Visible Light Positioning and Communication (VLPC) is a promising candidate for implementing Integrated Sensing And Communication (ISAC) in the unlicensed 400 THz to 800 THz band. The current Visible Light Positioning (VLP) systems mainly operate based on the Received Signal Strength (RSS) of the Line-of-Sight (LoS) path. However, its accuracy is degraded by interferences from Non-LoS (NLoS) paths. Furthermore, in Visible Light Communication (VLC) systems, the estimation of Channel State Information (CSI) also becomes challenging, when the optical channel becomes dispersive. Against this background, we propose a new VLPC scheme using Direct Current (DC) biased Optical Orthogonal Frequency-Division Multiplexing (VLPC-DCO-OFDM), where OFDM-based sensing is applied for the sake of improving the resolution of the estimated Channel Impulse Response (CIRs) exploited for positioning functionality. The CIRs estimated by sensing are further exploited to provide enhanced CSI for communication data detection. Moreover, we propose a hybrid Radar-RSS based solution, where the conventional RSS-aided VLP method is invoked for the sake of refining OFDM radar. Our simulation results demonstrate that the proposed VLPC-DCO-OFDM scheme – which simultaneously supports the triple functionalities of illumination, bi-static sensing and communication – is capable of achieving centimeter-level positioning accuracy and Giga-bits-per-second data rate.

We demonstrate that the proposed optical orthogonal time frequency space (O-OTFS) is capable of improving the bandwidth-/power-/energy-efficiencies of optical orthogonal frequency-division multiplexing (O-OFDM). The bandwidth-efficiency is improved because only a single cyclic prefix (CP) is needed for an entire O-OTFS frame. The power-efficiency is enhanced thanks to the diversity gain achieved by its symplectic finite Fourier transform (SFFT), which also leads to a reduced peak-to-average power ratio (PAPR), hence improving its energy-efficiency. These features are facilitated by the proposed layered asymmetrically clipped O-OTFS (LACO-OTFS), which is capable of removing the direct current (DC) bias while retaining the full optical throughput. Nonetheless, there exists an inherent trade-off, where increasing the O-OTFS frame size leads to a commensurately reduced CP percentage at the cost of an increased PAPR. In order to mitigate this, we propose to perform discrete Fourier transform based spreading (DFT-S) in the delay-Doppler (DD)-domain. Furthermore, we demonstrate that regardless of the choice of domain in which the information is modulated (i.e. O-OFDM/O-OTFS with/without DFT-S), the frequency-selectivity of the quasi-static but dispersive optical channel can always be equalized by single-tap frequency-domain equalization (FDE). Moreover, the channel estimation techniques are conceived to operate in the time-/frequency-/DD-domains for both O-OFDM and O-OTFS. Our simulation results demonstrate that for a multi-user optical wireless system associated withM= 64 subcarriers and the OTFS frame length ofN= 64, LACO-OTFS is capable of achieving a 7 dB power-efficiency gain over LACO-OFDM, where the CP overhead is reduced by a factor ofN= 64. DFT-S-LACO-OTFS is also capable of providing a 7 dB power-efficiency gain over DFT-S-LACO-OFDM, where the low PAPR of single-carrier transmission is retained.

In order to maximize the spectral efficiency (SE) in multicarrier-division duplex (MDD) enabled cell-free massive MIMO (CF-mMIMO), a heterogeneous graph neural network (HGNN), referred to as CF-HGNN, is specifically introduced to optimize the power allocation (PA). To efficiently manage the interference invoked, a meta-path based mechanism is applied in CF-HGNN to enable individual access point (AP) and mobile station (MS) nodes to aggregate information from the interfering and communication paths with different priorities during message passing. Moreover, the proposed CF-HGNN employs the adaptive node embedding layer and adaptive output layer to make it scalable to the various numbers of APs, MSs and subcarriers. For comparison, a quadratic transform and successive convex approximation (QT-SCA) algorithm is proposed to solve the PA problem in classic way. Numerical results show that CF-HGNN is capable of achieving 99% of the SE achievable by QT-SCA but using only 10−4 times of its operation time, and it can outperform the conventional learning-based and greedy unfair methods in terms of SE performance. Furthermore, CF-HGNN exhibits good scalability to the CF networks with various numbers of nodes and subcarriers, and also to the large-scale CF networks when assisted by user-centric clustering.

The recently-developed orthogonal time frequency space (OTFS) modulation is capable of transforming the time-varying fading of the time-frequency (TF) domain into the time-invariant fading representations of the delay-Doppler (DD) domain. The OTFS system using orthogonal frequency-division multiplexing (OFDM) as inner core naturally requires the subcarrier spacing (SCS) Δfto be larger than the maximum Doppler frequency ϑmax, i.e. Δf> ϑmax, when perfect channel state information (CSI) knowledge is assumed. However, for the first time in literature, we explicitly demonstrate that the practical OFDM-based OTFS systems have to double their SCS in order to facilitate CSI estimation, requiring Δf′ = 2Δf> 2ϑmax. In order to mitigate this loss, we propose a novel noncoherent OTFS system, which is capable of operating at Δf> ϑmax. The major challenge in this context is the mitigation of the DD-domain interference without CSI. Against this background, we draw an analogy between the input-output model of OTFS and that of V-BLAST, where V-BLAST’s blind inter-antenna interference mitigation technique is invoked. Moreover, we propose to partition the DD-domain modulated symbols into groups, where space-time block coding is invoked in order to eliminate the DD-domain interference within each group. Our simulation results demonstrate that the proposed noncoherent OTFS is capable of substantially outperforming its coherent counterparts relying on CSI estimation.

The recently developed reconfigurable intelligent surfaces (RISs) are capable of improving the coverage of space–air–ground integrated networks (SAGINs), where the signals can be reflected in the desired direction without relying on power-thirsty radio-frequency (RF) chains. However, in the face of the substantially increased Doppler frequency, the classic orthogonal frequency-division multiplexing (OFDM) becomes inadequate in supporting RIS for the following reasons. First, the detrimental doubly selective fading leads to intersymbol interference (ISI) and intercarrier interference (ICI), which result in error floors for OFDM operating in the time–frequency (TF) domain. Second, it is far from trivial to configure RIS based on the time-varying fading channels. Third, the interpolation-based TF-domain channel estimation methods become impractical for the high-Doppler and high-dimensional RIS systems. Against this background, in this article, we propose the powerful 2-D orthogonal time–frequency space (OTFS) modulation for RIS-aided SAGINs, which transforms the time-varying fading encountered in the TF-domain to the time-invariant fading in the delay-Doppler (DD) domain. More explicitly, first, for the first time in the literature, we devise the DD-domain channel model of RIS-assisted SAGINs in the face of doubly selective fading. Second, in order to facilitate the RIS configuration in the DD-domain, we propose to create “virtual” Doppler frequencies that guide the phase changes at the RIS, even though the RIS phase rotations do not suffer from Doppler effects. Third, we conceive an attractive DD-domain RIS channel estimation method that can support both OFDM and OTFS, where the TF-domain interpolation is eliminated. Our simulation results demonstrate that the proposed DD-domain RIS configuration and channel estimation methods for both OFDM and OTFS are capable of mitigating the error floors encountered in the TF-domain. Furthermore, our simulation results confirm that OTFS-based RIS-assisted SAGIN systems are capable of outperforming their OFDM counterparts and exhibit excellent performance across a wide range of SAGIN channel parameters including the Ricean K factor, Doppler frequency, delay spread, coverage distance, and carrier frequency.

Reconfigurable intelligent surface (RIS) is capable of improving the wireless system performance by steering the reflected signal in the desired direction. One of the major challenges is that both the transceiver and RIS have to be jointly optimized, where the optimization problems have to be reformulated for different system models and scenarios. To circumvent this challenge, new low-complexity antenna selection (AS) algorithms for transceiver-agnostic RIS configuration are proposed. Given a multiple-input multiple-output (MIMO) channel, the proposed RIS-AS opts for accurately aligning the RIS both with the transmit antenna (TA) and receive antenna (RA) for the sake of maximizing the MIMO channel’s overall output power. The proposed RIS-AS only has to configure the RIS alone, i.e. without iterations with the transceiver optimization. As a result, the proposed RIS-AS has the compelling benefit that they are generically applicable, regardless of the specific transceiver architecture. Our simulation results confirm that the proposed RIS-AS is capable of supporting any MIMO configuration, regardless of their closed/open-loop, single-/ full-RF and multiplexing-/diversity-oriented setups.

Data packet routing in aeronauticalad hocnetworks (AANETs) is challenging due to their high-dynamic topology. In this article, we invoke deep learning (DL) to assist routing in AANETs. We set out from the single objective of minimizing the end-to-end (E2E) delay. Specifically, a deep neural network (DNN) is conceived for mapping the local geographic information observed by the forwarding node into the information required for determining the optimal next hop. The DNN is trained by exploiting the regular mobility pattern of commercial passenger airplanes from historical flight data. After training, the DNN is stored by each airplane for assisting their routing decisions during flight relying solely on local geographic information. Furthermore, we extend the DL-aided routing algorithm to a multiobjective scenario, where we aim for simultaneously minimizing the delay, maximizing the path capacity, and maximizing the path lifetime. Our simulation results based on real flight data show that the proposed DL-aided routing outperforms existing position-based routing protocols in terms of its E2E delay, path capacity, as well as path lifetime, and it is capable of approaching the Pareto front that is obtained using global link information.

