An adaptive equalizer structure based on a state space formulation and maximum likelihood sequence estimation (MLSE) is developed for a time-varying, frequency-selective channel. Differential phase shift keying (DPSK) is assumed at the transmitter and effective differential decoding is performed at the receiver. The standard models of a time-varying linear channel and the Karhunen-Loeve (KL) expansion underpin the receiver structure. Analytical and simulation results for the receiver are shown. The resulting receiver is a per-survivor structure.

Abstract — We present a space-time coded system which achieves high throughput and good performance using low-complexity detection and decoding. We focus on a Rate 2 quasi-orthogonal space-time block code structure which enables us to achieve an overall throughput of 5.6 bits/symbol period with good performance and relatively simple decoding using iterative parallel interference cancellation. We show that this can be achieved through the use of a bit-mapped coded modulation (BMCM) structure using parallel short low density parity check codes. The proposed system is shown to perform well on flat Rayleigh fading channels with a wide range of normalized fade rates, and to be robust to channel estimation errors. A comparison with bit-interleaved coded modulation (BICM) is also provided. Index Terms — Error correcting coding, iterative methods, low density parity check codes, MIMO systems, space-time coding. I.

In this paper we study the normalization of mathematical models of linear vector channels. The ratio of the received energy to the transmitted energy is the energy gain of the channel. All physical systems follow an energy conservation law which implies that the energy gain of the channel is less than or equal to one. The major approaches for normalization include setting of either the average energy gain or the peak energy gain to unity. The peak energy gain of many mathematical fading models is infinite and those models cannot be normalized by the peak energy gain. We propose a new approach to normalization where the mathematical channel model is guaranteed to represent a physical system with a predefined probability. We show that the proper normalization of the mathematical model and the selection of the correct performance measure are of paramount importance in comparative performance analysis of adaptive transmission systems.

This paper describes the concept, architecture and development of a real time, cognitive radio testbed for physical layer, interoperability and spectrum sensing experiments. It is based on a reconfigured wireless 4-transmit 4-receiver multiple input and multiple output (MIMO) smart antenna software radio test system (SASRATS) platform using the the Xilinx University Program (XUP) Virtex-II Pro development boards as the heart of the system. We discuss various cognitive experiments in the context of public safety applications and show that cyclo- stationarity can be used for spectrum sensing and recognition applications.

A reduced complexity multiple-input multiple-output (MIMO) channel estimator known as the polynomial-predictor-based vector generalized least mean squares (VGLMS) estimator is developed. It is a simplification of a previously developed polynomial-predictor-based vector generalized recursive least squares (VGRLS) estimator, achieved by replacing the online recursive computation of the 'intermediate' matrix by an offline pre-computed matrix. Similar to the VGRLS estimator, it is able to operate in Rayleigh or Rician fading environments without reconfiguration of the state transition matrix to accommodate the non-random mean components. It is seen to offer a trade-off between reduced complexity channel estimation and good system performance.

Excessive co-channel interference (CCI) occurs in overloaded wireless environments where the number of transmitted signals exceeds the number of available receive antennas. We present a list-based multiuser detection scheme that effectively mitigates CCI in overloaded receivers with a circular array. The algorithm is named parallel symbol detection with reduced complexity interference estimation (PSD-RCIE). It employs an iterative parallel detector with list feedback of the best symbol estimates. Simulations show that our algorithm achieves near joint-maximum likelihood detection at a fraction of the complexity. Its good complexity-performance trade-off together with its parallel structure make PSD-RCIE well suited for practical implementation.

Lattice coding of orthogonal frequency-division multiplexing (OFDM) systems is considered. Mapping of multilevel construction lattices to OFDM blocks is shown, and a methodology for probabilistic analysis of multistage generalized minimum distance (GMD) decoding of the received OFDM blocks is shown. As a case study, transmission of points from a 128-dimensional Barnes-Wall lattice is considered. Tight approximations to the system-error rate are obtained and verified by simulation. It appears that GMD decoding of lattice-encoded OFDM provides high coding gain at low complexity

This paper studies optical communications using subcarrier phase-shift keying (PSK) intensity modulation through atmospheric turbulence channels. The bit error rate (BER) is derived for optical communication systems employing with 00K or subcarrier PSK intensity modulation. It is shown that a${\rm BER}=10^{-6}$and a scintillion level$\sigma=0.1$, an optical communication system employing subcarrier BPSK is 3 dB better than a comparable system using fixed threshold 00K. When$\sigma=0.2$, an optical communication system employing subcarrier BPSK achieves a${\rm BER}=10^{-6}$at SNR 13.7 dB, while the BER of a comparable system employing 00K can never be less than${\hbox{10}}^{-4}$. Convolutional codes are discussed for optical communication through atmospheric turbulence channels. Interleaving is employed to overcome memory effect in atmospheric turbulence channels. An upper bound on BER is derived for optical communication systems employing convolution codes and subcarrier BPSK modulation.

This paper studies optical communications using subcarrier phase shift keying (PSK) intensity modulation through atmospheric turbulence channels. The bit error rate (BER) is derived for optical communication systems employing either on/off key (OOK) or subcarrier PSK intensity modulation. It is shown that at${\hbox{BER}} =10^{-6}$and a scintillation level of$\sigma = 0.1$, an optical communication system employing subcarrier BPSK is 3 dB better than a comparable system using fixed-threshold OOK. When$\sigma =0.2$, an optical communication system employing subcarrier BPSK achieves a${\hbox{BER}} = 10^{-6}$at${\hbox{SNR}} = 13.7$dB, while the BER of a comparable system employing OOK can never be less than$10^{-4}$. Convolutional codes are discussed for optical communication through atmospheric turbulence channels. Interleaving is employed to overcome memory effect in atmospheric turbulence channels. An upper bound on BER is derived for optical communication systems employing convolutional codes and subcarrier BPSK modulation.

We develop space-time trellis coded (STC) schemes using continuous-phase modulation (CPM). We employ the Rimoldi model of CPM to create a decomposed model of STC-CPM. The decomposition separates the coding from the modulation. The space-time encoding and the inherent CPM encoding is combined into a single trellis encoder on the ring of integers modulo-. This is followed by a bank of memoryless modulators. The model allows the search for good space-time codes to take into account the inherent encoding of the modulation.

For a power limited orthogonal frequency-division multiplexing (OFDM) system transmitting a large number N of subcarriers over a Rayleigh-fading channel, the distribution of the instantaneous capacity is shown to be approximately Gaussian. The mean and variance of the approximating distribution are derived. It is also shown that, in the limit as Nrarrinfin, the capacity approaches a constant value equal to the capacity of the infinite-bandwidth Gaussian channel

This paper describes the concept, architecture, development and demonstration of a real time, channel estimator system on a Xilinx Virtex 2 Pro field programmable gate array for a 4-transmit 4-receiver multiple input and multiple output (MIMO) wireless test platform. It is designed and developed for research into receiver diversity and MIMO wireless systems. Hardware, firmware, use of the Xilinx Core Generator Intellectual Property modules and experimental verification of the channel estimator are discussed

In this paper we discuss the proper normalization of a fading channel model. Physically, radio channels correspond to passive circuits and follow the energy conservation law. The ratio of the received energy to the transmitted energy is the energy gain of the channel. The representative energy gain is defined as the average energy gain for a signal that is uniformly distributed in time, frequency and space. The major approaches for normalization include setting of either the average representative energy gain or the peak energy gain to unity. The peak energy gain of many fading models including Rayleigh fading is infinite, which is obviously impossible in a passive system where the peak energy gain should be less than or equal to unity. Our aim is to show that it is due to the normalization that in some cases the performance in a fading channel is better than in a nonfading channel.

