In this paper, a MIMO-OFDM system operating with SFBC encoded signals and affected by inter-cell interference is analyzed, and the performances of Interference Aware-Minimum Mean Square Error (IA-MMSE) symbol estimators are discussed in presence of realistic estimation of the interference auto-correlation matrix. Different estimation techniques are described and compared, based both on a system description using a 4×2 channel matrix (requiring the estimation and inversion of a 4×4 auto-correlation matrix), or using two 2×2 matrices (requiring the estimation and inversion of two 2×2 auto-correlation matrices). The first approach offers better performances, but the second approach has a very limited loss, counterbalanced by a simpler implementation complexity, characteristics that makes it an interesting solution in practical applications. An extensive simulation campaign using realistic system level parameters has been used for performance assessment.

One of the main open problems for next generation wireless networks, is to find the new OFDM-based waveform to be used in 5G. The new modulation scheme must primarily be able to achieve higher spectral efficiency than its predecessor. The main 3GPP's candidate is a new version of OFDM, called Filtered Orthogonal Frequency-Division Modulation (f-OFDM), which is similar to OFDM but with additional filtering in order to reduce Out-Of-Band (OOB) emissions and to obtain a better spectral-localization. Another option is windowed-OFDM (w-OFDM), which is basically a classical OFDM scheme where each symbol is windowed and overlapped in the time domain. In this paper we compare classic OFDM signals using Cyclic Prefix (CP-OFDM) with f-OFDM and w-OFDM, each one with multiple parametric options and numerologies. A multirate transmitter simultaneously operating with multiple numerologies is considered, where the transmitted sub-bands must be up-sampled and interpolated in order to generate the composite numerical signal fed to the Digital to Analog Converter (DAC). Finally, we discuss advantages and disadvantages of the various schemes.

Consider a two-dimensional domain S ⊆ ℜ2 containing two sets of nodes from two statistically independent uniform Poisson point processes with constant densities pL and pNL. The first point process identifies the distribution of a set of nodes having information about their positions, hereafter denoted as L-nodes (localised-nodes), whereas the other is used to model the spatial distribution of nodes that need to localise themselves, hereafter denoted as NL-nodes (not localised-nodes). For simplicity, both kinds of nodes are equipped with the same kind of transceiver, and communicate over a channel affected by shadow fading. As a first goal, the authors derive the probability that a randomly chosen NL-node over S gets localised as a function of a variety of parameters. Then, the authors derive the probability that the whole network of NL-nodes over S gets localised. As with many other random graph properties, the localisation probability is a monotone graph property showing thresholds. In this work, the authors derive both finite (when the number of nodes in the bounded domain is finite and does not grow) and asymptotic thresholds for the localisation probability. In connection with the asymptotic thresholds, the authors show the presence of asymptotic thresholds on the network localisation probability in two different scenarios. The first refers to dense networks, which arise when the domain S is bounded and the densities of the two kinds of nodes tend to grow unboundedly. The second kind of thresholds manifest themselves when the considered domain increases but the number of nodes grow in such a way that the L-node density remains constant throughout the investigated domain. In this scenario, what matters is the minimum value of the maximum transmission range averaged over the fading process, denoted as dmax, above which the network of NL-nodes almost surely gets asymptotically localised.

This paper presents a modified proportional fairness (PF) criterion suitable for mitigating the rate anomaly problem of multirate IEEE 802.11 Wireless LANs employing the mandatory distributed coordination function (DCF) option. Compared to the widely adopted assumption of saturated network, the proposed criterion can be applied to general networks whereby the contending stations are characterized by specific packet arrival rates, lambdas, and transmission rates Rds. The throughput allocation resulting from the proposed algorithm is able to greatly increase the aggregate throughput of the DCF while ensuring fairness levels among the stations of the same order of the ones available with the classical PF criterion. Put simply, each station is allocated a throughput that depends on a suitable normalization of its packet rate, which, to some extent, measures the frequency by which the station tries to gain access to the channel. Simulation results are presented for some sample scenarios, confirming the effectiveness of the proposed criterion.

Consider a bidimensional domain S sube Rfr2and throw two statistically independent uniform poisson point processes with constant densities equal toPLandPNL, respectively. The first point process identifies the spatial distribution of a set of nodes which has information about their position, hereafter denoted as L-nodes, while the other one is used to model the spatial distribution of nodes which need to localize themselves, hereafter denoted as NL-nodes. Both kind of nodes are equipped by the same kind of transceiver, and communicate over a channel affected by shadow fading. The goal of this paper is to derive the probability that a randomly chosen NL-node over the domain S gets localized as a function of a variety of transmission parameters. As many random graph properties, the localization probability is a monotone graph property presenting thresholds. We derive finite thresholds for the localization probability. The envisaged scenario refers to the case in which the number of deployed nodes of both point processes is finite. Simulation results closely match the theoretical derivations confirming the effectiveness of the employed probabilistic model.