An edge-cancellation-aided iterative detection and decoding (EC-IDD) algorithm is proposed for polar-coded sparse code multiple access (SCMA), which jointly performs Gaussian-approximated message passing (GA-MP) detection of SCMA supported by the soft list decoding (SLD) of polar codes. A reduced-edge factor graph is formulated in each consecutive iteration with the aid of the cyclic redundancy check (CRC) and EC. Based on the simplified factor graph, the EC-IDD gradually reduces its complexity in each subsequent iteration, while improving the bit error rate (BER) performance, compared to the state-of-the-art joint detection and decoding (JDD) of polar-coded SCMA. Furthermore, an embedded decision-directed channel estimator (DD-CE) is proposed for our polar-coded SCMA system under realistic imperfect channel state information (CSI). Our simulation results demonstrate that the proposed EC-IDD achieves better BER performance than the state-of-the-art JDD under both perfect and imperfect CSI, despite achieving a complexity reduction of 92%. Finally, the BER of the proposed joint DD-CE and EC-IDD algorithm under imperfect CSI converges to that of EC-IDD operating under perfect CSI.

The secrecy capacity based on the assumption of having continuous distributions for the input signals constitutes one of the fundamental metrics for the existing physical layer security (PHYS) solutions. However, the input signals of real-world communication systems obey discrete distributions. Furthermore, apart from the capacity, another ultimate performance metric of a communication system is its symbol error ratio (SER). In this article, we pursue a radically new approach to PHYS by considering rigorous direct SER optimization exploiting the discrete nature of practical modulated signals. Specifically, we propose a secure precoding technique based on a multi-objective SER criterion, which aims for minimizing the confidential messages' SER at their legitimate user, while maximizing the SER of the confidential messages leaked to the illegitimate user. The key to this challenging multi-objective optimization problem is to introduce a priority factor that controls the priority of directly minimizing the SER of the legitimate user against directly maximizing the SER of the leaked confidential messages. Furthermore, we define a new metric termed as the security-level, which is related to the conditional symbol error probability of the confidential messages leaked to the illegitimate user. Additionally, we also introduce the secure discrete-input continuous-output memoryless channel (DCMC) capacity referred to as secure-DCMC-capacity, which serves as a classical security metric of the confidential messages, given a specific discrete modulation scheme. The impacts of both the channel's Rician factor and the correlation factor of antennas on the security-level and the secure-DCMC-capacity are investigated. Our simulation results demonstrate that the proposed priority-aware secure precoding based on the direct SER metric is capable of securing transmissions, even in the challenging scenario, where the eavesdropper has three receive antennas, while the legitimate user only has a single one.

In order to meet the latency requirements of the ultra-reliable low latency communication (URLLC) mode of the third-generation partnership project's long term evolution (LTE) mobile communication standard, this paper proposes a novel turbo decoding algorithm that supports an arbitrarily high degree of parallel processing, facilitating significantly higher processing throughputs and substantially lower processing latencies than the state-of-the-art (SOTA) LTE turbo decoder. As in conventional turbo decoding algorithms, the proposed Arbitrarily Parallel Turbo Decoder (APTD) decomposes each frame of information bits into a sequence of windows, where the bits within different windows are processed simultaneously using forward and backward recursions in a serial manner. However, in contrast to conventional turbo decoding algorithms, the APTD does not require different windows to be composed of an identical number of bits, which allows the use of an arbitrary number of windows and hence an arbitrary degree of parallelism, when decoding information bits of an arbitrary frame length. Furthermore, conventional turbo decoding algorithms alternate between simultaneously processing the windows in the upper decoder and those in the lower decoder. By contrast, the APTD processes the odd-indexed windows in the upper decoder at the same time as the even-indexed windows in the lower decoder and alternates between this and the reversed arrangement, hence further improving the decoding throughput and latency. Furthermore, the APTD achieves a reduced hardware resource requirement by calculating the extrinsic information based only on the outputs of the forward recursions, rather than based on both the forward and backward recursions of conventional turbo decoding algorithms. We demonstrate that the proposed APTD achieves superior latency, throughput, and computational efficiency than the SOTA LTE turbo decoder at all frame lengths, but particularly at the short frame lengths that are typically used in URLLC approaches. For example, at a frame length of N = 504 bits, the proposed APTD achieves an FER of 10-5at the same Eb/N0as I = 8 iterations of a conventional turbo decoder but with a computational efficiency that is 6 times higher than that of the SOTA turbo decoder, while achieving a latency and throughput that are 0.7 and 1.4 times those of the SOTA decoder, respectively.

The birth of wireless communication systems nearly a century ago has transformed and redefined the way humans communicate and interact. This transformation has evolved over many years and has brought along not only seamless connectivity for human interactions but also communication between machines and devices. While these communication systems are manmade artifacts, the research community has more recently turned its attention to other communication strategies that have spontaneously evolved in nature.

The engineering vision of relying on the “smart sky” for supporting air traffic and the “internet-above-the-clouds” for in-flight entertainment has become imperative for the future aircraft industry. Aeronautical ad hoc networking (AANET) constitutes a compelling concept for providing broadband communications above clouds by extending the coverage of air-to-ground (A2G) networks to oceanic and remote airspace via autonomous and self-configured wireless networking among commercial passenger airplanes. The AANET concept may be viewed as a new member of the family of mobile ad hoc networks (MANETs) in action above the clouds. However, AANETs have more dynamic topologies, larger and more variable geographical network size, stricter security requirements, and more hostile transmission conditions. These specific characteristics lead to more grave challenges in aircraft mobility modeling, aeronautical channel modeling, and interference mitigation as well as in network scheduling and routing. This paper provides an overview of AANET solutions by characterizing the associated scenarios, requirements, and challenges. Explicitly, the research addressing the key techniques of AANETs, such as their mobility models, network scheduling and routing, security, and interference, is reviewed. Furthermore, we also identify the remaining challenges associated with developing AANETs and present their prospective solutions as well as open issues. The design framework of AANETs and the key technical issues are investigated along with some recent research results. Furthermore, a range of performance metrics optimized in designing AANETs and a number of representative multiobjective optimization algorithms are outlined.

In this treatise, first of all, we conceive a generic multiple-symbol differential sphere detection (MSDSD) solution for both single- and multiple-antenna-based noncoherent schemes in both uncoded and coded scenarios, where the high-mobility aeronautical Ricean fading features are taken into account. The bespoke design is the first MSDSD solution in the open literature that is applicable to the generic differential space-time modulation (DSTM) for transmission over Ricean fading. In the light of this development, the recently developed differential spatial modulation and its diversity counterpart of differential space-time block coding using index shift keying are specifically recommended for aeronautical applications owing to their low-complexity single-RF and finite-cardinality features. Moreover, we further devise a noncoherent decision-feedback differential detection and a channel-state information estimation aided coherent detection, which also take into account the same Ricean features. Finally, the advantages of the proposed techniques in different scenarios lead us to propose for the aeronautical systems to adaptively: 1) switch between coherent and non-coherent schemes; 2) switch between single- and multiple-antenna-based schemes as well as; and 3) switch between high-diversity and high-throughput DSTM schemes.

This paper proposes a compressive sensing (CS) based active user identification scheme over frequency-selective fading channels. Unlike the conventional cyclic prefix (CP) based preamble transmission, our approach does not utilize CP in order to conserve signaling overhead, in turn reducing the system-wide processing latency. Our approach first estimates the multi-user channel impulse response vectors by solving a mixed ℓ2/ℓ1-norm optimization problem; then, the active users are identified via a sorting of the norms of the estimated channel vectors. By exploiting the Toeplitz channel matrix structure resulting from CP-free preamble transmission, analytic performance guarantee in term of the block restricted isometry property of the preamble matrix is given. Computer simulations are used to illustrate the performance of the proposed method.

We propose a regularized zero-forcing transmit precoding (RZF-TPC)-aided and distance-based adaptive coding and modulation (ACM) scheme to support aeronautical communication applications, by exploiting the high spectral efficiency of the large-scale antenna arrays and link adaption. Our RZF-TPC-aided and distance-based ACM scheme switches its mode according to the distance between the communicating aircraft. We derive the closed-form asymptotic signal-to-interference-plus-noise ratio (SINR) expression of the RZF-TPC for the aeronautical channel, which is Rician, relying on a non-centered channel matrix that is dominated by the deterministic line-of-sight component. The effects of both realistic channel estimation errors and of the co-channel interference are considered in the derivation of this approximate closed-form SINR formula. Furthermore, we derive the analytical expression of the optimal regularization parameter that minimizes the mean square detection error. The achievable throughput expression based on our asymptotic approximate SINR formula is then utilized as the design metric for the proposed RZF-TPC-aided and distance-based ACM scheme. Monte-Carlo simulation results are presented for validating our theoretical analysis as well as for investigating the impact of the key system parameters. The simulation results closely match the theoretical results. In the specific example that two communicating aircrafts fly at a typical cruising speed of 920km/h, heading in opposite direction over the distance up to 740km taking a period of about 24 min, the RZF-TPC-aided and distance-based ACM is capable of transmitting a total of 77 GB of data with the aid of 64 transmit antennas and four receive antennas, which is significantly higher than that of our previous eigen-beamforming transmit precoding-aided and distance-based ACM benchmark.