Given digital transmission over two correlated Rician fading channels, we present a readily calculable expression for a tight approximation to the correlation between the probabilities of symbol error on each channel. For the Rayleigh fading channel the expression is exact. This correlation expression has applications in error performance analysis of fading systems, particularly multichannel systems and Markov modelling of error events. As an example we use these expressions to bound the block error rate of an OFDM system

The use of multiple-input multiple-output (MIMO) configurations in wireless systems is becoming increasingly popular, due to the potential capacity enhancing properties. The use of such configurations, requires deployment of multiple RF chains and results in an increase in cost and complexity. This paper focuses on transmitter design, where cost is often dominated by the need to utilise linearized power amplifiers. Thus, the effect of reducing the number of active transmit antennae through selection can yield a significant cost advantage. Three criteria for transmit antenna selection are evaluated and characterised through the use of real channel data in the UHF band. The criteria are based only on the channel gains and have relatively low complexity; thus are suitable for practical purposes. This study examines the performance of the tested selection algorithms under a variety of operating conditions. The paper also considers the issue of delayed switch time due to reverse-link communication latency

In this paper, we focus on an advanced space-time system using a rate 2 quasi-orthogonal space-time block code which enables us to achieve very good performance for an overall throughput of up to 4.3 bits/s/Hz. This is done through the use of a bit-mapped coded modulation structure using short low density parity check component codes. At the receiver, parallel interference cancellation (PIC) and belief propagation (BP) decoding are employed. Iterative decoding is performed between the PIC and BP decoding stages. Given a fixed total number of decoding iterations, we investigate the frequency of PIC updates required to achieve good performance

This paper studies the performance of ultra-wideband (UWB) radio communications systems employing phase shift keying (PSK) modulation and fully saturated power amplifiers through additive white Gaussian noise (AWGN) channel or Rayleigh fading channel. An impulse response method is provided to analyze the signal-to-noise ratio (SNR) degradation. Both convolutional code and turbo code are employed with the Viterbi decoding and the iterative decoding, respectively. The decoding algorithms are modified for the Rayleigh fading channel. Near optimal bit error rate (BER) performance is achieved when employing either convolutional code or turbo code. The simulation results match the derivation. Therefore, fully saturated power amplifiers can be employed in UWB PSK systems to significantly reduce radio cost, simplify radio circuit, and increase battery life for portable terminals.

We propose a simple class of encoding/decoding techniques that can be used to improve the information outage rate of certain multiple-input-multiple-output (MIMO) systems. Gains in outage rate can be achieved at low to moderate signal-to-noise ratios in MIMO systems that have fewer receive antennas than transmit antennas. This performance improvement is due to extra "virtual" receive antennas that are created with low-complexity signal processing. A simple space-time block code system using virtual receive antennas is also investigated

To date there has been little work done on multilevel codes for the space-time environment. In this paper we develop multi-dimensional space-time multilevel codes (ST-MLCs). Several construction methods are proposed, including a coset code approach. The proposed space-time multilevel encoders partition a 2Nt-dimensional signalling space, which spans all Nttransmit antennas. The multi-dimensional partitioning can be designed to reduce the complexity of detection/decoding. We develop a space-time multistage decoder for the proposed ST-MLCs. It allows the complexity of soft decision decoding to be significantly reduced compared to a single level approach. In addition, significant performance gains over a single level approach are obtained

Energy is a basic resource in digital transmission links. Physically, radio channels correspond to passive circuits and most of the transmitted energy is lost in the channel. Two alternative approaches are used for performance measurements in terms of energy. Either the average transmitted or received energy per bit is used, both usually normalized by the receiver noise spectral density. This leads to the average transmitted or received signal-to-noise ratio (SNR) per bit, respectively. However, the transmitted energy is the basic system resource. The average energy gain of a channel depends on the transmitted signal. For convenience, the transmitted SNR referred to the receiver is defined to be the product of the transmitted SNR and the representative energy gain, which is defined as the average energy gain of a signal that is uniformly distributed in all dimensions: time, frequency and space. An explicit relationship between the transmitted and received SNR's using the covariance concept is derived. Limitations of the use of different SNR definitions are summarized

A space-time (ST) trellis encoded scheme with M-ary continuous phase frequency shift keying with a modulation index of 1/M (M-CPFSK) is presented. A ST-M-CPFSK system model that incorporates the Rimoldi decomposition of CPFSK is developed. The Rimoldi decomposition separates the non-memory component of CPFSK from the memory component. The latter can be represented as a continuous phase encoder (CPE) and implemented with a ring convolutional encoder. The ST codes used here are linear ring convolutional codes. This allows the ST encoder and the CPE to be combined into one encoder, which can be represented by a single trellis and is easily implemented using delay elements, modulo-M adders and scalar multipliers.

The iterative decoding structure and component maximum a posteriori decoders used for decoding binary concatenated codes can be extended to the nonbinary domain. This paper considers turbo codes over nonbinary rings, specifically ternary, quaternary, penternary, hexernary, and octernary codes. The best rate-1/2 component codes are determined using a practical search algorithm. The performance of the resulting rate-1/3 turbo codes on an additive white Gaussian noise channel using q-ary phase-shift keying modulation is given.

We consider a joint detection approach for cancellation of co-channel interference in time-division multiple access (TDMA) mobile communications systems like GSM/EDGE (enhanced data rates for GSM evolution). Concepts from reduced-state equalization of frequency-selective multiple-input multiple-output (MIMO) channels are applied together with prefiltering. A novel efficient prefilter computation algorithm is presented. Simulation results demonstrate the high performance of the proposed receiver for GSM/EDGE applications.

This paper describes the high-throughput error correcting space-time block codes. To design STBCs, systematic binary and nonbinary Reed-Solomon (RS) and Bose-Chaudhuri-Hocquenghem (BCH) ECCs, embedded error correction codes (ECCs) are considered. The resulting STBCs provide significantly more coding gain and throughputs with the same number of transmit and receive antennas. The cost is the increased decoding complexity. The increased throughput is used to concatenate an external ECC with the STBC for providing a good performance.

This work covers the experimental verification of 2 space-time (ST) algorithms on the smart antenna software radio test system (SASRATS) platform . This work also covers the concept and architecture of SASRATS, which was designed and developed to facilitate the field testing of ST processing architectures and algorithms. The paper covers the experimental verification of two robust ST algorithms developed by the authors for signal enumeration and channel order estimation. Simulation results are also presented and compared with the experimental results obtained on the SASRATS platform.

This letter investigates the influence of transmit and receive filtering on the design and associated performance of wireless communication systems. Based on derivations using power-series models of time- or frequency-selective channels, we present pulse-shaping filters which can be matched to the characteristics of the channel. The choice of filter parameters allows some degree of control over the received signal. The peak-to-average-power-ratio (PAPR) requirements in the transmitter are determined. Orthogonal frequency-division multiplexing (OFDM) represents an alternative method of achieving some of the stated objectives of the pulse-shaping methods developed. Consideration is given to PAPR, bandwidth efficiency, receiver complexity, and performance in terms of error probability for the derived pulses and related forms of OFDM.

This paper describes a new approach to list-based soft-input soft-output (SISO) decoding based on order-i reprocessing. Approximations to both the log-maximum a posteriori (MAP) and max-log-MAP algorithms are developed. Additional decoding steps are proposed to correct common types of errors remaining after iterative decoding. These steps can significantly improve performance at low bit-error rates in later iterations. The proposed algorithms offer a wide range of complexity versus performance tradeoffs, which are explored through Monte Carlo simulations of product code decodings. The algorithms improve performance over previous approaches.