The aim of this paper is twofold. On one hand, it presents a multi-dimensional Markovian state transition model characterizing the behavior at the medium access control (MAC) layer by including transmission states that account for packet transmission failures due to errors caused by propagation through the channel, along with a state characterizing the system when there are no packets to be transmitted in the queue of a station (to model non-saturated traffic conditions). On the other hand, it provides a throughput analysis of the IEEE 802.11 protocol at the data link layer in both saturated and non-saturated traffic conditions taking into account the impact of both transmission channel and multirate transmission in Rayleigh fading environment. Simulation results closely match the theoretical derivations confirming the effectiveness of the proposed model.

In this paper, we provide a saturation throughput analysis of the IEEE 802.11 protocol at the data link layer by including the impact of both transmission channel and capture effects in Rayleigh fading environment. Impacts of both non-ideal channel and capture effects, specially in an environment of high interference, become important in terms of the actual observed throughput. As far as the 4-way handshaking mechanism is concerned, we extend the multi-dimensional Markovian state transition model characterizing the behavior at the MAC layer by including transmission states that account for packet transmission failures due to errors caused by propagation through the channel. This way, any channel model characterizing the physical transmission medium can be accommodated, including AWGN and fading channels. We also extend the Markov model in order to consider the behavior of the contention window when employing the basic 2-way handshaking mechanism. Under the usual assumptions regarding the traffic generated per node and independence of packet collisions, we solve for the stationary probabilities of the Markov chain and develop expressions for the saturation throughput as a function of the number of terminals, packet sizes, raw channel error rates, capture probability, and other key system parameters. The theoretical derivations are then compared to simulation results confirming the effectiveness of the proposed models.

In this paper, we provide a throughput analysis of the IEEE 802.11 protocol at the data link layer in non-saturated traffic conditions taking into account the impact of both transmission channel and capture effects in Rayleigh fading environment. The impact of both non-ideal channel and capture become important in terms of the actual observed throughput in typical network conditions whereby traffic is mainly unsaturated, especially in an environment of high interference. We extend the multi-dimensional Markovian state transition model characterizing the behavior at the MAC layer by including transmission states that account for packet transmission failures due to errors caused by propagation through the channel, along with a state characterizing the system when there are no packets to be transmitted in the buffer of a station. Finally, we derive a linear model of the throughput along with its interval of validity. Simulation results closely match the theoretical derivations confirming the effectiveness of the proposed model.

In this paper, we provide a throughput analysis of the IEEE 802.11 protocol at the data link layer in non- saturated traffic conditions taking into account the impact of both transmission channel and capture effects in Rayleigh fading environment. Impacts of both non-ideal channel and capture become important in terms of the actual observed throughput in typical network conditions whereby traffic is mainly unsaturated, specially in an environment of high interference. We extend the multi-dimensional Markovian state transition model characterizing the behavior at the MAC layer by including transmission states that account for packet transmission failures due to errors caused by propagation through the channel, along with a state characterizing the system when there are no packets to be transmitted in the buffer of a station.

This article deals with localization probability in a network of randomly distributed communication nodes contained in a bounded domain. A fraction of the nodes denoted as L-nodes are assumed to have localization information while the rest of the nodes denoted as NL nodes do not. The basic model assumes each node has a certain radio coverage within which it can make relative distance measurements. We model both the case radio coverage is fixed and the case radio coverage is determined by signal strength measurements in a Log-Normal Shadowing environment. We apply the probabilistic method to determine the probability of NL-node localization as a function of the coverage area to domain area ratio and the density of L-nodes. We establish analytical expressions for this probability and the transition thresholds with respect to key parameters whereby marked change in the probability behavior is observed. The theoretical results presented in the article are supported by simulations.

This letter considers low-density parity-check (LDPC) coding of correlated binary sources and a novel iterative joint channel decoding without communication of any side information. We demonstrate that depending on the extent of the source correlation, additional coding gains can be obtained. Two stages of iterative decoding are employed. During global iterations, updated estimates of the source correlation are obtained and passed on to the sum-product decoder that performs local iterations with a predefined stopping criterion and/or a maximum number of local decoding iterations. Simulation results indicate that very few global iterations (2-5) are sufficient to reap significant benefits from implicit knowledge of source correlation. Finally, we provide analytical performance bounds for our iterative joint decoder and comparisons with sample simulation results.