An ideal visible light communication (VLC) system should facilitate reliable data transmission at high throughputs, while also providing flicker-free illumination at the user-defined dimming level. In this spirit, we conceive a unary code aided dimming scheme for ON-OFF keying (OOK) modulated VLC systems. The proposed unary-coded scheme facilitates joint dimming and throughput control, while relying on iterative decoding. It is demonstrated that the proposed unary-coded dimming scheme provides attractive throughput gains over its contemporaries and it is also capable of approaching the theoretical throughput limit. Furthermore, we design novel joint dimming-forward error correction coding schemes, which significantly outperform their compensation time dimming-based counterparts in terms of the attainable bit error rate performance as well as the throughput. Finally, in the quest for approaching the capacity, we also optimize our system using EXTRINSIC information transfer charts and demonstrate an SNR-gain of upto 6 dB over the compensation time dimming-based classic benchmarker.

In order to meet the demands of “Internet above the clouds,” we propose a multiple-antenna aided adaptive coding and modulation (ACM) for aeronautical communications. The proposed ACM scheme switches its coding and modulation mode according to the distance between the communicating aircraft, which is readily available with the aid of the airborne radar or the global positioning system. We derive an asymptotic closed-form expression of the signal-to-interference-plus-noise ratio (SINR) as the number of transmitting antennas tends to infinity, in the presence of realistic co-channel interference and channel estimation errors. The achievable transmission rates and the corresponding mode-switching distance-thresholds are readily obtained based on this closed-form SINR formula. Monte-Carlo simulation results are used to validate our theoretical analysis. For the specific example of 32 transmit antennas and four receive antennas communicating at a 5-GHz carrier frequency and using 6-MHz bandwidth, which are reused by multiple other pairs of communicating aircraft, the proposed distance-based ACM is capable of providing as high as 65.928-Mb/s data rate when the communication distance is less than 25 km.

Unary Error Correction (UEC) codes constitute a recently proposed Joint Source and Channel Code (JSCC) family, conceived for alphabets having an infinite cardinality, whilst out-performing previously used Separate Source and Channel Codes (SSCCs). UEC based schemes rely on an iterative decoding process, which involves three decoding blocks when concatenated with a turbo code. Owing to this, following the activation of one of the three blocks, the next block to be activated must be chosen from the other two decoding block options. Furthermore, the UEC decoder offers a number of decoding options, allowing its complexity and error correction capability to be dynamically adjusted. It has been shown that iterative decoding convergence can be expedited by activating the specific decoding option that offers the highest Mutual Information (MI) improvement to computational complexity ratio. This paper introduces an iterative demodulator, which is shown to improve the associated error correction performance, while reducing the overall iterative decoding complexity. The challenge is that the iterative demodulator has to forward its soft-information to the other two iterative decoding blocks, and hence the corresponding MI improvements cannot be compared on a like-for-like basis. Additionally, we also propose a method of eliminating the logarithmic calculations from the adaptive iterative decoding algorithm, hence further reducing its implementational complexity without impacting its error correcting performance.

Unlike a generic PSK/QAM detector, which may visit a constellation diagram only once, a depth-first Sphere Decoder (SD) has to re-visit the same constellation diagram multiple times. Therefore, in order to prevent the SD from repeating the detection operations, the Schnorr-Euchner search strategy of Schnorr and Euchner may be invoked for optimizing the nodes' search-order, where the ideal case is for the SD to visit the constellation nodes in a zigzag fashion. However, when the hard-decision Multiple-Symbol Differential Sphere Detection (MSDSD) of Lampe et al. is invoked for using multiple receive antennas NR≥ 1, the Schnorr-Euchner search strategy has to visit and sort all the MPSK constellation points. A similar situation is encountered for the soft-decision MSDSD of Pauli et al., when the a priori LLRs gleaned from the channel decoder are taken into account. In order to tackle these open problems, in this paper, we propose a correlation process for the hard-decision MSDSD of Lampe et al. and a reduced-complexity design for the soft-decision MSDSD of Pauli et al., so that the Schnorr-Euchner search strategy always opts for visiting the MPSK constellation points in a zigzag fashion. Our simulation results demonstrate that a substantial complexity reduction is achieved by our reduced-complexity design without imposing any performance loss. Explicitly, up to 88.7% complexity reduction is attained for MSDSD (Nw= 4) aided D16PSK. This complexity reduction is quite substantial, especially when the MSDSD is invoked several times during turbo detection. Furthermore, in order to offer an improved solution and a comprehensive study for the soft-decision MSDSD, we also propose to modify the output of the SD to harmonize its operation with the near-optimum Approx-Log-MAP. Then the important subject of coherent versus noncoherent is discussed in the context of coded systems, which suggests that MSDSD aided DPSK is an eminently suitable candidate for turbo detection assisted coded systems operating at high Doppler frequencies.

In this paper, we propose bit-by-bit iterative decoding for expediting the convergence of Repeat Accumulate (RA) decoders. In a conventional RA decoder, the repeat and accumulate component decoders are operated iteratively, to facilitate near-capacity communication. However, whenever one decoder is activated, the other is kept idle. The outputs of the active component decoder are stored until its operation is completed, whereupon the outputs are forwarded to the other decoder all at once and the activation of the decoders is swapped. The proposed bit-by-bit RA decoder expedites this process by allowing both component decoders to operate simultaneously, continuously exchanging outputs without buffering. We present both EXtrinsic Information Transfer (EXIT) charts and Bit Error Ratio (BER) results, which demonstrate that the proposed bit-by-bit RA decoder requires fewer decoding iterations to converge, at the cost of a slightly increased complexity per decoding iteration. Overall, we demonstrate that in a range of practical scenarios, the proposed bit-by-bit RA offers gains of up to 0.86 dB, without imposing any additional decoding complexity and without requiring any additional transmission-energy, -bandwidth or -duration.

This paper proposes a novel alternative to the Logarithmic Bahl-Cocke-Jelinek-Raviv (Log-BCJR) algorithm for turbo decoding, yielding significantly improved processing throughput and latency. While the Log-BCJR processes turbo-encoded bits in a serial forwards-backwards manner, the proposed algorithm operates in a fully-parallel manner, processing all bits in both components of the turbo code at the same time. The proposed algorithm is compatible with all turbo codes, including those of the LTE and WiMAX standards. These standardized codes employ odd-even interleavers, facilitating a novel technique for reducing the complexity of the proposed algorithm by 50%. More specifically, odd-even interleavers allow the proposed algorithm to alternate between processing the odd-indexed bits of the first component code at the same time as the even-indexed bits of the second component, and vice-versa. Furthermore, the proposed fully-parallel algorithm is shown to converge to the same error correction performance as the state-of-the-art turbo decoding algorithm. Owing to its significantly increased parallelism, the proposed algorithm facilitates throughputs and latencies that are up to 6.86 times superior to those of the state-of-the art algorithm, when employed for the LTE and WiMAX turbo codes. However, this is achieved at the cost of a moderately increased computational complexity and resource requirement.

Irregular joint source and channel coding (JSCC) scheme is proposed, which we refer to as the irregular unary error correction (IrUEC) code. This code operates on the basis of a single irregular trellis, instead of employing a set of separate regular trellises, as in previous irregular trellis-based codes. Our irregular trellis is designed with consideration of the UEC free distance, which we characterize for the first time in this paper. We conceive the serial concatenation of the proposed IrUEC code with an irregular unity rate code (IrURC) code and propose a new EXtrinsic Information Transfer (EXIT) chart matching algorithm for parametrizing these codes. This facilitates the creation of a narrow EXIT tunnel at a low Eb/N0value and provides near-capacity operation. Owing to this, our scheme is found to offer a low symbol error ratio (SER), which is within 0.4 dB of the discrete-input continuous-output memoryless channel (DCMC) capacity bound in a particular practical scenario, where gray-mapped quaternary phase shift keying (QPSK) modulation is employed for transmission over an uncorrelated narrowband Rayleigh-fading channel with an effective throughput of 0.508bits-1Hz-1. Furthermore, the proposed IrUEC-IrURC scheme offers a SER performance gain of 0.8 dB, compared to the best of several regular and irregular separate source and channel coding (SSCC) benchmarkers, which is achieved without any increase in transmission energy, bandwidth, transmit duration, or decoding complexity.

In this paper we propose a novel low-complexity Joint Source and Channel Code (JSCC), which we refer to as the Elias Gamma Error Correction (EGEC) code. Like the recently-proposed Unary Error Correction (UEC) code, this facilitates the practical near-capacity transmission of symbol values that are randomly selected from a set having an infinite cardinality, such as the set of all positive integers. However, in contrast to the UEC code, our EGEC code is a universal code, facilitating the transmission of symbol values that are randomly selected using any monotonic probability distribution. When the source symbols obey a particular zeta probability distribution, our EGEC scheme is shown to offer a 3.4 dB gain over a UEC benchmarker, when Quaternary Phase Shift Keying (QPSK) modulation is employed for transmission over an uncorrelated narrowband Rayleigh fading channel. In the case of another zeta probability distribution, our EGEC scheme offers a 1.9 dB gain over a Separate Source and Channel Coding (SSCC) benchmarker.

