Molecular Communication (MC) is a nanoscale interaction paradigm inspired by the natural ability of cells in biology to communicate through the processing, exchange, and transduction of information by biochemical reactions of molecules. The design and modeling of MC systems is the first step towards future applications based on the engineering of communication systems in biology. In this paper, a bandpass filter based on a specific type of biochemical reactions in cell communication, namely, signaling kinase cascade, is proposed for an MC receiver in a diffusion-based MC scenario. In particular, under the commonly accepted weak activation assumption, these biochemical reactions can be analytically modeled through linear systems theory. The characterization of the proposed filter, and the corresponding numerical results, demonstrate its passband properties, and its suitability for extracting signals in different frequency bands coming from different molecular transmitters.

Novel emerging tools are allowing the manipulation and control of biological cells and their functions, e.g., sensing, actuation, and communication through biochemical stimuli. These tools have the potential to enable the implementation of manmade networks of biological computing devices, i.e., Internet of Bio-things. In this work, signal transduction pathways, i.e., cells' chemical reactions that process biochemical signals, are proposed for the design of analog linear filters to be utilized as components in the Internet of Bio-things. These filters, which exploit the crosstalk of signal transduction pathways to achieve the desired response, are here modeled and analyzed. The relations between filter properties and biochemical parameters are presented with the goal of designing a notch filter. A preliminary numerical example is also given as proof-of-concept.

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

This paper addresses the design of semi-random, prunable interleavers for parallel concatenated convolutional codes (PCCC). The proposed technique is iterative and is based on the growth of a smaller interleaver up to the desired length N. The optimization is achieved via a minimization using a cost-function strictly related to both the correlation properties of the extrinsic information and the concept of spread of an interleaver. Performance of the designed interleavers are given in terms of bit error rate (BER) and frame error rate (FER). Comparisons are given with respect to other prunable and ad-hoc interleaver design techniques already proposed in literature. The designed interleavers are prunable and have a behavior very similar to the interleavers designed with techniques which maximize the spread of the permutation

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 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.

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 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.

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.

Abstract 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 a new generation of very agile reprogrammable communication systems generically known as Software Radio (SR) platforms. SR uses a mix of fixed and programmable and/or reconfigurable hardware entities for the implementation of radio transceiver functions capable of commanding the RF (Radio Frequency) spectrum used for wireless communications. While many articles have addressed the general architecture and features of such platforms, the literature on the mechanics of reconfiguring such units is rather scarce. This paper addresses the issue of reconfigurability for the SR platform. We look at a variety of techniques that may be employed to reconfigure the SR platform, list advantages and disadvantages of each, and address some of the questions that arise in the context of standardization efforts that may be needed to render such units truly universal in scope and in application. Copyright © 2002 John Wiley & Sons, Ltd.