Channel Coding Rate Research Articles

Performance Analysis of Wireless Information Surveillance in Machine-Type Communication at Finite Blocklength Regime.

The Internet of Things (IoT) will feature pervasive sensing and control capabilities via the massive deployment of machine-type communication devices in order to greatly improve daily life. However, machine-type communications can be illegally used (e.g., by criminals or terrorists) which is difficult to monitor, and thus presents new security challenges. The information exchanged in machine-type communications is usually transmitted in short packets. Thus, this paper investigates a legitimate surveillance system via proactive eavesdropping at finite blocklength regime. Under the finite blocklength regime, we analyze the channel coding rate of the eavesdropping link and the suspicious link. We find that the legitimate monitor can still eavesdrop the information sent by the suspicious transmitter as the blocklength decreases, even when the eavesdropping is failed under the Shannon capacity regime. Moreover, we define a metric called the effective eavesdropping rate and study the monotonicity. From the analysis of monotonicity, the existence of a maximum effective eavesdropping rate for a moderate or even high signal-to-noise (SNR) is verified. Finally, numerical results are provided and discussed. In the simulation, we also find that the maximum effective eavesdropping rate slowly increases with the blocklength.

MIMO GS OVSF/OFDM Based Underwater Acoustic Multimedia Communication Scheme

An underwater acoustic multimedia communication (UWAMC) system is proposed with 2400 transmission modes according to time-varying multipath underwater acoustic (UWA) channel conditions. The orthogonal variable spreading factor (OVSF) scheme and Gold sequence (GS) scramble code are integrated into multi-input multi-output UWAMC system based on orthogonal frequency-division multiplexing to achieve the quality of service of multimedia transmission in the UWA channel. Binary phase shift keying (BPSK) and quadrature phase shift keying (QPSK) adaptive modulation, direct mapping (DM) or space–time block code (STBC) transmission strategies, convolution channel code with rate 1/2 and 1/3, and a power assignment mechanism were adopted in the proposed system. Simulation results show that the bit error rate (BER) and power saving ratio (PSR) performance of the STBC strategy with transmission diversity is superior to that of the DM strategy without transmission diversity, and the performance of the BERs and PSRs of the transmission scheme with the GS scramble code surpasses that of the scheme without the code. The performance of the BERs and PSRs of BPSK modulation with a channel code rate of 1/3 is better than that of BPSK modulation with a channel code rate of 1/2, and the performances of BERs and PSRs of BPSK modulation with a channel code rate of 1/3 are better than that of QPSK modulation with a channel code rate of 1/3. As the length of the OVSF codes increases, the UWAMC system’s BERs decrease, and its PSRs increase. The UWAMC system can achieve either maximum transmission speed or maximum transmission power efficiency.

New adaptive link layer protocol using optimal packet length for free space optical communications

In this paper, the authors propose a new link layer protocol that adapts packet length and modulation and coding scheme (MCS) to propagation conditions for free space optical communications (FSO). In conventional FSO systems, only the MCS has been adapted with respect to propagation condition. For low signal to noise ratio (SNR), we reduce packet length and use robust modulation such as On Off Keying (OOK) and low rate channel coding. While, at high SNR, we increase packet length and use large modulation constellation. A significant enhancement in throughput is observed: the proposed link layer protocol offers 1–2 dB gain compared to the conventional one where all packets are transmitted with the same length.