A joint source-channel coding has attracted substantial attention with the aim of further exploiting the residual correlation residing in the encoded video signals for the sake of improving the reconstructed video quality. In our previous paper, a first-order Markov process model was utilized as an error concealment tool for exploiting the intra-frame correlation residing in the Wyner-Ziv (WZ) frame in the context of pixel-domain distributed video coding. In this contribution, we exploit the interview correlation with the aid of an interview motion search in distributed multi-view video coding (DMVC). Initially, we rely on the system architecture of WZ coding invoked for multiview video. Then, we construct a novel mesh-structured pixel-correlation model from the inter-view motion vectors and derive its decoding rules for joint source-channel decoding. Finally, we benchmark the attainable system performance against the existing pixel-domain WZ coding based DMVC scheme, where the classic turbo codec is employed. Our simulation results show that substantial bitrate reductions are achieved by employing the proposed motion-aware mesh-structured correlation modelling technique in a DMVC scheme.

Layered video coding is capable of progressively refining the reconstructed video quality with the aid of multiple layers of unequal importance. When the base layer (BL) is corrupted or lost due to channel impairments, the enhancement layers (ELs) must be discarded by the video decoder, regardless whether they are perfectly decoded or not, which implies that the transmission power assigned to the ELs is wasted. To circumvent this problem, we proposed a bit-level inter-layer forward error correction (IL-FEC) scheme for layered video transmission in our previous work, which implanted the systematic bits of the BL into the systematic bits of the ELs using exclusive-OR operations (XOR). This allowed the receiver to exploit the implanted bits of the ELs for assisting the BL's decoding and hence improved the overall system performance of our IL-FEC aided layered video scheme. In this treatise, we find the specific FEC coding rates in a real-time on-line fashion for the sake optimizing the overall system performance. The proposed procedure is widely applicable to diverse wireless transceivers and FEC codecs. Our simulation results show that the proposed optimized IL-FEC system outperforms the traditional optimal UEP by about 1.9 dB of Eb/N0at a peak signal-to-noise ratio (PSNR) of 38 dB. Viewing the improvements in terms of the video quality, 3.3 dB of PSNR improvement is attained at an Eb/N0of 10 dB, when employing a recursive systematic convolutional (RSC) code.

Driven by real-world applications such as fitness, wellbeing and healthcare, accelerometry-based activity recognition has been widely studied to provide context-awareness to future pervasive technologies. Accurate recognition and energy efficiency are key issues in enabling long-term and unobtrusive monitoring. While the majority of accelerometry-based activity recognition systems stream data to a central point for processing, some solutions process data locally on the sensor node to save energy. In this paper, we investigate the trade-offs between classification accuracy and energy efficiency by comparing on- and off-node schemes. An empirical energy model is presented and used to evaluate the energy efficiency of both systems, and a practical case study (monitoring the physical activities of office workers) is developed to evaluate the effect on classification accuracy. The results show a 40% energy saving can be obtained with a 13% reduction in classification accuracy, but this performance depends heavily on the wearer's activity.

An iterative Joint Source and Channel Coded Modulation (JSCCM) scheme is proposed for robust video transmission over two-way relaying channels. The system advocated was designed for improving the throughput, reliability and coverage area compared to that of conventional one-way relaying schemes. We consider a two-user communication system, where the users exchange their information with the aid of a twin-antenna Relay Node (RN). For each user the proposed lossless video scheme is comprised of a Variable Length Code (VLC) encoder and two Turbo Trellis Coded Modulation (TTCM) encoders one at the Source Node (SN) and one at the RN. The spatio-temporal redundancy of the video sequence is exploited for reducing the iterative decoding complexity. The decoding convergence behaviour of the decoder as well as the power sharing ratio between the two SNs and the RN are characterized with the aid of EXtrinsic Information Transfer (EXIT) charts. Our proposed scheme exhibits an SNR gain of 9 dB compared to the non-cooperative scheme, when communicating over Rayleigh fading channels.

Low Density Parity Check (LDPC) decoders have an inherent capability of correcting the transmission errors that occur, when communicating over a hostile wireless channel. This capability allows LDPC-coded schemes to employ lower transmission energies than uncoded schemes, at the cost of introducing a significant processing energy consumption during LDPC decoding. Traditional energy-reduction techniques, such as voltage and clock scaling can be employed for reducing the LDPC decoder's energy consumption. However, these techniques may induce timing errors, which can degrade the LDPC decoder's error correction capability. Our previous work has demonstrated that in contrast to other types of LDPC decoders, stochastic decoders have an inherent tolerance to timing errors, allowing them to maintain a high error correction capability in clockscaling scenarios. In this paper, we investigate this timing error tolerance in voltage-scaling scenarios, by extending our previous model of timing errors using extensive SPICE simulations. Furthermore, we use these SPICE simulations to characterize the processing energy consumption of stochastic LDPC decoders for the first time. We demonstrate that a modified stochastic LDPC decoder can operate at 0.8 V and a clock period of 915.11 ps, while maintaining the error correction capability of a conventional stochastic decoder operating at 1 V and a clock period of 1019.2 ps, offering a 36.7% reduction in processing energy consumption.

Unary Error Correction (UEC) codes have recently been proposed for the near-capacity Joint Source and Channel Coding (JSCC) of symbol values that are selected from a set having an infinite cardinality. In this paper, we characterize the computational complexity of UEC decoders and use complexity analysis for striking a desirable trade-off between the contradictory requirements of low complexity and near-capacity operation. We investigate a wide range of application scenarios and offer a deep insight into their beneficial parameterizations. In particular, we introduce puncturing for controlling the scheme's throughput and for facilitating fair comparisons with a Separate Source and Channel Coding (SSCC) benchmarker. The UEC scheme is found to offer almost 1.3 dB gain, when operating within 1.6 dB of the capacity bound. This is achieved without any increase in transmission energy, bandwidth, transmit duration or decoding complexity.

Iterative source-channel decoding (ISCD) exploits the residual redundancy of the source by iteratively exchanging extrinsic information with the channel codecs to achieve robust transmission. In this treatise we propose a video codec suitable for lossless video compression and IS CD. At the encoder the spatio-temporal redundancy is partially removed at a low complexity by simply evaluating the frame difference (FD), which is then variable-length encoded (VLC). At the receiver we invoke a three-stage concatenated IS CD scheme for exploiting the residual redundancy of the FD. A Markov Random Field (MRF) model based soft-in-soft-out (SISO) module is proposed for exploiting the spatial correlations amongst the adjacent video pixels, using intra-frame coding. More explicitly, the VLC SISO decoder operates by exchanging soft information with both the MRF module and the soft channel decoder. The convergence of the three-stage iterative decoding process is examined using 3D extrinsic information transfer (EXIT) charts. Finally, we show that our system exhibits a substantial Eb/NOimprovement of about 5.2 dB compared to the corresponding benchmarker schemes.

Since joint source‐channel decoding is capable of exploiting the residual redundancy in the source signals for improving the attainable error resilience, it has attracted substantial attention. In this treatise, the authors study iterative source‐channel decoding (ISCD) aided video communications, where the video signal redundancy is modelled by a first‐order Markov process. Firstly, the authors derive reduced‐complexity formulas for the first‐order Markov modelling (FOMM) aided source decoding. Then they propose a bit‐based iterative horizontal–vertical scanline model (IHVSM) aided source decoding algorithm, where a horizontal and a vertical source decoder are employed for exchanging their extrinsic information using the iterative decoding philosophy. The iterative IHVSM aided decoder is then employed in a forward error correction (FEC) encoded uncompressed video transmission scenario, where the IHVSM and the FEC decoder exchange softbit‐information for performing turbo‐like ISCD for the sake of improving the reconstructed video quality. Finally, the authors benchmark the attainable system performance against a near‐lossless H.264/AVC video communication system and the existing FOMM‐based softbit source decoding scheme. The authors simulation results show that E b / N 0 improvements in excess of 2.8 dB are attainable by the proposed technique in uncompressed video applications.

We design distributed multiple-component turbo-codes (MCTCs) for cooperative automatic repeat request (CARQ) aided systems, which are capable of attaining both a similar performance and throughput as the conventional twin-component turbo code (TCTC) aided system, while imposing a significantly lower decoding complexity. The reduction in the decoding complexity of the MCTC-CARQ scheme is a direct benefit of using low-memory MCTC component encoders, combined with the novel principle of deferred iteration (DI), where the decoding process is only initiated when sufficient information has been accumulated at the receiver so that an open tunnel exists in the Extrinsic information transfer (EXIT) charts. In the presence of multiple collaborating relay nodes (RNs), the specific RN which succeeds in decoding the source message, and additionally, has the highest signal-to-noise ratio at the destination node is selected as the retransmission node. Our numerical results demonstrate that the proposed MCTC-CARQ protocol is capable of reducing the decoding complexity by up to 40% compared to its TCTC-CARQ counterpart, while achieving the same throughput and reliability.