In this article we exploit the potential correlation existing between multiple information sources to achieve additional coding gains from the channel codes used for data protection. We do not assume the existence of, nor do we use channel side-information at the receiver. Instead, empirical estimates of the cross-correlation are used in partial decoding steps in an iterative joint soft decoding paradigm. Experimental results suggest that relatively few iterations (2 to 4) are sufficient to reap significant gains using this approach specially when the sources are highly correlated. Finally, we provide analytical performance bounds of the proposed technique showing a close match with the simulation results at sufficiently high SNR

The evaluation of the minimum distance of low-density parity-check (LDPC) codes remains an open problem due to the rather large dimension of the parity check matrix H associated with any practical code. In this article, we propose an effective modification of the error impulse (EI) technique for estimation of the minimum distance of the LDPCs. The EI method is successfully applied to suboptimum decoding algorithms such as the iterative MAP decoding algorithm for turbo codes. We present novel modifications and extensions of this method to the suboptimum iterative sum-product algorithm for LDPCs. Simulation results validate the functionality of the proposed technique. Simulations focus on a particular class of LDPC codes, but our approach is general and applies to any LDPC code.

This correspondence looks at the problem of joint decoding of serially concatenated convolutional codes (SCCCs) used for channel coding of multiple correlated sources. We assume a simple model whereby two correlated sources transmit SCCC encoded data to a single destination receiver. We do not assume the existence of, nor do we use channel side information at the receiver. In particular, we present a novel iterative joint channel decoding algorithm for correlated sources by using the empirical cross-correlation measurements at successive decoding iterations to provide extrinsic information to the outer codes of the SCCC configuration. Two levels of soft metric iterative decoding are used at the receiver: 1) iterative maximum a posteriori probability (MAP) decoding is used for efficient decoding of individual SCCC codes (local iterations) and 2) iterative extrinsic information feedback generated from the estimates of the empirical cross correlation in partial decoding steps is used to pass soft information to the outer decoders of the global joint SCCC decoder (global iterations). We provide analytical results followed by simulation studies confirming the robustness of the cross-correlation estimates to channel-induced errors, justifying the use of such estimates in iterative decoding. Experimental results suggest that relatively few global iterations (two to five) during which multiple local iterations are conducted are sufficient to reap significant gains using this approach specially when the sources are highly correlated.

This paper is focused on the problem of significantly reducing the complexity of the recursive interleaver growth algorithm (IGA) with the goal of extending the range of applicability of the algorithm to significantly larger interleavers for a given CPU time and processor. In particular, we present two novel modifications to IGA changing the complexity order of the algorithm from O(N/sub max//sup 4/) to O(N/sub max//sup 2/), present several further minor modifications reducing the CPU time albeit not fundamentally changing the complexity order, and present a mixed mode strategy that combines the results of complexity reduction techniques that do not alter the algorithm outcome itself, with a novel transposition value set cardinality constrained design that does modify the optimization results. The mixed strategy can be used to further extend the range of interleaver sizes by changing the complexity order from O(N/sub max//sup 2/) to O(N/sub max/) (i.e., linear in the interleaver size). Finally, we present optimized variable length interleavers for the Universal Mobile Telecommunications System (UMTS) and Consultative Committee for Space Data Systems (CCSDS) standards outperforming the best interleavers proposed in the literature.

This paper is aimed at the problem of designing optimized interleavers for parallel concatenated convolutional codes (PCCC) that satisfy several requirements simultaneously: 1) designing interleavers tailored to the constituent codes of the PCCC; 2) improving the distance spectra of the resulting turbo codes which dominate their asymptotic performance; 3) constructing optimized interleavers recursively so that they are implicitly prunable; and 4) completely avoiding short permutation cycles in order to reduce the risk of having strong correlations between the extrinsic information during iterative decoding. To this end, we present two theorems that lead to a modification of a previously developed iterative interleaver growth algorithm (IGA) that can be used to design optimized variable-length interleavers, whereby at every length the optimized permutation implemented by the interleaver is a single-cycle permutation. Two more modifications of the IGA are presented to improve the performance of the optimized interleavers at a reduced complexity. The optimization is achieved via constrained minimization of a cost function closely related to the asymptotic bit-error rate or frame-error rate of the code.