A High-Rate Software-Defined Underwater Acoustic Modem With Real-Time Adaptation Capabilities

There is an emerging need for high-rate underwater acoustic (UW-A) communication platforms to enable a new generation of underwater monitoring applications including video streaming. At the same time, modern UW-A communication architectures need to be flexible to adapt and optimize their communication parameters in real time based on the environmental conditions. Existing UW-A modems are limited in terms of achievable data rates and ability to adapt the protocol stack in real time. To overcome this limitation, we present the design, implementation, and experimental evaluation of a new high-rate software-defined acoustic modem (SDAM) with real-time adaptation capabilities for UW-A communications. We introduce new physical-layer adaptation mechanisms that enable either joint adaptation of communication parameters such as modulation constellation and channel coding rate or seamless switching between different communication technologies such as orthogonal-frequency-division-multiplexing and direct-sequence-spread-spectrum. The performance of the proposed SDAM has been evaluated in both indoor (water tank) and outdoor (lake) environments. We demonstrated that the SDAM achieves 104 kbit/s with bit-error-rate (BER) of $2 \times 10^{-5}$ , 208 kbit/s with BER of 10−3, and 260 kbit/s with BER of 10−2 in real time over a 200 m horizontal link at a very-shallow lake environment.

Adaptive Digitization and Variable Channel Coding for Enhancement of Compressed Digital Mobile Fronthaul in PAM-4 Optical Links

The standardization and development of LTE-A and 5G introduced advanced wireless technologies including multiple-input multiple-output and carrier aggregation, which require multiple wireless carriers to be delivered to and from each remote radio head. The common public radio interface (CPRI) as the mainstream standard in mobile fronthaul (MFH) with on-off-keying-based optical links cannot fulfill the capacity and efficiency requirement. Instead, using compressed CPRI in a high-speed pulse-amplitude-modulation-4 (PAM-4) link is actively researched and demonstrated. In this paper, we propose and demonstrate adaptive digitization and channel coding based on these compression and capacity boosting technologies. Depending on the optical link condition, the digitization bits and channel coding rates can be adaptively and dynamically changed to achieve the lowest error vector magnitude (EVM) of wireless carriers. By separating digitization bits into high bits and low bits, the coding overhead can be different between groups, while still keeping the same bit rate per wireless carrier. Based on the existing digital MFH infrastructure, the proposed scheme can significantly improve the capacity and sensitivity in the PAM-4-based compressed digital MFH. Capacity gains from 30% to 68%, sensitivity improvement of 2–9 dB, and significant EVM improvements are demonstrated experimentally, comparing with other compressed CPRI MFH solutions.

An Optimal Resource Allocation for Superposition Coding-Based Hybrid Digital–Analog System

Hybrid digital–analog (HDA) video transmission is a new cross-layer design, which can be widely used in Internet of Things. The key problem of HDA video transmission is to find the optimal resource allocation between the digital and analog part. This paper presents a new general resource allocation algorithm for superposition coding-based HDA system. On one hand, in order to achieve successful decoding in digital part, the bitrate is controlled by the quantization parameter (QP), and the channel coding rate and modulation order are determined by the signal to interference noise power ratio, in which analog part is considered as the interference. On the other hand, the overall video quality is directly determined by the mean square error of analog part, which depends jointly on the data variance of the analog part, the power allocated to the analog part and the channel noise power. We propose a prediction model to describe how the data variance of the analog part changes with the QP in the digital part. Based on the proposed model, the power allocation of two parts can be quantitatively connected to form an optimization problem. We prove the convexity of the resource allocation problem and the gradient descent method is utilized in system implementation. With extensive simulations, the proposed algorithm is validated, achieving 1.4–5.3 dB gain over the conventional digital system, and 6.2–7.4 dB gain over pseudo-analog system in peak signal-to-noise ratio.

Joint source–channel rate allocation with unequal error protection for space image transmission

High reliable and efficient image transmission is of primary importance for space communications. In the traditional space image transmission system design, the source coding module and channel coding module are separated. This separate design, although simple to be implemented, cannot explore the transmission performance to the most. In this article, we propose a joint source–channel rate allocation framework to improve the space image transmission performance. A joint rate allocation algorithm based on the packet loss rate is proposed. For certain required system transmission rate, the optimal source coding and channel coding rate pair can be selected from pre-configured code rate sets by sliding search. In addition, our design has taken into account the progressive scalability feature of the image compression results. Each of the source coding output packets is divided into several sub-packets with different levels of significance and then channel coding with different rates is applied on these sub-packets to achieve unequal error protection. Simulation results show that the proposed joint design can significantly improve the image reconstruction quality. Compared with the traditional separate design, the proposed joint design can achieve 3–5 dB performance gain in terms of peak signal-to-noise ratio.