A novel Joint Source and Channel Code (JSCC) is proposed, which we refer to as the Unary Error Correction (UEC) code. Unlike existing JSCCs, our UEC facilitates the practical encoding of symbol values that are selected from a set having an infinite cardinality. Conventionally, these symbols are conveyed using Separate Source and Channel Codes (SSCCs), but we demonstrate that the residual redundancy that is retained following source coding results in capacity loss. This loss is found to have a value of 1.11 dB in a particular practical scenario, where Quaternary Phase Shift Keying (QPSK) modulation is employed for transmission over an uncorrelated narrowband Rayleigh fading channel. By contrast, the proposed UEC code can eliminate this capacity loss, or reduce it to an infinitesimally small value. Furthermore, the UEC code has only a moderate complexity, facilitating its employment in practical low-complexity applications.

It was conventionally regarded that the approximate Bayesian theorem based existing probabilistic data association (PDA) algorithms output the estimated symbol-wise a posteriori probabilities (APPs) as soft information. In our recent work, however, we demonstrated that these probabilities are not the true APPs in the rigorous mathematical sense, but a type of nominal APPs, which are unsuitable for the classic architecture of iterative detection and decoding (IDD) aided receivers. To circumvent this predicament, in this paper we propose an exact Bayesian theorem based logarithmic domain PDA (EB-Log-PDA) method, whose output has similar characteristics to the true APPs, and hence it is readily applicable to the classic IDD architecture of multiple-input-multiple-output (MIMO) systems using the general M-ary modulation. Furthermore, we investigate the impact of the EB-Log-PDA algorithm's inner iteration on the design of EB-Log-PDA aided IDD receiver. We demonstrate that introducing inner iterations into EB-Log-PDA, which is common practice in conventional-PDA aided uncoded MIMO systems, would actually degrade the IDD receiver's performance, despite significantly increasing the overall computational complexity of the IDD receiver. Finally, we investigate the relationship between the extrinsic log-likelihood ratios (LLRs) of the proposed EB-Log-PDA and of the approximate Bayesian theorem based logarithmic domain PDA (AB-Log-PDA) reported in our previous work. Despite their difference in extrinsic LLRs, we also show that the IDD schemes employing the EB-Log-PDA and the AB-Log-PDA without incorporating any inner PDA iterations have a similar achievable performance close to that of the optimal maximum a posteriori (MAP) detector based IDD receiver, while imposing a significantly lower computational complexity in the scenarios considered.

Turbo codes have recently been considered for energy-constrained wireless communication applications, since they facilitate a low transmission energy consumption. However, in order to reduce the overall energy consumption, lookup table-log-BCJR (LUT-Log-BCJR) architectures having a low processing energy consumption are required. In this paper, we decompose the LUT-Log-BCJR architecture into its most fundamental add compare select (ACS) operations and perform them using a novel low-complexity ACS unit. We demonstrate that our architecture employs an order of magnitude fewer gates than the most recent LUT-Log-BCJR architectures, facilitating a 71% energy consumption reduction. Compared to state-of-the-art maximum logarithmic Bahl-Cocke-Jelinek-Raviv implementations, our approach facilitates a 10% reduction in the overall energy consumption at ranges above 58 m.

Layered video coding creates multiple layers of unequal importance, where the enhancement layers will be affected when the base layer is corrupted. In this treatise, a novel inter-layer FEC scheme is investigated, where the information of the base layer1is incorporated into the systematic bits of the enhancement layers with the aid of an exclusive-OR operation. When the base layer can be recovered independently, the soft information of the enhancement layers can be deduced by flipping the sign of the check information received. In this case, the protection strength of the enhancement layer is unaffected and no extra protection bits are required. Otherwise, the inter-layer FEC decoding philosophy related to the base layer and the enhancement layer will be activated to assist in decoding the base layer, where the protection of the enhancement layer is additionally exploited to protect the base layer. Data partitioning based experiments show that our proposed scheme outperforms the traditional unequal error protection FEC aided transmission system by about 1.8 dB of channel SNR or 7.7 dB of PSNR at an acceptable complexity.

Since the introduction of turbo code aided Hybrid Automatic Repeat reQuest (HARQ) schemes, their complexity reduction has drawn research attention. In our previous work, we proposed an Early Stopping (ES) strategy for a turbo HARQ scheme, which results in a beneficial complexity reduction, while maintaining a high throughput. However, this scheme was designed to strike a compromise across the full range of channel conditions. As a further advance, in this paper, we propose a new Deferred Iterations (DI) strategy, which is specifically designed for taking into account the prevalent channel conditions, as characterized by the Extrinsic Information Transfer (EXIT) chart tunnel opening. More specifically, the DI strategy delays the commencement of turbo decoding until an open EXIT chart tunnel appears. Our simulation results demonstrate that the complexity of the proposed DI aided turbo HARQ schemes is reduced by up to 50%, which is achieved without compromising the Packet Loss Ratio (PLR) or throughput.

Soft Decode and Forward (SDF) cooperation techniques have been shown to improve the performance of Distributed Turbo Coding (DTC) schemes, in which the Source Node (SN) and Relay Node (RN) employ convolutional codes, while the Destination Node (DN) employs a turbo decoder. However, this technique has not be able to mitigate the typical error floors that are associated with DTC. In this paper, we mitigate these error floors by upgrading the SDF technique to a scheme in which turbo codes are employed at the SN, RN and DN. We demonstrate that this scheme is capable of offering significant gains over both Amplify and Forward (AF) and classic direct-transmission schemes, even if only partial decoding is performed in the RN. More specifically, in our Partial Soft Decode and Forward (P-SDF) scheme, there is no excessive penalty associated with terminating the RN's iterative decoding process before convergence is achieved, as a benefit of the soft nature of its modulation scheme.

The computational complexity and hence energy consumption of a low-density parity-check (LDPC) decoder employing fixed point (FP) computation is commensurate with the operand width (OW) of the FP representation employed. Therefore low OWs are desirable, but, if the OW is too low, then an inevitable performance degradation will be introduced. For that reason it is desirable to determine the minimum OW that does not impose a significant performance degradation. Previous efforts have advocated different OWs based on results obtained using time-consuming bit-error ratio (BER) simulations. However, these BER simulations are extremely time consuming, owing to the requirement of considering a range of channel signal-to-noise ratios (SNRs). Therefore, in this paper, we propose the employment of extrinsic information transfer (EXIT) charts to overcome this drawback. Furthermore, EXIT-chart analysis has the additional benefit of offering insights into the specific causes of the performance degradations encountered. Finally, a FP scheme having an overall OW of 6 bits is proposed for the implementation of the min-sum algorithm (MSA).

Hybrid-Automatic-Repeat-reQuest (HARQ) has become an indispensable technique in reliable communications systems. However, its performance is inevitably affected by the channel's fading correlation. In this paper, we proposed a novel relay-switching aided HARQ scheme in order to mitigate the detrimental effects of correlated fading without unduly increasing the system's complexity and delay. Our results show that the proposed relay-switching regime operates efficiently in correlated channels, hence significantly reduces the error floor of turbo-coded HARQ. Additionally, a HARQ scheme using Segment Selective Repeat (SSR) is incorporated in the relay-switching scheme for achieving further improvements. Quantitatively, the proposed relay-switching aided turbo-coded HARQ scheme using SSR may achieve an approximately 2 dB gain, compared to the conventional amplify-and-forward aided turbo coded HARQ arrangement using Chase Combining.

Previous interleavers designed for Serially Concatenated Codes (SCCs) have achieved a limited Minimum Hamming Distance (MHD) between the legitimate permutations of the encoded bit sequence. Hence, we propose a novel Evolutionary Algorithm (EA) capable of designing improved interleavers for SCCs, without artificially limiting the achievable MHD. As a result, increased MHDs and therefore reduced error floors are achieved, even if only a modest EA complexity can be afforded.

Shannon's channel capacity specifies the upper bound on the amount of bits per channel use. In this paper, we explicitly demonstrate that twin-component turbo codes suffer from a capacity loss, when the component code rate is less than unity, which is shown by exploiting the so-called area properties of Extrinsic Information Transfer (EXIT) charts. This capacity loss is unavoidable for twin-component turbo codes, when the overall turbo coding rate is less than 1/2, while multiple-component turbo codes are capable of overcoming it by using unity-rate component codes. In order to demonstrate that multiple-component turbo codes are capable of exhibiting a better asymptotic performance, the minimum Signal Noise Ratio (SNR) required for the EXIT charts to have open convergence tunnels is used as our metric, which is referred to as 'the open tunnel SNR threshold'. Furthermore, the employment of conventional two-dimensional EXIT charts is extended to facilitate the analysis of N-component turbo codes. Our results confirm that multiple-component turbo codes approach the Discrete-input Continuous-output Memoryless Channel's (DCMC) capacity more closely and achieve a lower Bit Error Ratio (BER) than twin-component turbo codes at the same coding rate and the same complexity.

