This paper theoretically analyzes the usage of directional slotted Aloha schemes for managing the peer-to-peer random access in fifth and sixth generation (5G/6G) systems. To this aim, the physical layer is modeled by accounting for interference and noise, while a Markov chain approach is developed to investigate the network behavior in the presence and in the absence of a separate feedback channel, which provides information concerning the success or not of each transmission attempt. Closed-form expressions for the coverage probability and for the transition matrices with and without feedback are derived to then evaluate the corresponding throughput. The analytical results, which are validated by independent Monte Carlo simulations, are used to estimate the impact of the antenna gain, of the burst length, and of the node density on the achievable performance, as well as to discuss the directional random access benefit/complexity tradeoff.

The main concern in nowadays digital communications standards, such as IEEE802.11n, 802.16e (Wi-MAX), 100BaseT Ethernet, and Digital Video Broadcasting (DVB), is the need of adaptive and flexible communication techniques. The request for higher efficiency, both in bandwidth use and power consumption, increases the need for limit-achieving, flexible techniques of channel coding. Low Density Parity Check (LDPC) Codes are very interesting because of their high performances and potential flexibility introduced by puncturing. In this paper, the design of rate-compatible LDPC is addressed. A class of punctured LDPC codes is defined through their equivalent parity check matrices. A formal analysis is provided, based on a simplified approach on the decoding belief propagation algorithm, i.e., considering a Gaussian approximation for message densities under density evolution. A suitable design criterion for the puncturing patterns is then addressed.

In this paper we investigate the properties of the generator matrices of a new class of convolutional codes, called product convolutional codes, which were previously defined and investigated by the authors. The new codes are constructed using the well-known method of the direct product for combining block codes. Convolutional codes are considered as block codes over the field of rational functions F(D). The description of convolutional codes as block codes allows the successful application of the direct product method to convolutional codes, and, in addition, leads to a general method to construct new convolutional codes based on already known combining methods for block codes. Expressions for the generator matrices of the product convolutional codes are given and several of their properties, which were not addressed before, are determined. The relationship between the properties of the direct product encoder generator matrix and the properties of the vertical and horizontal constituent encoders generator matrices is derived. Rational generator matrices, as well as polynomial generator matrices, are addressed.

It has been widely observed that there exists a fundamental tradeoff between the minimum (Hamming) distance properties and the iterative decoding convergence behavior of turbo-like codes. While capacity-achieving code ensembles typically are asymptotically bad in the sense that their minimum distance does not grow linearly with block length, and they therefore exhibit an error floor at moderate-to-high signal-to-noise ratios, asymptotically good codes usually converge further away from channel capacity. In this paper, we introduce the concept of tuned turbo codes, a family of asymptotically good hybrid concatenated code ensembles, where asymptotic minimum distance growth rates, convergence thresholds, and code rates can be tradedoff using two tuning parameters:$\lambda $and$\mu $. By decreasing$\lambda $, the asymptotic minimum distance growth rate is reduced in exchange for improved iterative decoding convergence behavior, while increasing$\lambda $raises the asymptotic minimum distance growth rate at the expense of worse convergence behavior, and thus, the code performance can be tuned to fit the desired application. By decreasing$\mu $, a similar tuning behavior can be achieved for higher rate code ensembles.

In this paper we extend the algorithms for the termination and tailbiting of direct product convolutional codes, that we proposed in a previous paper, to the rate-k/n case. There, the relationship between the direct product encoder state sequence and the parameters of the vertical and horizontal constituent encoders was derived, and, given a generic information sequence, it was shown how to find the terminating sequence for the direct product encoder of Rate R = 1/n. An algorithm was also proposed for tailbiting rate-1/n direct product convolutional codes. In this paper, we investigate on the tailbiting failure conditions, which were not addressed before, and extend the proposed tailbiting algorithm to the general rate-k/n case.