The box and match decoding algorithm (BMA) significantly reduces the computational complexity of the ordered statistic decoding algorithm at the expense of increased memory requirements. A soft-input/soft-output version of the BMA is developed. Additional complexity-reduction techniques are also described.

This paper presents a polynomial-based generalized recursive least squares channel estimator for the dispersive Rayleigh fading channel. The estimator is formulated using a state transition matrix based on polynomial prediction instead of autoregressive modelling, and does not require channel statistics to operate. Analytical and simulated mean square deviation results for the tracking performance of the estimator are presented. The results show that by selecting appropriate values of the "forget factor" and polynomial order, the estimator performs almost as well as the optimal Kalman estimator even in fast fading.

For many existing blind channel estimation algorithms, accurate estimates of the channel order are a prerequisite. A novel scheme for blind channel order estimation using exclusively second-order statistics of the received signal is proposed, which can be embedded in channel estimation and is based on the examination of an indicator function constructed from initial channel estimates with overestimated order. It is demonstrated that significantly better order estimates can be obtained than by information-theoretic criteria like the minimum description length criterion.

This letter presents a simple polynomial predictor-based sequence detector for the Rayleigh nonselective fading channel. Unlike the polynomial predictor-based sequence detector proposed by Borah and Hart, the new receiver is not restricted to constant envelope modulation schemes. Analytical and simulated results are presented. In some instances, the proposed receiver performs within 6 dB of the equivalent maximum-likelihood sequence estimation receiver.

Previous work on concatenated single parity-check codes has yielded exceptionally good performance despite, or perhaps because of, their weak algebraic structure. In this article, maximum a posteriori single parity-check decoders are applied to the decoding of systematic binary algebraic block codes. Results for a range of Hamming codes show good performance compared to soft-decision brute force (maximum-likelihood) and algebraic decoding. The decoding complexity of the proposed technique grows only linearly with increasing block length.

Single parity-check (SPC) codes are applied in both parallel and serial concatenated structures to produce high-performance coding schemes. The number of concatenations or stages, M, is increased to improve system performance at moderate-to-low bit-error rates without changing the overall code parameters (namely, code rate and code block length). Analytical bounds are presented to estimate the performance at high signal-to-noise ratios. The SPC concatenated codes are considered with binary phase-shift keying and with 16-quadrature amplitude modulation bit-interleaved coded modulation on the additive white Gaussian noise channel and the independent Rayleigh fading channel. Simulations show that the four-stage serial or parallel concatenated SPC codes can, respectively, outperform or perform as well as 16-state turbo codes. Furthermore, decoding complexity is approximately 9-10 times less complex than that of 16-state turbo codes. The convergence behavior of both serial and parallel concatenated SPC codes is also discussed.

This article investigates the asymptotic performance of single parity-check (SPC) product codes (PCs) from a decoding point of view. Specifically, the probability of bit error is bounded before and after the decoding of each dimension, similar to the analysis of "iterated codes" by Elias (1954). It is shown that the asymptotic probability of bit error can be driven to zero as the number of dimensions, and hence the block length, increases at signal-to-noise ratios (SNRs) within 2 dB of capacity over the additive white Gaussian noise (AWGN) channel.

This paper describes an approach to combined equalization and decoding using iterative techniques and decision feedback equalization. We show that performance can be improved by using a bank of feedback filters, whose inputs are taken from the output of a list decoder. The terrestrial line-of-sight microwave radio channel is considered in this paper.

A generalised performance bounding technique for BICM systems is presented. We show that, in the Rayleigh fading channel, it is possible to estimate accurately the error performance of coded BICM systems employing large signal constellations, by using only the error performance of a BPSK system employing the same code in the same channel. Our method generalizes the work of G. Caire et al. (see IEEE Trans. on Information Theory, vol.44, no.3, p.927-46, 1998) to all SNR values and is independent of the channel coding scheme and the code rate. The work here implies that, in the Rayleigh fading channel, analytical bounds for any coding scheme derived for the BPSK system can be easily applied to BICM systems with larger signal constellations. Simulation results are presented for the Rayleigh fading channel to verify the analysis.

The paper "Synchronous Communications," originally published in this journal in December 1956 by John P. Costas, has had a profound effect on modem digital communications. At the time of its publication, virtually all communications were analog in nature and, in fact, the stated goal of the paper was to demonstrate an amplitude-modulated (AM) system that could provide significantly improved performance for analog signaling and to dispel the notion that single sideband (SSB) had inherent performance advantages over properly constructed AM signaling systems. This paper is particularly significant because it was the first to demonstrate that carrier phase could be reliably recovered from the received signal using a structure that has come to be known as the Costas Loop.

This article develops an alternative adaptive iterative Chase (1972) based decoding algorithm for block turbo/product codes. The decoder considers only a small subset of codewords, so that estimates of the extrinsic information are required in some cases. This article develops such an estimate based on code distance properties.

This paper develops a robust method to enumerate the incident signals impinging on a uniform but variable size linear array independent of their extent of correlation in a Rayleigh flat fading channel environment. The method also optimizes by minimizing the number of antennas to the number of signals and adapts continuously to maintain optimum performance in a mobile environment where users (signals) come and go. The technique is a modification of the matrix decomposition method of J.H. Cozzens and M.J. Sousa (see IEEE Transactions on Signal Processing, vol.42, no.2, p.304-17, 1994). A new set of stability, stopping and adaptive control criteria is presented. An algorithm is formulated with simulation and field results presented.

The whitened matched filter (WMF) and the finite-length feed-forward filter (FFF) of a minimum mean-squared error decision feedback equalizer are compared for use as digital prefilters in a multiple-input, multiple-output (MIMO) receiver front-end. For systems that are single-input, single-output (SISO), existing literature has shown the FFF approaches the WMF in the limit as signal-to-noise ratio (SNR) and FFF length go to infinity. In this paper, simulation results are given which show the same appears true in MIMO systems.

In this paper, single parity check (SPC) codes are applied in a parallel concatenated structure to produce a high performance coding scheme with very low complexity. We show a simple yet elegant way of improving system performance at medium-low bit error rates (BER), without changing the overall code parameters (code rate, code block length). We compare its BER performance with other coding schemes. Multiple parallel concatenated SPC (M-PC-SPC) codes are capable of outperforming not only the best 64-state convolutional coding schemes, but also the 16-state turbo codes. For bit interleaved coded modulation (BICM) schemes, the 4 parallel concatenated SPC code is about 10 times less complex than the 16-state turbo codes, and yet with only 5 iterations, it still performs better than the turbo code iterated 6 times in the fading and AWGN channel. Combined with multilevel modulation schemes, this high performance scheme is very suitable for practical systems due to its flexible code parameters, and low complexity.

This letter extends the theory of the matched filter bound for fading dispersive channels. We generalize to arbitrary fading, as well as delay, properties of the channel. The channel model is a standard time varying linear filter and the calculation of the matched filter bound is based on a Karhunen-Loeve expansion of the received pulse spectrum when an isolated pulse is transmitted. Results are presented illustrating the various diversity effects. We conclude that in addition to the effect on probability of bit error of implicit delay diversity there is the less well known effect of implicit Doppler diversity in fast fading. The effect on probability of bit error of pulse shaping and explicit diversity is also illustrated.