This letter presents the results of the search for optimum punctured recursive convolutional codes (RCCs) of rate k/k+1, for k=2,...,8, suitable for concatenated channel codes whose constituent encoders are recursive, systematic convolutional codes. The mother codes that are punctured are rate-1/2 RCCs proposed for use in parallel and/or serial concatenation schemes. Extensive tables of systematic and nonsystematic puncturing patterns, optimized relative to various objective functions suitable for concatenated channel codes, are presented for several mother codes.

In this letter, we view the implicit unequal error protection observed in the asymptotic performance of most turbo codes as an impetus to the formulation of a rate-allocation problem, associated with the distribution of a fixed quota of coded bits in the trellis sections of the upper and lower recursive systematic convolutional codes of the turbo-code structure. We then present an effective greedy approach for solving this rate-allocation problem via a two-phase process of puncturing and repetition coding, resulting in the improvement of the asymptotic performance of the code and lowering of its error floor. Sample simulation results are presented, confirming the potential gains of the approach.

In many practical applications requiring variable-rate coding and/or high-rate coding for spectral efficiency, there is a need to employ high-rate convolutional codes (CC), either by themselves or in a parallel or serially concatenated scheme. For such applications, in order to keep the trellis complexity of the code constant and to permit the use of a simplified decoder that can accommodate multiple rates, a mother CC is punctured to obtain codes with a variety of rates. This correspondence presents the results of extensive search for optimal puncturing patterns for recursive convolutional codes leading to codes of rate k/(k+1) (k an integer) to be used in serially concatenated convolutional codes (SCCC). The code optimization is in the sense of minimizing the required signal-to-noise ratio (SNR) for two target bit-error rate (BER) and two target frame-error rate (FER) values. We provide extensive sample simulation results for rate-k/(k+1) SCCC codes employing our optimized punctured CC.

This paper addresses the problem of interleaver design for serially concatenated convolutional codes (SCCCs) tailored to the constituent codes of the SCCC configuration. We present a theoretical framework for interleaver optimization based on a cost function closely tied to the asymptotic bit-error rate (BER) of the block code C/sub s/ resulting from proper termination of the constituent codes in the SCCC code. We define a canonical form of the interleaving engine denoted as the finite state permuter (FSP) and using its structural property, develop a systematic iterative technique for construction of interleavers. The core theoretical results focus on the asymptotic behavior of a class of cost functions and their martingale property, which is then used to develop an order recursive interleaver optimization algorithm. We address the issue of the complexity of the interleaver growth algorithm presented in the paper and demonstrate that it has polynomial complexity. Subsequently, we provide details about the application of the proposed technique and present a modification of the algorithm that employs error pattern feedback for improved performance at a reduced complexity. Sample experimental results are provided for an SCCC code of rate 1/3 and information block length 320 that achieves a minimum distance of d/sub min/=44.

In this paper, we address the issue of pruning (i.e., shortening) a given interleaver in a parallel concatenated convolutional code (PCCC). The principle goal of pruning is that of construction of variable-length and hence delay interleavers with application to PCCC, using the same structure (possibly in hardware) of the interleaver and deinterleaver units. As a side benefit, it is sometimes possible to reduce the interleaver length and hence delay for nearly the same and sometimes even better asymptotic performance. In particular, we present a systematic technique for interleaver pruning and demonstrate the average optimality of the strategy. Sample simulation results are presented confirming the average optimality of the proposed scheme.

In this correspondence, we present a technique for generation of linear subcodes of a given turbo code with better distance spectrum than the original mother turbo code, via an iterative process of trace-bit injection which minimally reduces code rate, followed by selective puncturing that allows recovery of the rate loss incurred during the trace-bit injection. The technique allows for asymptotic performance improvement of any linear turbo code. In effect, we trim the distance spectrum of a turbo code via elimination of the low distance and/or high multiplicity codewords from the output space of the code. To this end, we perform a greedy minimization of a cost function closely related to the asymptotic bit error probability (or frame error probability) of the code. This improves the performance of the code everywhere, but its main impact is a reduction in the error floor of the turbo code which is important for delay constrained applications employing short interleavers.