Hybrid Automatic Repeat reQuest (ARQ) plays an essential role in error control. Combining the incorrectly received packet replicas in hybrid ARQ has been shown to reduce the resultant error probability, while improving the achievable throughput. Hence, in this contribution, multi-level turbo codes have been amalgamated both with hybrid ARQ and efficient soft combining techniques for taking into account the Log-Likelihood Ratios (LLRs) of retransmitted packet replicas. In this paper, we present a soft combining aided hybrid ARQ scheme based on multi-level turbo codes, which avoid the capacity loss of the twin-level turbo codes that are typically employed in hybrid ARQ schemes. More specifically, the proposed receiver dynamically appends an additional parallel concatenated Bahl, Cocke, Jelinek and Raviv (BCJR) algorithm based decoder in order to fully exploit each retransmission, thereby forming a multi-level turbo decoder. Therefore, all the extrinsic information acquired during the previous BCJR operations will be used as a priori information by the additional BCJR decoders, whilst their soft output iteratively enhances the a posteriori information generated by the previous decoding stages. We also present link level Packet Loss Ratio (PLR) and throughput results, which demonstrate that our scheme outperforms some of the previously proposed benchmarks.

The operand-width specifications in fixed-point hardware implementations of turbo code decoders is an important design issue, since this governs the trade-off between the decoder's performance and its complexity, cost, area and energy consumption. The investigation of this issue would be extremely time-consuming in the conventional approach, which relies upon Monte-Carlo simulation based Bit Error Ratio (BER) analysis. In this paper, we propose a generic design method, which uses EXtrinsic Information Transfer (EXIT) chart analysis to simplify this design process. Our method is not only an order of magnitude faster than the conventional Monte-Carlo simulation based approach, but also offers deeper insights into why performance degradations are imposed by insufficient operand-width specifications. The benefits of our generic method are demonstrated in the context of a turbo decoder, allowing accurate specifications to be obtained and compared to those suggested by previous works.

Systematic Luby Transform (SLT) codes constitute rateless codes, which are capable of adaptively adjusting their code rate depending on the channel quality without any explicit channel state information (CSI) at the transmitters. In this paper, we investigate the performance of an SLT coded wireless system communicating over a correlated fading channel within an 802.11b network. The simulation results show that the SLT coded system can achieve a 3 dB gain in term of fading deviation when fading correlation changes from 1 Hz to 5 Hz. Moreover, it is found that a system employing an SLT code can provide similar performance to other rateless codes, namely the Luby Transform (LT) code and the Random Linear Fountain Code (RLFC). However, this level of performance is achieved with a reduced number of decoding operations. A 20% reduction is achieved when the standard deviation of the fading amplitude is 10 dB and the communication distance is 7 m. Greater reductions are observed under more benign propagation conditions.

The tolerable packet-loss ratio of an Internet Protocol (IP) based wireless networks varies according to the specific services considered. File transfer for example must be error free but tolerates higher delays, whereas maintaining a low delay is typically more important in interactive Voice Over IP (VOIP) or video services. Classic Forward Error Correction (FEC) may be applied to the data to provide resilience against bit errors. A wireless IP network provides the opportunity for the inclusion of FEC at the physical, transport and application layers. The demarcation between the analogue and digital domain imposed at the Physical layer (PHY) predetermines the nature of the FEC scheme implemented at the various layers. At the PHY individual packets may be offered FEC protection, which increases the likelihood of their error-free insertion into the protocol stack. Higher layers receive packets that are error free and the purpose of a FEC scheme implemented here is to regenerate any missing packets obliterated for example by the Binary Erasure Channel (BEC) of the IP network's routers. A rateless code may be beneficially employed at a higher Open Systems Interconnection (OSI) layer for replenishing the obliterated packets, but unless the characteristics of the channel are considered, the ultimate rate achieved by such a code may be compromised, as shown in this contribution.

In this treatise we propose a three-stage twin-transmit-antenna assisted Multiuser (MU) Direct Sequence Code-Division Multiple Access (DS-CDMA) system employing both a Unitary Rate Code (URC) and a Recursive Systematic Code (RSC) to carry out iterative turbo detection. A Space-Time Block Code (STBC) is used to provide second-order diversity gain in conjunction with a novel soft-output (SO) ant-colony-optimization (ACO) based space-time multiuser detection (ST/MUD) algorithm, which is capable of carrying out STBC decoding, while mitigating the Multiuser Interference (MUI) and generating the Log-Likelihood Ratios (LLR) for facilitating the iterative exchange of extrinsic information between the URC and RSC decoders of each user. With aid of the above mentioned state-of-the-art techniques, the proposed system becomes capable of approaching the performance of the single-user system within about 0.5dB, when K = 32 users are supported with the aid of 31-chip Gold-codes.

In this paper, we develop a capacity-approaching Cooperative Space-Time Coding CSTC scheme employing irregular design for a twin-relay aided network as an extension of our previous work cast in the context of a half-duplex single-relay-aided network. For the sake of recovering the multiplexing loss imposed by a half-duplex three-terminal network, we employ a successive relaying protocol in this paper, where an additional relay node is activated. Hence, in order to design a near-capacity coding system, first the capacity and the achievable information-rate of a specific space-time coding aided scheme are quantified for the successive relaying aided channel. More specifically, the cooperative space-time codes employed at the source and the relays are jointly designed with the aid of EXtrinsic Information Transfer EXIT charts for the sake of high-integrity operation at Signal-to-Noise Ratios SNRs close to the corresponding successive relaying channel's capacity. Furthermore, unlike in the half-duplex single-relay based system, the destination node performs frame-by-frame Successive Interference Cancellation SIC aided iterative detection, in order to mitigate the efforts of multiple-access interference. Finally, our numerical results demonstrate that our proposed Irregular Cooperative Space-Time Coding Ir-CSTC scheme is capable of near-capacity operation in the successive relaying aided network, which is an explicit benefit of our joint source-and-relay transceiver design.

This paper presents a low complexity iteratively detected space-time transmission architecture based on Generalized Multi-Layer Space-Time (GMLST) codes and Irregular Convolutional Codes (IRCCs). The GMLST combines the benefits of the Vertical Bell-Labs LAyered Space-Time (VBLAST) scheme and Space-Time Coding (STC). The GMLST is serially concatenated with a Unity-Rate Code (URC) and an IRCC which are used to facilitate near-capacity operation with the aid of an EXtrinsic Information Transfer (EXIT) chart based design. Reduced-complexity iterative multistage Successive Interference Cancellation (SIC) is employed in the GMLST decoder, instead of the significantly more complex Maximum Likelihood (ML) detection. For the sake of approaching the maximum attainable rate, iterative decoding is invoked to achieve decoding convergence by exchanging extrinsic information across the three serial component decoders. Finally, it is shown that the SIC-based iteratively detected IRCC-URC-GMLST system is capable of providing a feasible trade-off between the affordable computational complexity and the achievable system throughput.

In this paper, an Irregular Distributed Space-Time (Ir-DST) coding scheme is studied in the context of a twin-relay aided network in which the successive relaying protocol is employed. A tight upperbound of the successive relaying aided network's capacity is given. The distributed codes at the source and relays are jointly designed with the aid of Extrinsic Information Transfer (EXIT) charts for the sake of high-integrity operation at Signal-to-Noise Ratios (SNRs) close to the corresponding network's capacity. Finally, it is shown that our proposed Ir-DST coding scheme is capable of near-capacity cooperative communications in the successive relaying aided network, which is an explicit benefit of our joint source-and-relay mode design.

We present a novel tri-functional multiple-input multiple-output (MIMO) scheme that intrinsically amalgamates space-time spreading (STS), the Vertical Bell Labs Layered Space-Time (V-BLAST) scheme and beamforming with generalised multicarrier direct sequence code division multiple access (MC DS-CDMA). Further system performance improvements can be attained by employing channel coding, where the source bits are serial-to-parallel converted to two layers, each constituted by a serial concatenation of an outer code amalgamated with a unity-rate code for the sake of improving the convergence behaviour of the proposed system. Additionally, the convergence behaviour of the iteratively detected scheme is evaluated with the aid of extrinsic information transfer (EXIT) charts. We also propose a novel logarithmic likelihood ratio (LLR) postprocessing technique for improving the iteratively detected system's performance. Explicitly, after I = 10 decoding iterations and employing an interleaver depth of Dint= 160,000 bits, the proposed system supporting K = 4 users attains a BER below 10-5for Eb/Novalues in excess of -2.8 dB.

The powerful tool of EXtrinsic Information Transfer (EXIT) charts is used to design a new serially concatenated Irregular Variable Length Coded (IrVLC) and unity-rate precoded Time-Hopping (TH) Pulse Position Modulation (PPM) aided Ultra-Wide Bandwidth (UWB) Spread-Spectrum (SS) impulse radio for near-capacity operation in Nakagami-m fading channels contaminated by Partial Band Noise Jamming (PBNJ). The benefits of the 3-stage concatenation of the TH-UWB detector, the unity-rate decoder and the outer IrVLC decoder are quantified. A number of novel Variable Length Coding (VLC) codebooks having different coding rates are utilized by the IrVLC scheme for encoding specific fractions of the input source symbol stream. More explicitly, EXIT charts are employed to appropriately select these input stream fractions to shape the inverted EXIT curve of the IrVLC in order to match that of the inner decoder and hence to achieve an infinitesimally low Bit Error Ratio (BER) at near-capacity SNR values.