In this paper, we derive pulses that yield Nyquist or near Nyquist properties when convolved with the initial terms of the series model of a slow fading dispersive channel. For the f-power series these are the so called powers of Nyquist pulses. For a Karhunen-Loeve series, model pulses are found that maximize the ratio of the mean peak-sample energy to mean residual intersymbol interference energy for a given bandwidth expansion factor. Under certain conditions, the approach significantly lessens the need for, or complexity of, equalization. Other benefits attributable to the use of the derived pulses, including those related to timing recovery, are discussed.

We describe an approach to combined equalization and decoding of multilevel block-coded modulation. This approach has better performance than conventional, concatenated equalization and decoding. The proposed structure uses a simple iterative scheme to decode and equalize multilevel block-coded modulation based on decision feedback. We show via computer simulation that the combined scheme gives a performance gain of up to 0.8 dB at a bit-error rate of 10/sup -4/.

Blind equalization attempts to remove the interference caused by a communication channel without using any known training sequences. Blind equalizers may be implemented with linear prediction-error filters (PEFs). For many practical channel types, a suitable delay at the output of the equalizer allows for achieving a small estimation error. The delay cannot be controlled with one-step predictors. Consequently, multistep PEF-based algorithms have been suggested as a solution to the problem. The derivation of the existing algorithms is based on the assumption of a noiseless channel, which results in zero-forcing equalization. We consider the effects of additive noise at the output of the multistep PEF. Analytical error bounds for two PEF-based blind equalizers in the presence of noise are derived. The obtained results are verified with simulations. The effect of energy concentration in the channel impulse response on the error bound is also addressed.

This paper develops an approach to iterative multistage decoding of multilevel codes. This involves passing reliability information to previous and subsequent decoders instead of only hard decisions to subsequent decoders. The paper also develops an adaptive version of the suboptimal soft output decoding algorithm of Picart and Pyndiah (1996). This adaptive algorithm provides a gain of approximately 0.24 dB at a bit error rate (BER) of 10/sup -5/ after four iterations and approximately 0.43 dB after ten iterations over the algorithm of Picart et al. If the adaptive algorithm is used in conjunction with iterative multistage decoding then a gain of approximately 0.62 dB is obtained at a BER of 10/sup -5/ after four iterations and approximately 0.9 dB after ten iterations over the algorithm of Picart et al.

To increase wireless system capacity using co-channel signals and multiple receiver antennas, we develop the partitioned Viterbi algorithm (PVA). The PVA estimation complexity increases linearly with each additional co-channel signal rather than exponentially as it does with joint maximum-likelihood sequence estimation (MLSE). The estimation problem involves multiple signals simultaneously transmitted and observed through slow-fading, frequency-selective channels. Although transmission is assumed to be in bursts according to a time-division multiple-access scheme, more than one signal can occupy the same time and frequency slot (these signals are referred to as "co-channel" signals). Separation and estimation of the symbol bursts is accomplished by exploiting channel differences, PVA estimation consists of a set of Viterbi detectors, one per signal, that operate in parallel with cross-coupling to allow approximate interference cancellation by means of tentative decisions. The forward filter of a decision feedback equalizer (DFE) is used to "prefilter" received signals prior to PVA estimation. Prefiltering delays the energy of interfering signals so that tentative decisions become reliable enough to use. Simulation results show PVA performance remains near-optimal with respect to the performance of joint MLSE.

Turbo-codes are decoded using iterative decoding algorithms with somewhat elusive convergence properties. In this paper, the modes of convergence observed in an extensive series of turbo-code simulations are categorized by examining the bit convergence behavior for each frame. Based on these results, a new method for decoder termination based on average log-likelihood ratios is presented and compared with other methods. As an application, a selective repeat automatic repeat request (ARQ) system is considered.

This paper extends earlier work of the authors to the multipath fading channel. For a normalized Doppler spread of f/sub D/T=0.005 we show that a reduced state sequence estimation reduced-complexity receiver with K/sub s/=2 users per state and a processing gain of N/sub s/=127 performs only 0.5 dB worse than the single-user receiver for a bit-error rate of 10/sup -4/.

This paper presents a simple polynomial predictor based sequence detector for the nonselective fading channel. Although the estimator used in the receiver is mathematically similar to the one presented by Borah and Hart (see IEEE Trans. Commun., vol.47, p.862-73, 1999), the implementation is much simpler. Instead of requiring matrix operations and recursions, the proposed estimator is simply a bank of FIR filters. Unlike the polynomial predictor based sequence detector (PPSD) of Borah and Hart (see IEEE Trans. Commun., vol.47, p.360-64, 1999), it is not restricted to constant envelope modulation schemes.

This paper develops an adaptive approach to iterative decoding of block turbo codes. The method developed in this paper allows the receiver to adapt to the conditions of the channel, the codes and the modulation scheme. The approach yields improved performance especially in early iterations, for a very small increase in complexity.

In this paper, MAP decoders for single parity check (SPC) codes are applied to the decoding of systematic binary algebraic block codes. We compare the performance of this decoding technique to hard decision syndrome decoding and soft decision brute force decoding. Simulation results are presented for the (7,4,3) and (15,11,3) Hamming codes in the AWGN and Rayleigh fading channels. Using a product (7,4,3) Hamming code, the proposed scheme is also compared to the dual code technique as described Berrou and Glavieux (see IEEE Trans. Communications, vol.44, no.10, p.1261-71, 1996). Although the BER performance is slightly inferior, the proposed scheme is much less complex to implement.

This paper describes an approach to combining multilevel block coded modulation with decision feedback equalization. We show that this approach has better performance than independent equalization and soft-decision decoding. The proposed structure uses a simple iterative scheme to decode and equalize multilevel block coded modulation based on decision feedback. The system performance is evaluated via computer simulation. We show that the combined decoding and equalization scheme gives a performance gain of up to 0.9 dB at a bit error rate of 10/sup -4/ over conventional, concatenated equalization and decoding.

Single parity check (SPC) codes are applied in a serial concatenation structure to produce a high performance coding scheme with very low complexity. We show a simple yet elegant way of improving the system performance at low bit error rates (BER), without changing the overall code parameters (code rate, code block length). We compare its BER performance to the best 64-state convolutional codes, and the 16-state turbo codes. Comparisons show that its performance is only slightly poorer. Despite this inferior performance, the scheme is very suitable for practical implementation due to its flexible code parameters. We conclude by showing that its decoding complexity is more than 1 order of magnitude less than that of a 16-state turbo code.

Decompositions of fading dispersive channels find utility in channel modeling and in the estimation of channel state information in a receiver. This letter considers decompositions based on the f-power series and Karhunen-Loeve expansions and investigates their relative ability to model selected channels. Development of such models should enable the design of reduced complexity receiver structures.

A model is developed for finding codes for differentially demodulated continuous phase frequency-shift keying with modulation index h=1/M, where M is the size of the symbol alphabet. The results of codes searches show that there is very little loss in squared Euclidean distance when comparing differential codes with coherent codes.

Blind equalization of a communication channel using a prediction-based lattice blind equalizer (LBE) is considered. Second order cyclostationary statistics and a single-input multiple-output model arising from fractional sampling of the received data are used. The performance of the LBE algorithm is studied in extensive simulations where commonly used example channels are employed. Convergence in the mean square error (MSE) and symbol error rate (SER) as well as the number of symbols required to open the eye are studied at different SNRs. Robustness in the face of channel order mismatch and channels with common subchannel zeros is considered. The simulation results are compared to the results obtained by the fractionally spaced constant modulus algorithm, the cyclic-RLS algorithm and the subspace method by Moulines et al. (see IEEE Trans. Signal Proc., vol.43, no.2, p.516-25, 1995).