This paper is aimed at the problem of designing optimized interleavers for parallel concatenated convolutional codes (PCCC) that satisfy several requirements simultaneously: 1) designing interleavers tailored to the constituent codes of the PCCC; 2) improving the distance spectra of the resulting turbo codes which dominate their asymptotic performance; and 3) constructing optimized interleavers recursively so that they are implicitly prunable. Two more modifications of a previously developed iterative interleaver growth algorithm (IGA) of polynomial complexity [F. Daneshragan et al., Sept. 1999] are presented to improve the performance of the optimized interleavers at a reduced complexity: 1) a growing window is used to trap error patterns of proper length in order to form the cost function; and 2) we employ error feedback to further improve the distance spectrum, of the optimized codes and to reduce complexity. The optimization is achieved via constrained minimization of a cost function closely related to the asymptotic bit error rate (BER) or frame error rate (FER) of the codes.

This paper looks at the problem of the implicit unequal error protection observed in the asymptotic performance of most turbo codes. In particular we propose a new formulation of the rate allocation problem associated with the distribution of a fixed quota of coded bits in the trellis sections of the upper and lower recursive systematic convolutional (RSC) codes of the turbo code structure. The main goal of this work is the design of an effective greedy approach for solving this rate allocation problem via a two phase process of puncturing and repetition coding. Sample simulation results confirm that the asymptotic performances in terms of error floor are improved confirming the potential gain of this approach.

This paper deals with the problem of channel identification for single input multiple output (SIMO) slow fading channels using clustering algorithms. The received data vectors of the SIMO model are spread in clusters because of the AWGN. Each cluster is centered around the ideal channel output labels without noise. Starting from the Markov SIMO channel model, simultaneous maximum-likelihood estimation of the input vector and the channel coefficients reduces to one of obtaining the values of this pair that minimizes the sum of the Euclidean norms between the received and the estimated output vectors. The Viterbi algorithm can be used for this purpose provided the trellis diagram of the Markov model can be labeled with the noiseless channel outputs. The problem of identification of the ideal channel outputs, which is the focus of this paper, is then equivalent to designing a vector quantizer (VQ) from a training set corresponding to the observed noisy channel outputs. The Linde-Buzo-Gray (1980) type clustering algorithms could be used to obtain the noiseless channel output labels from the noisy received vectors. This paper looks at two critical issues with regards to the use of VQ for channel identification. The first has to deal with the applicability of this technique in general. We present theoretical results showing the conditions under which the technique may be applicable. The second aims at overcoming the codebook initialization problem by proposing a novel approach which attempts to make the first phase of the channel estimation faster than the classical codebook initialization methods.

This paper looks at the problem of optimizing a given parallel concatenated convolutional code (PCCC) via an iterative process of trace-bit injection which minimally reduces the code rate, followed by selective puncturing that allows recovery of the rate loss incurred during the trace-bit injection. The technique allows for optimization of any linear PCCC. The optimization is in the sense of trimming the distance spectrum of the overall PCCC via elimination of the low distance and/or high multiplicity codewords from the output space of the code, using a cost function closely related to the asymptotic bit error probability of the code. This improves the performance of the code everywhere, but the main impact of the optimization process is a reduction in the error floor of the PCCC.

We present the results of the optimization applied to the design of interleavers for rate-1/n parallel concatenated convolutional codes (PCCC) tailored to specific recursive systematic convolutional (RSC) constituent codes. The emphasis is on low-latency codes associated with interleavers of block length less than or equal to 160. The error floors of the optimized codes are significantly lower than those associated with the use of random interleavers. The distance spectra of the equivalent block codes resulting from trellis termination applied to PCCC are evaluated and used to obtain asymptotic bit error rate (BER) curves for the optimized codes.

Abstract A major bottleneck to the design of a truly wideband reconfigurable Software Radio (SR) transceiver is the front end of the system. It is desirable to push the analog‐to‐digital boundary of the SR transceiver as close to the antenna as possible. This places great demands on analog‐to‐digital converters that must produce high‐resolution samples of the incoming signal centered around a carrier frequency in the gigahertz range. A very promising architecture for the design of such converters is the ΣΔ architecture. A critical component of the ΣΔ converter is the rate‐conversion filter that is responsible for rejection of the out‐of‐band noise. Cascaded‐Integrator‐Comb (CIC) filters are efficient antialiasing rate‐conversion filter structures realized by cascading integrator and comb cells separated by a decimation block. High‐order structures, attempting to increase the rejection of the out‐of‐band noise that folds into the useful signal bandwidth because of the decimation, have the drawback of inserting multiple zeroes in the same positions throughout the stop‐band and to increase the edge‐band attenuation. In this paper, we propose a class of decimation filter architectures composed of a cascade of modified CIC filters, which have higher attenuation of the quantization noise produced by a ΣΔ modulator around the folding bands and lower passband drop than classic CIC structures. The design criteria of the proposed filters are described, with the goal of maximizing the denoising effect and minimizing the passband drop, and the problem of the practical realization of the proposed decimation scheme is addressed. Simulations confirm the effectiveness of the proposed decimation filter architectures. Copyright © 2002 John Wiley & Sons, Ltd.