In this paper we devise an analytical framework for the characterization of wireless transceivers. Subsequently, we utilize the proposed methodology in order to calculate the constrained capacity of the delay-limited wireless transceivers. We demonstrate that the realistic constrained capacity may be significantly lower than the idealized unconstrained Shannon capacity. Furthermore, the constrained capacity of a delay-limited transceiver may vary significantly depending on the particular combination of application scenario and channel conditions.

Fountain codes constitute novel erasure codes, which have been standardized for Forward Error Correction (FEC) in broadcast network protocols and by the Third-Generation Partnersonhip Project (3 GPP). The basic operational units of Fountain codes are source packets, which have a particular fixed length. These codes are invoked here in an 802.11 Wireless Local Area Network (WLAN) scenario for protecting file transfers. More specifically, the optimal packet length is selected by considering the 802.11 Media Access Control (MAC) retransmission rate and the properties of the physical layer's modulation scheme. Naturally, owing to the limited memory of the encoders/decoders, large source files must be decomposed into shorter transport blocks. Therefore, methods for partitioning the file and acknowledging the successful transmission of each block are also proposed here. Compared to the file transfer regime operating without FEC over the classic TCP protocol, the proposed regime requires a lower threshold SNR for accomplishing a successful file transfer and hence enhances the transmission efficiency by about 50%.

This paper presents an Irregular Distributed SpaceTime (Ir-DST) coding scheme designed for near-capacity cooperative communications. A serial concatenated scheme comprising an IRregular Convolutional Code (IRCC), a recursive Unity-Rate Code (URC) and a Space-Time Block Code (STBC) was designed for the sake of approaching the corresponding source-to-relay link capacity, where the IRCC was optimized with the aid of Extrinsic Information Transfer (EXIT) charts which was used at the source node. At the relay node, another IRCC is concatenated serially with an identical STBC. The relay's IRCC is re-optimized based on EXIT chart analysis for the sake of approaching the relay network's capacity, before transmitting the relayed information. We will demonstrate that the topology of the Ir-DST system coincides with that of a Distributed Turbo Code (DTC). At the destination node, a novel three-stage iterative decoding scheme is constructed in order to achieve decoding convergence to an infinitesimally low Bit Error Ratio (BER). Finally, it is shown that our joint source-and-relay mode design based on EXIT chart analysis is capable of near-capacity cooperative communications.

In this contribution we propose the novel serial concatenation of Irregular Variable Length Coding (IrVLC) and Irregular Unity Rate Coding (IrURC), where we matched the corresponding Extrinsic Information Transfer (EXIT) functions to each other. This approach facilitates a higher degree of design freedom than matching the EXIT function of an irregular codec to that of a regular codec. As a result, a narrow EXIT chart tunnel can be created, facilitating operation at Eb/N0values that are closer to the channel's capacity bound. The computational complexity and Bit Error Ratio (BER) performance of our IrVLC-IrURC scheme is favourable in comparison to the bench markers that replace either one or both of the irregular codecs by the equivalent-rate regular codec.

In this paper we evaluate the performance of a data-partitioned (DP) H.264 coded video transmission system using unequal error protection (UEP) irregular convolutional codes (IRCC). Using UEP, perceptually more important bits are provided with more strong protection relative to less important bits. An iterative detection aided combination of IRCC and a rate-1 precoder was used to improve the overall BER performance and to enhance the objective video quality expressed in terms of peak signal-to-noise ratio (PSNR). The effect of different error protection schemes on the attainable system performance is demonstrated, while keeping the overall bit-rate budget constant for the transmission of DP H.264 source coded video over correlated narrowband Rayleigh fading channels. In this paper we exploited the high design flexibility of IRCCs, which constitutes a family of different rate subcodes, while maintaining an excellent iterative decoding convergence performance. Additionally, due to the use of different-rate subcodes, IRCCs have the capability of providing UEP for the H.264 coded video stream. An EXIT chart matching procedure was used for the design of our specific IRCC. Additionally, EXIT charts were used for analysing the attainable system performance of various error protection schemes employed. Explicitly, our experimental results show that the proposed UEP scheme using IRCC outperforms its equal error protection (EEP) counterpart employing regular convolutional codes by about 0.5 dB Eb/Noat the PSNR degradation point of 1 dB.

Irregular Variable Length Coding Design for serial concatenated and iteratively decoded Time-Hopping (TH) Pulse Position Modulation (PPM) UltraWideBand (UWB) Spread-Spectrum (SS) Impulse radio system is considered. The proposed design is capable of low Signal-to-Noise Ratio (SNR) operation in Nakagami-m fading channel amalgamated with joint source and channel coding schemes. A number of component Variable Length Coding (VLC) codebooks with different coding rates are being utilized by IVLC scheme for encoding specific fractions of the input source symbol stream. The EXtrinsic Information Transfer (EXIT) charts are used to select these fractions in order to shape inverted EXIT curve of IVLC according to EXIT curve of the inner decoder match. The proposed scheme can achieve near-zero bit error ratio at low SNR values. This IVLC based scheme provides a gain of up to 0.45 dB over the identical-rate single class VLC based scheme.

In this contribution, a novel soft-output ant colony optimization (SO-ACO)-based multiuser detector (MUD)—namely the MUlti-input-Approximation (MUA) assisted SO-ACO-based MUD—is proposed for the synchronous direct-sequence code-division-multiple-access (DS-CDMA) uplink (UL). The previously proposed conventional ACO based MUDs were unable to provide soft log-likelihood ratio (LLR) values for the channel decoder. Hence, to solve this open problem, we commence by proposing the MAximum-Approximation (MAA) assisted SO-ACO algorithm, leading to a novel MUA assisted SO-ACO algorithm, which subsumes the MAA algorithm as a particular case and outperforms the MAA algorithm. More explicitly, at a signal-to-noise ratio (SNR) of 13 dB, the BER performance of the convolutional coding (CC) aided CDMA UL employing the MAA SO-ACO is improved from$5.2\cdot 10^{-6}$to$2.7\cdot 10^{-6}$by employing the MUA SO-ACO. Our numerical results also demonstrate that the MUA assisted SO ACO-MUD is capable of approaching the optimum performance of the Bayesian detector, when$K=32$UL users are supported with the aid of 31-chip Gold codes, while the complexity of the former is a fraction of$10^{-8}$lower than that of the latter.

In this paper we propose a novel real-valued free distance metric (RV-FDM) for comparing the error correction capabilities of variable length error correction (VLEC) codebooks that have the same integer-valued free distance lower bounds. We demonstrate that VLEC codebooks having higher RV-FDMs tend to have extrinsic information transfer (EXIT) functions with more pronounced 'S'-shapes. Furthermore, we show that higher-accuracy EXIT chart matching can be achieved if the component EXIT functions of an irregular code exhibit more variety. This motivates the employment of our novel genetic algorithm for designing the component VLEC codes of irregular variable length coding, that have particular EXIT functions, in addition to exhibiting desirable bit entropies and decoding complexities.

In this contribution we introduce an extrinsic information transfer (EXIT) chart matching technique for the design of two serially concatenated irregular codecs, each constituted by a variety of component codes. This approach facilitates a higher degree of design freedom than matching the EXIT function of an irregular codec to that of a regular codec, comprising only a single component code. As a result, a narrower EXIT chart tunnel can be created, facilitating operation at Eb/N0values that are closer to the channel's capacity bound. This is demonstrated for a serial concatenation of iteratively decoded irregular variable length coding (IrVLC) and irregular unity rate coding (IrURC), which is favourably compared with an IrVLC and regular unity rate coding (URC) based benchmarker. Finally, we show that the iterative decoding complexity of our IrVLCIrURC scheme can be reduced by about 25% upon employing a method of jointly performing EXIT chart matching, while seeking a reduced iterative decoding complexity.

A near-capacity irregular bit-interleaved coded modulation based iterative decoding (Ir-BICM-ID) aided scheme is proposed. The irregular design of the scheme pervades the three basic components of BICM-ID, namely the encoder, the unity-rate precoder and the bit-to-symbol mapper. As a result, irregular BICM-ID schemes constituted by irregular components are created, which are capable of approaching the capacity of coded modulation. This is achieved by creating a marginally open extrinsic information transfer (EXIT) chart tunnel, and exploiting the theorem that the open tunnel's area is characteristic of how closely the scheme operates to the channel's capacity. The proposed Ir-BICM-ID scheme employs irregular convolutional codes (IrCC), irregular unity-rate codes (IrURC) and irregular mappers (IrMapper).