This paper presents a simple DPSK receiver for the frequency-selective fading channel. The channel is modelled as an f-power series truncated to the first two terms. Two time invariant receiver filters are used to remove the channel induced ISI and to separate the two implicit diversity branches. The received samples from each branch are then differentially decoded and combined. Analytical and simulation results for binary DPSK show that this receiver out performs the conventional receiver using matched filtering and a product demodulator.

Linearly time-varying fading models are used to investigate both noncoherent detection of FSK signals and differential detection of PSK signals over frequency-flat fading channels. The structure of the optimal FSK and DPSK detectors is derived and an analytical technique is proposed to compute their error performance in fast fading.

Linearly time-varying fading models are used to investigate differential detection of PSK signals transmitted over frequency-flat fading channels. The structure of the optimal differential PSK detector is derived and an analytical technique is developed to compute its error performance. Results obtained by computer simulation are in excellent agreement with the analytical predictions.

Linearly time-varying fading models are used to investigate the differential detection of PSK signals transmitted over frequency-flat fading channels. The structure of an optimal differential PSK detector is derived and an analytical technique is proposed to compute its error performance. The results obtained by computer simulation are in excellent agreement with the analytical predictions.

Yu and Pasupathy (see ibid., vol.43, p. 1534-44, 1995) derived a maximum-likelihood sequence estimation (MLSE) receiver structure for unknown time varying frequency-selective Rayleigh-fading channels and uncorrelated diversity sources. This receiver design is extended to the case of correlated diversity sources. Correlated diversity sources typically arise with space diversity, where constraints on antenna volume require that diversity antennae be placed too closely together. Analytic and simulated bit-error rate (BER) curves are presented for receivers which exploit and ignore the correlation. In the former case, we find a small BER improvement that reduces with decreasing correlation. However, for a fixed receiver complexity, superior performance is achieved when the correlation is ignored.

This paper develops a receiver structure to perform jointly maximum-likelihood (ML) synchronization, equalization, and detection of a linearly modulated signal transmitted over a time-varying frequency-selective Rician-faded channel, corrupted by additive white Gaussian noise (AWGN). The receiver is particularly suited to a fast-fading channel, where other receivers that rely on estimating the channel cannot track it quickly enough. The signal mean and autocovariance are needed, and a scheme is proposed for estimating these quantities adaptively. The receiver processes the specular and diffuse components (corresponding to the signal mean and autocovariance) separately. Processing the known specular component is the classical detection problem. The unknown diffuse component is processed by predictors. We show that the predictors can achieve synchronization in a novel manner, if synchronization is required. A union bound on the receiver's bit-error rate (BER) is derived, and it tightly bounds simulated BERs in fast-fading at high signal-to-noise ratios (SNRs).

This paper describes a carrier-frequency offset estimator derived using maximum-likelihood techniques. The estimator is designed for a digital space-diversity receiver, operating in a wide-band frequency-selective multipath fading channel. The estimator is suited to a burst-mode time-division-multiple-access system, because the estimate is formed in an open-loop manner, and relies on a training sequence normally used in burst-mode systems. The main advantage is that it does not require channel state information; rather, it only requires knowledge of the auto-correlation of the channel. Simulation results show the estimator to be unbiased over a wide frequency range. The normalized error standard deviation is shown to be 0.0015 across the frequency offset range for a receiver with four diversity branches, when the normalized channel delay spread is 0.1, the signal-to-noise ratio (SNR) is 10 dB, and the training sequence length is 23 symbols. It is found that, for most parameter configurations, there is approximately a factor of two improvement in the estimate error standard deviation when the antenna diversity is doubled.

Spectral-amplitude-coding optical code-division multiple-access (OCDMA) systems are limited by interference between incoherent sources. A detailed analysis of this limit for a system with a balanced receiver is presented. Additional pulse-position modulation (PPM) coding is proposed as a method to improve the system performance beyond this limit. A simple and robust PPM decoding structure is proposed, and the performance analysis of the whole PPM-OCDMA system is presented. The interference-limited performance of the PPM-OCDMA system is found to be superior to that of the original system when the number of simultaneous users is of the order of the PPM word length or larger. In particular, for a PPM word length of two, an increase in spectral efficiency of up to 100% is possible with no change in the signaling rate, data rate, or bit-error rate (BER).

We compute the error performance of noncoherent detection receivers for FSK signals transmitted over fast frequency-flat Rician fading channels. Linearly time-varying fading models are used to derive closed-form expressions for the error probability of binary FSK signaling. Error bounds are established for the performance of M-ary orthogonal FSK. Simulation results are in excellent agreement with analytical predictions.

A maximum likelihood sequence estimator for the dispersive Rayleigh fading channel is developed. Following previous work, the MLSE uses a Kalman based channel estimator to acquire the channel parameters necessary to formulate the maximum likelihood metric. However, unlike the MLSE receiver presented previously, the proposed receiver uses the f-power series channel model to formulate the ML metric and the state space representation of the channel and the received samples. For channels with small delay spreads, this approach is advantageous because only a small number of parameters are required to be estimated by the Kalman channel estimator. Simulation results are presented for various channel parameters.

Linearly time-varying fading models are used to investigate noncoherent detection of FSK signals transmitted over frequency-flat fading channels. The structure of the optimal noncoherent FSK detector is derived and a novel analytical technique is proposed to compute the error performance of noncoherent FSK detectors in fast fading. Error performance results obtained by computer simulation are in excellent agreement with the analytical predictions.

We analyze the performance of coherent optical communication systems which employ polarization shift keying (PolSK) modulation, introducing for the first time the requirements for multilevel, differential PolSK. For M-ary signal constellations with M>2, the three-dimensional (3-D) nature of the signal constellations within Stokes space requires the use of double-differential modulation and detection (M-DDPolSK). Coding constraints on the signals place an additional restriction on such systems. We present detailed analysis and simulations for a 6-DDPolSK modulation scheme in which the signal points are located at the vertices of an octahedron in Stokes space.

This paper develops a maximum-likelihood sequence estimation (MLSE) diversity receiver for the time- and frequency-selective channel corrupted by additive Gaussian noise when linear constellations (M-ASK, M-PSK, M-QAM) are employed. The paper extends Ungerboeck's derivation of the extended MLSE receiver for the purely frequency-selective channel to the more general channel. Although the new receiver structure and metric assume ideal channel-state information (CSI), the receiver can be used wherever high-quality CSI is available, such as a comb of pilot tones or time-isolated symbols. The major contributions of this paper are as follows: (1) the derivation of a finite-complexity diversity receiver that is maximum likelihood (ML) for all linear channel models and sources of diversity, as long as ideal CSI is available; (2) a benchmark, in that the new receiver's performance is a lower bound on the performance of practical systems, which either lack ideal CSI or are not ML; (3) insight into matched filtering and ML diversity receiver processing for the time- and frequency-selective channels; and (4) bounds on the new receiver's bit-error rate (BER) for ideal CSI and pilot tone CSI, in a fast Rayleigh-fading channel with multiple independently faded paths. The new receiver can seamlessly tolerate square-root Nyquist pulses without a fading-induced ISI error floor.

A simple technique for removing intersymbol interference (ISI) introduced by "linearly frequency-selective" fading channels is presented. The technique involves the optimization of the overall impulse response of the transmit and receive filters and effectively reduces the channel to one which is flat fading. Computer simulation results show that this equalization method works for channels with small delay spreads.