The goal of the fourth-generation (4G) wireless systems is to provide broadband access to the mobile users: ‘easier access to faster information services at any place’, a goal that follows the introduction of multimedia services by IMT-2000 systems as, for example, UMTS and cdma2000. The future may provide truly broadband wireless access through such technologies as Mobile Broadband System (MBS) operating in the 40- and 60-GHz bands, interactive broadcast pseudosatellites and wireless access to Local Area Networks (LANs). Advanced antenna configurations and space-time processing leading to Multiple-Input Multiple-Output (MIMO) systems and the use of ever-increasing complex signal processing algorithms will squeeze evermore data through a given window of frequency. For mobility, seamless interoperability is the order of the day, and the wireless devices must be able to engage different systems using such diverse technologies as LMDS and MMDS, DVB, UMTS, IEEE802.11, Hiperlan, Bluetooth, xDSL and others. Within this context, Software Radio (SR) promises to provide the flexibility and the interoperability capability needed for the future reconfigurable communication systems. The term Software Radio SR is used to encompass a wide range of agile, multimode, programmable radio systems that can operate over a wide frequency band and be able to ‘speak’ the language of the different protocols. The evolution in the semiconductor industry leading to constant increase in the clock speeds and increase in device density and the advent of reconfigurable hardware have paved the way to the creation of SR platforms. SR uses a mix of hardware entities for the implementation of radio transceiver functions capable of commanding the Radio Frequency (RF) spectrum. The collection of papers in this special issue encompasses a wide range of scopes and topics. The aim has been to capture a snapshot of the current state of R&D activities in this very fertile field, yet at the same time provide the basis of the why and how of SR through several tutorial papers and example applications. Note that the terms ‘software radio’ and ‘reconfigurable systems’ are essentially synonymous. As is often the case, from time to time the commonly used term changes with the fashion of the day. At the time of this writing, ‘Reconfigurable Systems (RS)’ tend to be more fashionable. The ferment in any area is often driven by economic factors. This is indeed true with RS in which the driving force in several segments of the industry is economics. In mobile radio systems there are significant benefits to the construction of Base Transceiver Stations (BTS) in RS technology. This allows the operator to invest in an adaptable technology that allows it to perform system upgrades, create nodes capable of interoperating between different protocols or even completely change the communication protocol without the necessity of investing in a completely new hardware infrastructure. In the automotive industry where the role of infotainment is becoming ever more important, the RS technology can offer significant cost benefits and a whole array of location-dependent services that could not otherwise be offered, or offered at a significantly higher cost to the customer. The list goes on and the trend in migration from fixed architecture systems to RS continues. The papers in this special issue are of three basic categories: Tutorial (T), Application-oriented (A) and Research papers (R). We shall use the appropriate symbol when introducing the papers in what follows. The paper by Noll et al. (T) provides a perspective on the system requirements for supporting future-generation protocols and services from an operator point of view. The paper presents a scenario of the potential mobile wireless services demanded by the user on the basis of his/her profile, the key element being the ability to roam and interoperate in a heterogeneous network of diverse devices supporting a variety of protocols yet receiving a user-defined set of services that are transparent to the user regardless of location and access protocol. In this vision, the RS terminal is a central element. The authors then draw an evolutionary path from the fixed architecture hardware devices supporting single protocols to truly universal RS terminals. The paper by Cianca et al. (T) presents an update on the status of R&D and the current thinking of how the somewhat inflexible networks of today may evolve to support the services demanded of the future generation of wired and wireless communication systems. In particular, the paper delves into the concepts of adaptivity at the physical layer of the protocol stack via adaptive coding, modulation and Automatic Repeat Request (ARQ), and presents the argument for the need of an across-layer optimization that addresses the interplay of the physical and the upper layers of the protocol stack to support the future generation of services more efficiently and cost-effectively. In the paper by Laddomada (T), the author addresses a very interesting issue often overlooked in the literature on RS. In particular, assuming that a universal RS user terminal is available, an interesting issue is how