In this paper, a novel class of serially concatenated convolutional codes (SCCCs) is addressed. In contrast to standard SCCCs, where high rates are obtained by puncturing the outer code, the heavy puncturing is moved to the inner code, which can be punctured beyond the unitary rate. We derive analytical upper bounds on the error probability of this code structure by considering an equivalent code construction consisting of the parallel concatenation of two codes, and address suitable design guidelines for code optimization. It is shown that the optimal puncturing of the inner code depends on the outer code, i.e., it is interleaver dependent. This dependence cannot be tracked by the analysis for standard SCCCs, which fails in predicting code performance. Based on the considerations arising from the bounds analysis, we construct a family of rate-compatible SCCCs with a high level of flexibility and a good performance over a wide range of code rates, using simple constituent codes. The error rate performance of the proposed codes is found to be better than that of standard SCCCs, especially for high rates, and comparable to the performance of more complex turbo codes.

This paper applies three low complexity methods, previously proposed by the authors, to estimate the distortion deriving from packet losses in error-prone video communication systems. In a first application scenario, the distortion estimated by the encoder is encapsulated in each video packet and used as side information by a modular router, in which the architecture is properly modified for taking into account the distortion impact produced by the loss of each packet. Computational complexity considerations, compared with traditional rate distortion optimization (RaDiO) techniques, are also discussed, showing that the proposed distortion estimation algorithms may be easily used and integrated in a wireless router with limited CPU resources. In the paper a multi-hop scenario is also considered as possible application environment, in which the prioritization and selective discarding policy are based on the estimated distortion and on the loss pattern deriving form previous hop transmissions.

This article proposes three methods to estimate the distortion deriving from packet losses in wireless video communication. The proposed methods take into account the short-term properties of the encoded video sequences. A suitable set of functions is adopted to model the distortion envelope resulting from multiple losses. The estimated performance is compared with the actual distortion, evaluated by decoding the received sequence with a properly designed decoder. Numerical results confirm the accuracy of the proposed models in approximating the actual Mean Square Error (MSE) for a wide range of loss rates. Some applications of the proposed algorithms are presented.

This paper proposes three methods to estimate the distortion deriving from packet losses in wireless video communication. The proposed models take into account the short-term properties of the encoded video sequences and the properties of the wireless links. A suitable set of functions is adopted to model the distortion envelope resulting from multiple losses. The performance of the employed models, which depends on the average packet loss rate (PLR), is compared with the actual distortion, evaluated by decoding the received sequence with a properly designed, robust decoder. Numerical results confirm the accuracy of the proposed models, that are able to closely approximate the actual mean square error (MSE) behavior for a wide range of loss rates. Some applications of the proposed algorithms are also presented in the paper.

In this paper, we propose and evaluate compatible puncturing schemes for turbo codes for Forward Error Correction / Automatic Repeat Request (FEC/ARQ) algorithms, in which the data transmission is supposed to go on progressively. The aim is 1) to obtain a better performance w.r.t. hybrid ARQ schemes already proposed in the literature when the user has to cope with some random packet erasures due, e.g., to deep fading conditions on wireless links, or to congestion on wired networks, i.e., when the number of lost data packets (erasures on an erasure channel) is hard to predict; 2) to obtain a performance comparable to one of the most efficient hybrid ARQ schemes proposed in the literature when all the packets are available to the final user, even if seriously corrupted, as on a pure wireless link. To obtain an unbiased comparison of the different schemes, an analytical method based on the sphere-packing bound is employed. To assess the accuracy of the bound, simulation results are also presented.

We provide a performance analysis of a new class of serial concatenated convolutional codes (SCCC) where the inner encoder can be punctured beyond the unitary rate. The puncturing of the inner encoder is not limited to inner coded bits, but extended to systematic bits. We derive analytical upper bounds to the error probability of this particular code structure and address suitable design guidelines for the inner code puncturing patterns. We show that the proportion of systematic and parity bits to be deleted strongly depends on the SNR region of interest. Furthermore, we show that puncturing of the inner code systematic bits should be interleaver dependent. Based on these considerations, we derive design guidelines to obtain well-performing rate-compatible SCCCs families. Throughout the paper, the performance of the proposed codes are compared with analytical bounds, and with the performance of PCCC and SCCC proposed in literature