An adaptive receiver is presented in this paper for the reception of linearly modulated signals transmitted over a time- and frequency-selective fading channel. The channel is modeled as a truncated power series which represents the dispersive fading channel as a sum of three elementary flat-fading channels. The proposed receiver consists of a sequence estimator with a parallel channel estimator. The channel estimator recovers the instantaneous fading processes associated with each elementary channel and filters them to generate one-step predictions of each fading process. Some implementation difficulties and solutions are also discussed. Computer simulations using quadrature phase-shift keying (QPSK) and channels with moderate delay spreads and fade rates have been used to evaluate the performance of the receiver. The results show that our technique has potential in channels with delay spread of about 20%, signal-to-noise ratio (SNR) greater than 15 dB, and applications requiring bit-error rates (BER's) less than 10/sup -2/.

Conventional differential detection is known to perform poorly on the fast-fading Rayleigh-fading channel due to the rapid variation of the channel state. The technique of multiple symbol differential detection (MSDD) improves the error performance by detecting over a sequence of symbols and exploiting the inherent correlation of the fading process. To further reduce the error rate, we need to introduce some form of coding. Successful coding for the Rayleigh-fading channel requires interleaving to decorrelate the channel-a conflicting requirement to MSDD. We propose a solution consisting of interleaving blocks of L M-ary phase-shift keying (MPSK) symbols that are multiple symbol differentially detected. The blocks are treated as elements of an M/sup L/ symbol alphabet over which a multilevel code based on geometrically uniform partitions is defined.

Linearly time-varying fading models are used to investigate noncoherent detection of frequency shift keying (FSK) signals transmitted over frequency-flat fading channels. The structure of the optimal noncoherent FSK detector is derived and a novel analytical technique is proposed to compute the error performance of noncoherent FSK detectors in fast fading. Error performance results obtained by computer simulation are in excellent agreement with the analytical predictions.

The majority of previous analytical studies of signal-space coding techniques (includes trellis and block codes) on the Rayleigh-fading channel have assumed ideal interleaving. The effect of finite interleaving on the performance of different coding schemes has been studied only by simulation In this paper we first derive a maximum likelihood (ML) decoder for codewords transmitted over a noninterleaved Rayleigh flat fading channel, followed by an exact expression for the pairwise error event probability of such a decoder. It includes phase shift keying (PSK), quadrature amplitude modulation (QAM) signal sets, trellis coded modulation (TCM) and block coded modulation (BCM) schemes, as well as coherent (ideal channel state information) and partially coherent (e.g., differential, pilot tone, etc.) detection. We derive an exact expression for the pairwise event probability in the case of very slow fading-i.e., the fading experienced by all the symbols of the codeword is highly correlated. We also show that the interleaving depth required to optimize code performance for a particular minimum fading bandwidth can be approximated by the first zero of the fading channel's auto-correlation function.

An algorithm, based on previous work [Vitetta and Taylor 1994, 1995], for the detection of coded and uncoded PSK sequences transmitted on a frequency-flat fading channel is investigated. It is based on the Viterbi algorithm and processes more than one signal sample per signaling interval. Its performance is evaluated by means of computer simulations for both uncoded and coded systems.

This paper describes novel detection algorithms for coded and uncoded M-ary phase shift keying (M-PSK) signals transmitted over frequency-flat fading channels. Fading distortion is estimated with per-survivor methods using the sampled outputs of two receive filters. Some of the proposed algorithms do not require information on the fading statistics and are dubbed blind algorithms. The error rate performance of blind and nonblind algorithms is assessed by simulation for uncoded and trellis-coded phase shift keying (PSK) and is compared to the performance of other detection schemes proposed in the literature.

Knowledge of the probability density function (pdf) of intersymbol interference (ISI) can be used in the computation of system error probabilities and to help identify the cause of the errors. Exact computation of the probability density is only practicable in a few special cases. Several algorithms exist for computing approximate pdfs of ISI but these only work with certain signal constellations and transmitted symbols that are mutually independent or coded by partial response. This paper considers an algorithm for the computation of ISI pdfs for trellis-coded modulation with general 1D and 2D constellations. The results suggest that symbol dependencies introduced into the transmitted signal by symmetric trellis-codes have negligible effect on the pdf of the ISI. Although the ideas presented are general in nature, examples are given for ISI pdfs on a line-of-sight radio channel.>

The problem of maximum likelihood (ML) detection for uncoded and coded M-PSK signals on Rayleigh fading channels is investigated. It is shown that, if the received signal is sampled at baud-rate, a ML receiver employing per-survivor processing can be implemented. The error rate performance of this receiver is evaluated by means of computer simulations and its limitations are discussed. In addition, it is shown that, on a fast fading channel, the error floor in the BER curve can be appreciably lowered if more than one received signal sample per symbol interval is processed by the receiver algorithm, Finally, a sub-optimum two-stage receiver structure for interleaved coded PSK systems is proposed. Its error rate performance is assessed for simple trellis-coded modulation schemes and compared to that provided by other receiver structures.

Maximum Likelihood (bit) detection of differentially encoded M-PSK signals on Rayleigh fading channels is investigated. It is shown that the solution to this problem can be related to previous results [Vitetta and Taylor, 1994] and leads to the implementation of a receiver structure based on the Viterbi algorithm and employing per-survivor processing. The error rate performance of this receiver is evaluated by means of computer simulations for both trellis-coded and uncoded systems.>

Multiuser signal set design for the energy-constrained linear real-additive symbol-synchronous additive white Gaussian noise channel is investigated in this correspondence. An argument is presented showing the suboptimality of binary alphabets which, in turn, uncovers a rule for the formation of multiuser signal sets. This simple rule leads to a theorem stating the dimensionality at which additional users can be added to a multiuser system without decreasing the minimum Euclidean distance. Specific vector sets with the desired property are then constructed as examples and some generalizations are discussed. The theorem may also be used as a platform for the design of more efficient multiuser codes incorporating redundant signal sets. Two such codes are presented. In particular, a code combining four-dimensional unit-energy signals with a rate-2/3 convolutional code obtains a summed rate of 1.75 bits/dimension in the two-user adder channel with received minimum Euclidean distance of 2.>

The performance of trellis-coded modulation (TCM) on additive white Gaussian noise channels is well understood, and tight analytical bounds exist on the probability of the Viterbi decoder making a decision error. When a channel is also time-dispersive, the performance of TCM systems has been studied mainly by simulation. However, simulation is limited to symbol error probabilities greater than 10/sup -6/ and is not a particularly useful tool for designing codes. Tight analytical bounds on the error probability of TCM on time-dispersive channels are required for a more thorough study of performance. Moreover, the design of good codes and optimum metrics for time-dispersive channels requires tight analytical bounds. In this paper we derive analytical upper bounds, which, although requiring numerical techniques for tractable evaluation, are tight for a wide range of time-dispersive channel conditions. The bounds are based on a union bound of error events that leads to a summation of pairwise error probabilities, which are themselves upper bounded.>

The paper develops a receiver structure for random Gaussian signals in additive noise based on the classic maximum likelihood (M-L), estimator-correlator derivation of Kailath [1960], and applies it to differential phase shift keying (DPSK) on the correlated Rayleigh-fading channel. It is shown to lower the error floor found in the performance of conventional DPSK receivers by orders of magnitude. In addition, the maximum-likelihood procedure is shown to make uncorrelated symbol decisions. The performance of both conventional and optimal receivers, which require knowledge of the channel statistics, is examined analytically. A recursive, channel-adaptive version of the optimal receiver, utilizing decision feedback to estimate the channel statistics, is developed. Its simulated performance shows no penalty compared to theoretical calculations which require explicit knowledge of the channel statistics.>

This paper examines the joint synchronization and detection of Ungerboeck coded modulation. Estimation theory is used to derive a synchronization structure that is efficient in estimating carrier phase and symbol timing . The maximum like-Lihood receiver generates estimates of carrier phase and timing that are free of data-dependent jitter for any continuous pulse shape. Various feedback schemes to be used with Ungerbdeck codes are presented and simulated. Simulations show that efficient estimates of camer and clock can be found when joint data and parameter estimation of an Ungerboeck coded signal is performed.