would such a universally programmable unit receive its program and operational software? Or for that matter, what typology of software blocks, routines and so on should be defined for a universally programmable device? The paper addresses both the issues of a potential software architecture for such a programmable unit and looks at different mechanisms of software download, providing a qualitative comparison of various techniques. The paper by Dovis et al. (A) looks at a very interesting application of the RS to the design and implementation of an interoperable localization receiver. The Galileo system that is currently being developed in Europe is essentially envisioned to provide precise localization services for commercial applications. It is to be an alternative to the Global Positioning System (GPS), which is controlled by the United States. GPS cannot offer a reliable localization service owing to the fact that it may be occasionally inaccessible to the civilian users at times of crisis or as needed by the US military. Hence, no major commercial enterprise has embraced its use to offer location-dependent services. The paper presents details of the architecture and algorithms used for the implementation of a prototype receiver implemented using reconfigurable logic and Digital Signal Processing (DSP) boards. Such a receiver would be capable of interoperating between GPS and Galileo systems, in addition to offering the possibility of integration of various localization signals to improve performance. Simulation results obtained with the prototype receiver are also presented in the paper. In the paper by Mondin et al. (A), the authors present another very interesting application of the RS technology to the design and implementation of a Satellite and/or Unmanned Aerial Vehicle (UAV) transponder or signal processing payload. The use of RS in such applications offer significant advantages in comparison to the traditional fixed architecture systems. Current trends in satellite technology are toward implementation of smaller, cheaper and lighter mini- and micro-satellites that can significantly reduce the upfront costs associated with the platform launch into the orbit. In addition, there is a steady move toward the use of Commercial Off The shelf (COTs) and Commercial Avionics Military (CAM) hardware for construction of the satellite payload, as opposed to the use of radiation-hardened, often very expensive hardware that technologically lag behind their terrestrial counterparts by several device generations. These trends in addition to the significant cost benefits offered by reconfigurable payloads favor the eventual deployment of the RS technology in the space sector. The paper by Falletti et al. (R) is a research paper in the general area of MIMO systems. In particular, the paper looks at a promising new blind technique for spatiotemporal signal processing of Direct Sequence (DS) Code Division Multiple Access (CDMA) signals. The CDMA technology in addition to multicarrier modulation based on Orthogonal Frequency Division Multiplexing (OFDM) have become the de facto multiple access technologies of choice for many wireless applications. This is due to their inherent robustness to time-varying multipath channels and flexibility of use for offering a variety of services with diverse data rate requirements. The most common architecture for the CDMA receiver utilizes a RAKE receiver, whereby the multipath components of the desired signal are coherently combined at the receiver. The paper looks at a novel solution to the Multiple Access Interference (MAI) problem by combining blind adaptive beamforming techniques, with RAKE processing in an architecture resembling a wideband beamformer. Simulation results are presented demonstrating the potential gains of this promising technique. Finally, the paper by Daneshgaran and Laddomada (R) looks at a novel decimation filter design technique for Sigma-Delta (Σ-Δ) Analog to Digital Converters (ADC). The creation of a true RS unit requires moving the analog-to-digital boundary of the system as close to the antenna as possible. This way, digital processing can begin as early as possible in the processing chain, providing the greatest degree of flexibility in the reconfigurable device. Unfortunately, direct digitization of the RF signals has a significant power penalty in addition to placing great demands on the ADC. Yet, direct digitization of the RF signal is a waste of system resources since the intelligence signal often occupies a narrow band of frequencies around the RF carrier. The Σ-Δ ADC architecture offers the greatest promise in terms of providing the possibility of direct digitization of the intelligence signal around the RF carrier frequency (i.e. the process of frequency translation is embedded in the system architecture). A key element in the design of such converters is the rate-conversion filter, which is responsible for shaping the output noise spectrum and rejection of the out-of-band noise. This research paper presents a novel hardware efficient architecture for a class of rate-conversion filters suitable for the design of wideband Σ-Δ ADCs. Enjoy the special issue!!