In this paper, we introduce the concept of nonsystematic turbo codes and compare them with classical systematic turbo codes. Nonsystematic turbo codes can achieve lower error floors than systematic turbo codes because of their superior effective free distance properties. Moreover, they can achieve comparable performance in the waterfall region if the nonsystematic constituent encoder has a low-weight feedforward inverse. A uniform interleaver analysis is used to show that rate R=1/3 turbo codes using nonsystematic constituent encoders have larger effective free distances than when systematic constituent encoders are used. Also, mutual information-based transfer characteristics and extrinsic information transfer charts are used to show that rate R=1/3 turbo codes with nonsystematic constituent encoders having low-weight feedforward inverses achieve convergence thresholds comparable to those achieved with systematic constituent encoders. Catastrophic encoders, which do not possess a feedforward inverse, are shown to be capable of achieving low convergence thresholds by doping the code with a small fraction of systematic bits. Finally, we give tables of good nonsystematic turbo codes and present simulation results comparing the performance of systematic and nonsystematic turbo codes.

Partially systematic rate-compatible serial concatenated convolutional codes (SCCCs) are considered. These are generated from a classical rate 1/3 serial concatenated mother code. To obtain rate-compatible SCCCs, the puncturing is limited to inner coded bits, by puncturing both the inner code's parity and systematic bits. It is shown that the performance can be enhanced in the "waterfall" or in the "error-floor" region, by simply spreading the puncturing over the parity and the systematic bits without increasing encoding and decoding complexity.

A new design criterion to obtain well performing rate-compatible serial concatenated convolutional codes (SCCC) families is proposed. To obtain rate-compatible SCCCs, the puncturing is limited to inner coded bits. However, the puncturing is not restricted to inner parity bits, but extended also to inner systematic bits, thus obtaining high rate SCCCs (i.e., beyond the outer code rate). The considerations presented in S. Benedetto, et al., (1998) to find "best" component encoders for SCCCs construction are extended to find the "best" rate-compatible puncturing patterns for a given input decoding delay I. A rate-compatibility restriction to the puncturing rule is used, implying that all the code bits of a high-rate punctured code are used by the lower rate codes. The two main applications of this technique are its use in hybrid ARQ/FEC schemes and to achieve unequal error protection (UEP) of an information sequence.

In this paper, we address the design of high rate turbo codes using high rate nonsystematic constituent encoders and compare their distance and iterative decoding convergence properties with systematic turbo coding schemes.

In this letter, we propose and compare some design criteria for the search of good rate-compatible systematic turbo codes (RCPTC) families. The considerations presented by Benedetto et al. to find "best" component encoders for turbo-code construction are extended to find good rate-compatible puncturing patterns leading to codes with promising performances.

We propose and compare some good rate-compatible serial concatenated convolutional code (SCCC) families. To obtain rate-compatible SCCCs, the puncturing is limited to inner coded bits. However, and this is the novelty proposed, we do not limit the puncturing to inner parity bits only, but we extend it also to inner systematic bits, thus obtaining higher rate SCCCs (i.e., beyond the outer code rate). The two main applications of this technique are its use in hybrid ARQ/FEC schemes and to achieve unequal error protection (UEP) of an information sequence.

The performance of turbo codes is evaluated through analysis and simulation over the Rice multiplicative fading channel. The analysis is conducted extending the results presented by A.M. Viterbi and A.J. Viterbi (see Proc. ISIT'98, p.29, 1998) to the Rice multiplicative fading channel case. Simulation results are obtained using an extended version of the improved Jakes' channel simulator, first presented by M.F. Pop and N.C. Beaulieu (see IEEE Trans. Commun., vol.49, p.699-708, 2001).

In this paper, the achievable performance of a turbo coded system adapted to a block fading channel model is evaluated. This model is suitable for analyzing, for instance, frequency-hopped multiple access (FHMA) systems that operate in a mobile satellite environment characterized by frequency-nonselective slow Rician fading, provided that the spacing between carriers is larger than the coherence bandwidth, resulting in basically uncorrelated blocks. In such systems, coded information is transmitted over a small number of fading channels in order to achieve diversity. The best coded information allocations over a certain number of fading channels are evaluated. Bounds on the achievable performance due to coding are derived using information-theoretic techniques. Moreover, simulation results are obtained and compared with the theoretical ones.