Trellis-coding techniques are applied to continuous-phase frequency-shift keying (CPFSK). A new coding scheme based on convolutional codes on the ring of integers modulo-P is shown to be a natural way to apply trellis coding to CPFSK. Previous work has decomposed CPFSK into two parts: a linear encoder, with memory called the continuous phase encoder (CPE), and a memoryless modulator (MM), where the CPE often has a code structure defined over the ring of integers modulo-P. The combination of a modulo-P convolutional channel encoder (CE) and the CPE is a linear modulo-P encoder. Design examples are given for rate-1/2 coded quaternary CPFSK with modulation indexes 1/2 and 1/4, and rate-2/3 coded octal CPFSK with modulation index 1/8. Combinations are optimized in the normalized minimum Euclidean distance sense for a given total number of states in the overall MLSE receiver. Numerical results show that this new coding scheme consistently obtains better performance than previous schemes.>

Local area networks (LANs) with a branching rooted-tree topology are considered. In these systems, the metropolitan area gateway (MAG) is responsible for coordinating the flow of traffic into and out of the local subnetworks. A traffic scheduling technique is proposed for this topology. The algorithm is motivated by a finite-horizon maximization of conditional channel cycle utilization, resulting in a zero/infinite scheduling window scheme. Improved mean delay performance is achieved over a much wider range of traffic conditions than for other known algorithms. In order to obtain this robust behaviour, the local media access algorithms are modified to interact with the dynamic scheduling scheme in the MAG.>

Blind equalization performance can be improved by controlling the equalizer adaption step-size according to the region in which the received signal lies in the constellation plane. A simple technique based on this principle is proposed as a modification to blind equalization algorithms, and its effectiveness is demonstrated by simulation. As an example, the algorithm of Picchi and Prati (1987) is considered since it is simpler and converges faster. The proposed modification is applied to the zero-forcing (ZF) algorithm.>

Comparisons are made of a genie-aided sequential algorithm due to D. Haccoun and M.J. Ferguson (1975), the Viterbi algorithm, the M-algorithm, and the Fano algorithm for rate-1/2 and rate-2/3 trellis modulation codes on rectangular signal sets. The effects of signal-to-noise ratio and decoding-delay constraints on the choice of decoding algorithms for framed data are examined by computer simulation. Additionally, the genie-aided algorithm is used as a tool in estimating the asymptotic behavior of the M-algorithm. In general, the results conform closely to experience with convolutional codes due to the similar distance structure of the codes. The Fano algorithm produces good error performance with a low average number of computations when long decoding delay is permissible. The M-algorithm provides a savings in computation compared to the Viterbi algorithm if a small decoding delay is required.>

Following V.V. Ginzburg (1984), a hierarchy of codes is proposed to match the geometric partitioning of a signal set. The authors show that coset codes (including Ungerboeck, lattice, and binary codes) and indeed any codes which rely on partitioning of the signal set are all subclasses of the proposed coding scheme. The combination of such codes in a multilevel scheme often leads to reduced complexity in comparison with previously published schemes. A variety of decoder structures is discussed.>

Sphere-packing arguments are used to develop upper bounds on the free distance of trellis codes. A general bounding procedure is presented. Sphere packing bounds, including bounds on the density of infinite regions, packings for hypercubes, and packings on the surface of a unit sphere, are then used to produce bounds for a wide variety of trellis codes. Among the applications are convolutional codes, Ungerboeck codes for phase-shift keying and quadrature amplitude modulation, coset codes, and continuous phase modulation codes. The new bounds are significantly tighter than existing bounds in many cases of practical interest.>

A method of utilizing the available communication bandwidth efficiently is provided by multi-hphase coding. A receiver structure for joint estimation of data, carrier phase, and symbol timing is derived based on ML estimation techniques and subsequently assessed analytically and by means of Monte Carlo simulation procedures. Results indicate that the receiver structure, although not strictly optimum, yields near-optimum performance on the additive white Gaussian noise channel.

Multi-hschemes are bandwidth-efficient CPM signaling formats which offer means of coding without explicit redundancy. However, the required (optimal) receiver structures tend to be very complex [1], [4], [8] requiring a large number of filters of correlators. Here a reduced complexity receiver for the joint estimation ofM-ary data, carrier phase, and symbol timing is realized through the use of an approximate representation of the likelihood function. Evaluation of receiver performance indicates a fraction of a decibel degradation in coding gain and almost no loss in synchronization performance.

A method of finding good Ungerboeck codes for large rectangular [quadrature amplitude modulation (QAM)] signal sets is described. Using the concept of Euclidean weights due to Ungerboeck, we prove that a2^{n}point basic constellation may be employed to determine exactly the free distance for an Ungerboeck-coded rectangular2^{m}point set, whenm-n-1bits are uncoded and the remaining bits pass through a rate(n-1)/nconvolutionai encoder. It is shown that rate2/3encoders may be used to achieve most of the theoretically possible coding gain in the proposed scheme where the effect of the error coefficient on the coding gain has been considered.

A receiver structure based on maximum likelihood sequence estimation (MLSE) for receiving quaternary phase shift keyed signals over band-limited, nonlinear satellite channels, in the presence of additive downlink Gaussian noise, is derived. This work is an extension of work done by Mesiya et al. [2] for binary PSK transmission. Error performance of the receiver at low to moderate signal-to-noise ratios is estimated using computer simulation. For moderate to high signal-to-noise ratios, an upper bound on the probability of symbol error is obtained. A suboptimum receiver structure is then developed and its performance evaluated using computer simulation. It is shown that the performance of the suboptimum receiver is very close to that of the MLSE receiver.

The use of adaptive decision feedback techniques in the joint estimation of data, carrier phase, and symbol timing is investigated for a 49-QPRS 90 Mbit/s digital radio. The performance of the carrier and timing recovery loops are analyzed, first in a multipathfree environment and then for the case of a multipath distorted signal and a fixed decision feedback equalizer. Finally, the joint problem of data recovery, estimation of carrier phase, and symbol timing for a

The performance of a 49-QPRS, 90 Mbit/s digital radio receiver equipped with a decision feedback equalizer (DFE) to counter multipath fading is investigated via computer simulation. The simulation includes the transmitted data, multipath fade model, receiver model, and DFE. The results indicate that a DFE equipped with five forward and five feedback taps can adequately compensate a 40 dB minimum-phase fade anywhere in the receiver passband. The study is extended to other receiver configurations including the use of space diversity and/or slope equalizers and the use of a transversal equalizer (TE) with the same delay-span in place of the DFE. The results indicate that the DFE equipped receiver outperforms the TE receiver and that still better performance may be achieved using a combination of space diversity and DFE.

A theoretical analysis of the problem of estimating frequency selective fade model parameters from measurements on existing analog links is presented. These measurements essentially consist of obtaining records of AGC and continuity pilot level variations from which a first approximation ofa, \alpha, \omega_{0}, and τ (fade model parameters) can be obtained. Field results obtained on a working link in New Zealand are reported.