We present a systematic technique for obtaining all the input sequences that are mapped by a given permutation either to themselves or to shifted versions of themselves (generically called permutation fixed points). Such sequences or their subsets, represent the primary candidates for examination in connection with obtaining estimates of the minimum distance of parallel concatenated codes, specially for interleaver lengths for which the determination of the actual minimum distance may be very difficult. Subsequently, we present a new class of permutations that nearly achieve the lower bound on the number of possible fixed points associated with a given permutation of prime length p. Preliminary experimental evidence suggests that certain permutations of this class lead to turbo codes with large minimum distances fur short interleaver lengths.

The use of orthonormal scaling functions, wavelets and wavelet packets for modulation has recently been proposed. In this paper, we study the performance of different families of wavelets, for modulation over FDM nonlinear satellite channels, and we propose the use of a splitting technique to obtain two-channels modulation schemes with high spectral efficiency. Simulation results indicate that in certain cases the wavelet based shaping pulses could out-perform the traditional techniques in extremely critical transmission conditions, such as interchannel interference impaired satellite transmission.

This paper addresses the problem of designing interleavers for parallel concatenated convolutional codes (PCCCs) tailored to specific constituent codes. We start by establishing the role of the interleaver in the PCCC and the various parameters that influence the performance of the PCCC with a given interleaver. Subsequently, we define a canonical form of the interleaving engine denoted as the finite-state permuter (FSP) and demonstrate the minimal delay property of this canonical form. For any given permutation, we present a procedure for deriving the canonical FSP engine. We address the issue of implementation of the FSP and propose a very simple structure for the FSP. Next, using the structural property of the FSP engine, we develop a systematic iterative technique for construction of interleavers with a complexity that is polynomial in the interleaver size. Subsequently, we develop a cost function that, coupled with the iterative interleaver growth procedure, can be used to design optimized interleavers for PCCCs. We provide examples of application of the interleaver design technique, and compare the designed interleavers with some of the interleavers of comparable size found in the literature.

This paper addresses two issues in clock synchronization in the context of the theory of wavelets: design of prefilters eliminating pattern-dependent jitter (PPJ) and design of low-jitter shaping pulses for timing recovery. We derive low-jitter symmetric and time-limited shaping pulses via a perturbational technique from the autocorrelation of the Daubechies (1992) scaling function.

We present a procedure to design maximum likelihood (ML) decoders for the new class of geometrically uniform (GU) trellis coded modulation (TCM) codes, exploiting the algebraic properties of such codes. The proposed design has a very efficient VLSI implementation. The design of the decoders for the GUTCM codes is more complicated in comparison to the standard convolutional codes because between any pair of states in the trellis diagram of a GUTCM code, there is usually a large number of parallel transitions, and the trellis diagram of the code has a much higher degree of connectivity in comparison to binary convolutional codes. We present a novel technique for solving the parallel transitions using the algebraic structure of the GUTCM codes, which represents a significant reduction in complexity in comparison to the direct approach. The proposed technique is applied to the design of a simplified Viterbi decoder (VD) for a 64-state nonbinary GUTCM code defined over (Z/sub 8/)/sup 4/. For this example, we obtain a 58 fold reduction in complexity for the parallel transition solver in comparison to a direct implementation.

The paper addresses in depth the system tradeoff issues of the VLSI implementation of long constraint length and/or high rate Viterbi decoders (VDs) and presents the iterative collapse algorithm (ICA) for the systematic generation of parallel architectures for the VD. The ICA is a specific form of partitioning the trellis diagram of the encoder using its inherent symmetries and permits the optimal down scaling of the design to the point where a single-chip VD is feasible. The effectiveness of the ICA is demonstrated by obtaining better than linear tradeoff between throughput and complexity for a wide range of complexity reduction factors. The existence of isomorphic topologies for the VDs associated with encoders having different memories derived through the application of the ICA permits the design of programmable decoders. Programmable decoders in turn can be used in a portable communication environment to provide power control for the portable transceiver depending on the channel state and the required data rate. The feasibility of the design of single-chip VDs leads to a significant power saving in comparison to the multi-chip design option.>

For pt.I see ibid., vol.43, no.2/3/4, p.1409-18 (1995). Part I presented the iterative collapse algorithm (ICA) for obtaining a variety of throughput-complexity tradeoffs for the implementation of parallel VDs and demonstrated that better than linear tradeoff can be obtained in this respect for many combinations of the design parameters M and K. Part I also demonstrated that when (MK+1)/spl les//spl nu//spl les/(M+1)K the ACSUs in the design are fully connected. In this case, clustering of the ACSUs makes no sense since it produces a single cluster containing all the ACSUs. The direct implementation of the decoder requires the full interconnection of L/spl les/2/sup K/ ACSUs (i.e., the ACSU topology is that of a fully connected graph on L nodes denoted K/sub L/). For K/spl ges/3 this is a rather challenging task. The present authors provide efficient means of emulating the ACSU interconnections on a linear or a mesh topology array. Such networks are more suitable for implementation in VLSI. The results presented are quite general and may be easily extended to the emulation of other networks with lower degrees of connectivity than that of the complete graph. The technique presented can be used to build programmable decoders for convolutional and trellis codes in addition to making it possible to build multi-chip VDs on a printed circuit board or as a multi chip module.>