We propose and compare some design criteria for the search of good rate-compatible punctured turbo (RCPT) code families. The considerations presented by S. Benedetto et al. (see IEEE Trans. on Commun., vol.46, no.9, p.1101-5, 1998) to find "best" component encoders for turbo code construction are extended to find the "best" rate-compatible puncturing patterns for a given interleaver length k. This approach is shown to be the optimal one, both in the maximum-likelihood sense (using transfer function bounds) and in the iterative decoding sense (through simulation results). A rate-compatibility restriction to the puncturing rule is used, implying that all the code bits of a high-rate punctured code are used by the lower rate codes. The two main applications of this technique are its use in modified type II ARQ/FEC schemes and to achieve unequal error protection of an information sequence.

In this work, the performance of turbo codes over the Rice multiplicative fading channel is evaluated through analysis and simulation. The analysis is conducted through improved upper bounds on bit error probabilities in conjunction with both the fully interleaved channel and the correlated Rice slow fading channel. The analytical expression of the improved bound is obtained by extending the results of Viterbi and Viterbi (see Proc. IEEE International Symposium on Information Theory, p.29, 1998) to the Rice multiplicative fading channel case. Simulation results are obtained and presented also for intermediate fading rates, ranging between the two extreme multipath fading rates considered in the analysis. Simulations are performed using the improved Jakes' channel simulator to simulate Rayleigh fading channels, and extended to adequately model the multipath Rice fading propagation environment.

This paper investigates the performance of hybrid ARQ schemes for the mobile low Earth orbit (LEO) satellite channel. The proposed schemes adapt their characteristics to the time-varying behavior of the channel. To obtain an unbiased comparison of different ARQ schemes a theoretical viewpoint is adopted. More precisely, the sphere-packing bound is used to evaluate the best possible performance of a hybrid ARQ scheme. To assess the correctness and the accuracy of the analysis, simulation results, based on the use of turbo codes, are also presented.

As a powerful coding technique, turbo codes are a prime candidate for improving the reliability of communication over wireless channels, where fading is the main impairment. However, to date, only limited attention has been given to the performance of turbo codes on fading channels, and the influence of multipath fading has been fairly accurately investigated as far as the Rayleigh fading distribution is concerned only. Therefore, it is the purpose of this work to conduct a study of turbo codes performance considering the more general Rice fading case: the study is performed in conjunction with both the fully-interleaved channel and the correlated Rice slow-fading channel.

The bit-stream generated by most source coding algorithms, for voice, audio and images, usually consists of bits of different significance and error sensitivity. By exploiting this property, in the design of a wireless communication system, it is possible to obtain significant advantages. First, the network capacity can be increased, with a smoother quality degradation under heavy traffic or severe interference conditions. Secondly, the application of source sensitivity matched FEC (forward error correction) protection schemes, leads to a higher robustness of source bits to channel impairments. We evaluate the usefulness of prioritized, variable-rate speech encoding schemes, for handling congestion over packet switching systems. The system performance is shown to be sensitive to the fraction of information placed in each priority and to the number of priorities. The applicability of source-sensitivity matched error protection schemes, for variable-rate speech transmission on a radio channel, is also investigated.

Packet switching allows one to exploit the intrinsic variable-rate behaviour of real-time information, such as voice and video. Indeed, packet switching permits one to handle in a more flexible way the information bits and to obtain, therefore, a significant increment in system capacity. A relative measure of the capacity increment we obtain over circuit switched systems is the multiplexing gain. This paper presents preliminary results of the multiplexing gain we can achieve when we transmit multi-mode and variable-rate speech on a slotted and framed radio channel. For our evaluation we use an embedded, variable bit-rate speech coding scheme, namely the Mobile Audio Visual Terminal (MAVT) coder, candidate to the new standard MPEG-4. The peculiar characteristics of this coding scheme can be exploited for obtaining a significant multiplexing gain. Moreover, at high loads, a smooth and graceful degradation of speech quality can be obtained by exploiting the embedded property of the encoder. In fact, being that the information bits are classified by priority, we observe a graceful quality reduction during network congestion, if the transmission algorithm is designed to discard first the bits with the minimum impact on the output quality.