In a time-division or burst mode communications systems, synchronization must be achieved within a short preamble time at the beginning of each burst. When used in such a system, a conventional phase-locked loop (PLL) occasionally exhibits a prolonged phase acquisition transient. This effect, known as hangup, occurs at a large value of phase error and may cause loss of a data burst. The present paper investigates a modification of a PLL known as a limit-switched loop (LSL). The LSL offers faster phase acquisition and better hangup immunity than a PLL. The LSL estimates its position in the phase plane at the beginning of acquisition and, if hangup is about to occur, makes a phase correction so as to forestall its occurrence. The structure and properties of the LSL are investigated in this paper.

A rapidly convergent approximation to the likelihood ratio for the generic problem of detecting a stationary discrete-time stochastic process in additive white Gaussian noise is formulated. The derivation is based on the innovations approach to detection theory.

A decision feedback receiver structure for receiving binary phase-shift keyed signals over bandlimited, nonlinear satellite channels, in the presence of both uplink and downlink additive Gaussian noise, is derived. As an illustrative example of the receiver, a hard-limiting channel model is considered in detail, and two special cases are examined. Based on the analysis, an alternative receiver structure which is more suitable for implementation is derived. The performance of the receiver is evaluated by computer simulation techniques, and it is shown that significant performance improvement over the single sample sign detector can be obtained.

Multi-hphase coding is a bandwidth-efficient modulation which offers substantial performance improvement over conventional digital modulations with little or no bandwidth expansion. This paper examines the demodulation of the multi-hsignals, and in particular examines the behavior of a decision-directed carrier loop which utilizes the Viterbi algorithm decoder in the calculation of the loop error signal. Both analytic and simulation results are presented and the agreement between the two is shown to be very close.

Signal space coded signaling systems have the potential of achieving significant error-rate improvement without the bandwidth expansion normally encountered when error-correcting codes are used. This paper investigates two such systems by means of computer simulation. A band-limited, nonlinear channel is assumed and both systems exhibit performance improvenment compared to uncoded systems.

An expression is derived for the error probability of binary CPSK signals received over a soft-limited channel in the presence of pre- and post-limiter additive Ganssian noise. Although this expression is not in closed form, it may be efficiently evaluated to any desired accuracy using a recurrence method developed in the appendix. System performance is also evaluated when the signals are subjected to bandlimiting as well as Gaussian noise preceding the limiter. Moment space techniques are used to obtain upper and lower bounds on the errorrate in terms of the error-rate in the presence of Gaussian noise alone. Numerical examples are presented for a channel which includes a Chebyshev filter preceding the limiter. These illustrate that the bounds are tight enough for most practical purposes.

The Canadian RD-3 digital radio system will utilize duobinary encoded, offset QPSK modulated signals. This complex signal structure is such that a conventional fourth order nonlinearity and phase-locked loop cannot be used for carrier recovery. In this paper, a decision-directed phase-lock loop is proposed for carrier recovery and demodulation of these signals. The equations describing loop operation are obtained, and the effective phase-detector characteristic is examined. Fokker-Planck techniques are then employed to approximately find the steady-state probability density function of the phase error, and this is used to find an upper bound on the average probability of error.

A method is presented for analyzing the performance of M-ary coherent phase-shift-keyed (CPSK) signals which have been transmitted through a hard limiting channel in the presence of linear intersymbol interference preceding the limiter and additive Gaussian noise both preceding and following it. The probability of error is obtained in the form of an infinite series as a function of the expected values of trigonometric functions of the interference. The characteristic function of the interference is expanded in a power series and used to obtain these expected values. The resulting error probability is obtained as the product of two absolutely convergent series. Numerical examples are presented to illustrate the technique.

A Maximum Likelihood receiver for binary phase-shift keyed signals is derived for the case when the channel includes a bandpass nonlinearity preceded and followed by sources of additive noise. This optimum receiver is then approximated for the case of a limitertype of channel and is seen to be a nonlinear function of both pre- and post-limiter signal to noise ratios and a normalized limiter softness factor λ*. The probability of error for this receiver is evaluated for the case of a large time-bandwidth productWTusing a central limit theorem argument.

A new class of codes in signal space is presented, and their error and spectral properties are investigated. A constant-amplitude continuous-phase signal carries a coded sequence of linear-phase changes; the possible signal phases form a cylindrical trellis in phase and time. Simple codes using 4-16 phases, together with a Viterbi algorithm decoder, allow transmitter power savings of 2-4 dB over binary phase-shift keying in a narrower bandwidth. A method is given to compute the free distance, and the error rates of all the useful codes are given. A software-instrumented decoder is tested on a simulated Gaussian channel to determine multiple error patterns. The error parameterR_{o}is computed for a somewhat more general class of codes and is shown to increase rapidly when mere phases are employed. Finally, power spectral density curves are presented for several codes, which show that this type of coding does not increase the transmitted signal bandwidth.

An analysis is presented showing the effect of carrier phase error on the error-rate of a duobinary shaped, quaternary, phaseshift keying digital radio system. A simple upper bound on the probability of error as a function of phase error is derived and the results are compared to those obtained by computer simulation. The conclusion is reached that such a system exhibits approximately 1.0 dB degradation for a carrier phase error of 3° at an error-rate of 10-5and a 10 dB degradation at a phase error of 15°.

An analytical approach is developed to investigate the performance of coherent phase-shift keying (CPSK) systems transmitted through a nonlinear satellite transponder. A novel quadrature model of the satellite nonlinearity is introduced, and is found to yield an excellent fit to the actual traveling wave tube (TWT) characteristics as commonly used in a satellite transponder, with the advantage of requiring the choosing of only a few parameter values. Based on this quadrature model an expression for the bit error rate is derived in the form of an infinite series where the leading term corresponds to the error-rate for a linear channel and the remaining terms are correction terms accounting for nonlinear effects. The performance of a binary PSK system is computed and presented as a function of the up-link and down-link carrier-to-noise-power ratios (CNR) and a bandwidth spread factor.

A computer simulation study has been performed to evaluate the performance of a variety of digital modulation techniques in transmitting high-rate digital data over a satellite channel. Results are presented showing comparative performance of various techniques in the form of error-rate curves and signal-to-noise ratio (SNR) degradation curves.

With regard to Mark's statements concerning the adaptive algorithm described in the previous paper, he is correct in stating that the basic form of the algorithm is due to him. The author wishes to extend an apology to Dr. Mark for failing to credit him specifically in the previous paper1 for developing the basic form of this algorithm. The final form of the algorithm used in that paper and described therein by (36) and (37) represents an extension to the algorithm developed by Mark, and should have been labeled as such rather than as a new algorithm. With regard to Mark's statement concerning the use of the tanh(·) nonlinearity, the author did publish a brief letter (1971) concerning some simulations using such a nonlinearity. However, the present paper contains the theoretical justification for this nonlinearity and its use in equalization, the results of extensive simulations using it, and a comparison of its performance to presently existing equalizers. All of this last, contrary to Mark's claim, represents new work.

A recursive nonlinear equalizer has been developed. The Bayes estimation theory has been used to obtain an unrealizable nonlinear minimum mean-square error (MMSE) receiver for the reception of binary pulse-amplitude-modulation signals in the presence of intersymbol interference and noise. A realizable approximation to the Bayes structure has been derived as the combination of a matched filter and a nonlinear recursive equalizer structure. The equalizer has been made adaptive using a new algorithm that defines and maintains the time frame of reference and is constrained so that the equalizer's parameters always move toward their optimum values. Computer simulations have been used to demonstrate the properties of the estimate feedback equalizer (EFE) and to compare its performance to that of presently known equalizers.