In this contribution, a novel soft-output Ant Colony Optimization (ACO) based Multi-User Detector (MUD) is proposed for the synchronous Direct-Sequence Code-Division- Multiple-Access (DS-CDMA) uplink. The foraging behaviour of the ant colony in nature motivates the employment of reduced- search ACO-based MUDs, which are capable of approaching the optimum Maximum Likelihood (ML) MUD's performance at the cost of a computational complexity, which maybe as low as that of the Mathched Filter (MF) based Single User Detector (SUD). However, the previously proposed conventional ACO based MUDs were unable to provide soft Log-Likelihood Ratio (LLR) values for the channel decoder. Hence in this paper, we present a novel soft-output ACO-MUD capable of delivering soft LLRs, which allows a CDMA system to achieve a near-single-user performance without any additional information feedback from the channel decoder, even when the number of users supported is as high as the number of chip in the spreading sequence. Our numerical results show that at a BER of 10-3, the performance of the currently known ACO-assisted state-of-the-art systems can be improved by about 17 dB with the aid of the proposed soft-output ACO-MUD. More explicitly, the soft-output ACO- MUD is capable of approaching the optimum performance of the Bayesian detector, when K = 32 UL users are supported with the aid of 31-chip Gold codes, while the complexity of the former is only a fraction of 10-8of the latter.

In this paper, we propose a method for desirably redistributing a wireless sensor network's energy consumption from its sensor nodes (which may have scarce energy resources obtained through energy harvesting, for example) to its central node (which often has an abundant energy resource, such as the mains). At the cost of increasing the central node's decoding complexity, our method facilitates (1) a significant reduction in the number of times the sensor nodes are required to retransmit data owing to transmission errors and/or (2) a reduction of up to 3.99 dB in the sensor node's total transmit energy consumption. We show that our approach can reduce the overall energy consumption of transmitting sensor nodes by more than 20% in practice.

An irregular bit-interleaved coded modulation based iterative decoding (Ir-BICM-ID) aided scheme is proposed. The irregularity of the scheme pervades the three basic components of BICM-ID, namely the encoder, the unity-rate precoder and the bit-to-symbol mapper. As a result, adaptive BICM-ID schemes constituted by irregular components are created, which are capable of approaching the capacity of coded modulation. This is achieved by creating a narrow extrinsic information transfer (EXIT) chart, using a novel EXIT curve matching algorithm. The proposed Ir-BICM-ID scheme employs irregular convolutional codes (IrCC), irregular unity-rate codes (IrURC) and irregular mappers (IrMapper).

Serially concatenated and iteratively decoded Irregular Variable Length Coding (IrVLC) combined with preceded Fast Frequency Hopping (FFH) M-ary Frequency Shift Keying (MFSK) is considered. We employ Extrinsic Information Transfer (EXIT) charts to investigate the 3-stage concatenation of the FFH-MFSK demodulator, the rate-1 decoder and the outer IrVLC decoder. The proposed joint source and channel coding scheme is capable of operating at low Signal-to-Noise Ratio (SNR) in Rayleigh fading channels contaminated by Partial Band Noise Jamming (PBNJ). The IrVLC scheme is comprised of a number of component Variable Length Coding (VLC) codebooks employing different coding rates for encoding particular fractions of the input source symbol stream. These fractions may be chosen with the aid of EXIT charts in order to shape the inverted EXIT curve of the IrVLC codec so that it can be matched with the EXIT curve of the inner decoder. We demonstrate that using the proposed scheme an inflnitesimally low bit error ratio may be achieved at low SNR values.

In this paper we demonstrate that our ability to match the Extrinsic Information Transfer (EXIT) function of an Irregular Variable Length Code (IrVLC) to that of a serially-concatenated inner code depends on the availability of a suite of component Variable Length Error Correction (VLEC) codebooks having a wide variety of inverted EXIT function shapes. We also show that the inverted EXIT function shape of a VLEC codebook depends on its coding rate and Error Correction Capability (ECC). This motivates the design of a Genetic Algorithm (GA) that searches the large VLEC parameter space to find codebooks having specific coding rates, ECCs and, hence, EXIT function shapes. The employment of this GA therefore facilitates the design of component VLEC codebook suites without the manual trial-and-error that is required when employing the state-of- the-art Heuristic Algorithm (HA) used as our bench marker, which cannot design component codebooks having specific EXIT function shapes.

In this paper we propose a novel irregular variable length coding (IrVLC) scheme for near-capacity joint source and channel coding. We employ a number of component variable length coding (VLC) codebooks having different coding rates for encoding particular fractions of the input source symbol stream. These fractions may be chosen with the aid of extrinsic information transfer (EXIT) charts in order to shape the inverted EXIT curve of the IrVLC codec for ensuring that it does not cross the EXIT curve of a serially concatenated channel codec. In this way, an open EXIT chart tunnel may be created even at low Eb/N0values that are close to the capacity bound of the channel. We propose iteratively decoded serially concatenated IrVLC designs amalgamated with Trellis coded modulation (TCM). These schemes are shown to be capable of operating within 0.6 dB of the uncorrelated narrowband Rayleigh fading channel's capacity bound using an average interleaver length of 217,500 bits and an effective bandwidth efficiency of 1.56 bit/s/Hz, assuming ideal Nyquist filtering. By contrast, the equivalent-rate regular VLC-based bench-marker schemes were found to be capable of operating at a higher distance of 1.1 dB from the capacity bound, which is nearly twice that of the proposed IrVLC-TCM scheme. Additionally, an irregular convolutional coiling (IrCC) based bench-marker was found to be capable of operating at 0.8 dB from the capacity bound, owing to its slightly eroded performance when operating with the considered interleaver length.

Serially concatenated and iteratively decoded irregular variable length coding (IrVLC) combined with preceded fast frequency hopping (FFH) M-ary frequency shift keying (MFSK) is considered. The proposed joint source and channel coding scheme is capable of low signal-to-noise ratio (SNR) operation in Rayleigh fading channels contaminated by partial band noise jamming (PBNJ). The IrVLC scheme is comprised of a number of component variable length coding (VLC) codebooks employing different coding rates for encoding particular fractions of the input source symbol stream. These fractions may be chosen with the aid of extrinsic information transfer (EXIT) charts in order to shape the inverted EXIT curve of the IrVLC codec so that it can be matched with the EXIT curve of the inner decoder. We demonstrate that using the proposed scheme near-zero bit error ratio may be achieved at low SNR values and the IrVLC based scheme yields a further gain of up to 0.3 dB over the identical-rate single-class VLC based benchmarker scheme

In this paper we consider serially concatenated and iteratively decoded irregular variable length coding (IrVLC) combined with preceded differential space-time spreading (DSTS) aided multidimensional sphere packing (SP) modulation designed for near-capacity joint source and channel coding. The IrVLC scheme comprises a number of component variable length coding (VLC) codebooks having different coding rates for the sake of encoding particular fractions of the input source symbol stream. The relative length of these source-stream fractions can be chosen with the aid of extrinsic information transfer (EXIT) charts in order to shape the EXIT curve of the IrVLC codec, so that an open EXIT chart tunnel may be created even at low Eb/N0values that are close to the capacity bound of the channel. These schemes are shown to be capable of operating within 0.9 dB of the DSTS-SP channel's capacity bound using an average interleaver length of 113,100 bits and an effective bandwidth efficiency of 1 bit/s/Hz, assuming ideal Nyquist filtering. By contrast, the equivalent-rate regular VLC-based bench- marker scheme was found to be capable of operating at 1.4 dB from the capacity bound, which is about 1.56 times the corresponding discrepancy of the proposed IrVLC-aided scheme.

A joint video and channel coded system employing an iteratively decoded serial concatenation of a vector quantization (VQ) based video codec and a trellis-coded modulation (TCM) scheme is proposed. The video codec imposes VQ-induced code constraints, which may be completely described by a trellis structure, which is employed as the basis for optimal minimum mean-squared-error VQ-encoding and -decoding. In the latter case, the Bahl-Cocke-Jelinek-Raviv (BCJR) algorithm is employed to facilitate the iterative exchange of soft information between the VQ and TCM decoder. An error-free video reconstruction quality is supported using 16-Ievel quadrature amplitude modulation (16QAM) based TCM for transmission over Rayleigh-fading channels at a signal-to-noise ratio (SNR) per bit of 5.25 dB. This value is within 1.29 dB of the Rayleigh channel's capacity at our system's effective bandwidth-efficiency of 2 bits/s/Hz. Owing to its ability to exploit the VQ-induced code constraints during iterative decoding, the joint video and channel coding approach is found to consistently outperform the Shannonian source and channel separation philosophy. This is achieved at the cost of a 1.6 times higher computational complexity. Finally, the convergence of the iterative decoder is investigated with the aid of a novel so-called extrinsic information transfer (EXIT) chart

In this contribution an iterative (turbo) channel equalization and source decoding scheme is considered. In our investigations the source is modelled as a Gaussian-Markov source, which is compressed with the aid of vector quantization. The communications channel is modelled as a time-invariant channel contaminated by intersymbol interference (ISI). Since the ISI channel can be viewed as a rate-1 encoder and since the redundancy of the source cannot be perfectly removed by source encoding, a joint channel equalization and source decoding scheme may be employed for enhancing the achievable performance. In our study the channel equalization and the source decoding are operated iteratively on a bit-by-bit basis under the maximum aposteriori (MAP) criterion. The channel equalizer accepts the apriori information provided by the source decoding and also extracts extrinsic information, which in turn acts as a priori information for improving the source decoding performance. Simulation results are presented for characterizing the achievable performance of the iterative channel equalization and source decoding scheme. Our results show that iterative channel equalization and source decoding is capable of achieving an improved performance by efficiently exploiting the residual redundancy of the vector quantization assisted source coding.