On the capacity of MIMO-OFDM based diversity and spatial multiplexing in Radio-over-Fiber system

This paper proposes a realistic and global simulation to predict the behavior of a Radio over Fiber (RoF) system before its realization. In this work we consider a 2 ×2 Multiple-Input Multiple-Output (MIMO) Orthogonal Frequency Division Multiplexing (OFDM) RoF system at 60 GHz. This system is based on Spatial Diversity (SD) which increases reliability (decreases probability of error) and Spatial Multiplexing (SMX) which increases data rate, but not necessarily reliability. The 60 GHz MIMO channel model employed in this work based on a lot of measured data and statistical analysis named Triple-S and Valenzuela (TSV) model. To the authors best knowledge; it is the first time that this type of TSV channel model has been employed for 60 GHz MIMO-RoF system. We have evaluated and compared the performance of this system according to the diversity technique, modulation schemes, and channel coding rate for Line-Of-Sight (LOS) desktop environment. The SMX coded is proposed as an intermediate system to improve the Signal to Noise Ratio (SNR) and the data rate. The resulting 2 ×2 MIMO-OFDM SMX system achieves a higher data rate up to 70 Gb/s with 64QAM and Forward Error Correction (FEC) limit of 10−3 over 25-km fiber transmission followed by 3-m wireless transmission using 7 GHz bandwidth of millimeter wave band.

EFFICIENT MOBILE VIDEO TRANSMISSION BASED ON A JOINT CODING SCHEME

In this paper, we propose a joint coding design which uses the Symbol Forward Error Correction (S-FEC) at the application layer. The purpose of this work is on one hand to minimize the Packet Loss Rate (PLR) and, on the other hand to maximize the visual quality of video transmitted over a wireless network (WN). The scheme proposed is founded on a FEC adaptable with the semantics of the H.264/AVC video encoding. This mechanism relies upon a rate distortion algorithm, controlling the channel code rates under the global rate constraints given by the WN. Based on a data partitioning (DP) tool, both packet type and packet length are taken into account by the proposed optimization mechanism which leads to unequal error protection (UEP). The performance of the proposed JSCC unequal error control is illustrated over wireless network by performing simulations under different channel conditions. The simulation results are then compared with an equal error protection (EEP) scheme.

Superadditivity of Quantum Channel Coding Rate With Finite Blocklength Joint Measurements

The maximum rate at which classical information can be reliably transmitted per use of a quantum channel strictly increases in general with $N$, the number of channel outputs that are detected jointly by the quantum joint-detection receiver (JDR). This phenomenon is known as superadditivity of the maximum achievable information rate over a quantum channel. We study this phenomenon for a pure-state classical-quantum (cq) channel and provide a lower bound on $C_N/N$, the maximum information rate when the JDR is restricted to making joint measurements over no more than $N$ quantum channel outputs, while allowing arbitrary classical error correction. We also show the appearance of a superadditivity phenomenon---of mathematical resemblance to the aforesaid problem---in the channel capacity of a classical discrete memoryless channel (DMC) when a concatenated coding scheme is employed, and the inner decoder is forced to make hard decisions on $N$-length inner codewords. Using this correspondence, we develop a unifying framework for the above two notions of superadditivity, and show that for our lower bound to $C_N/N$ to be equal to a given fraction of the asymptotic capacity $C$ of the respective channel, $N$ must be proportional to $V/C^2$, where $V$ is the respective channel dispersion quantity.

Error-resilient surveillance video transmission based on a practical joint source-channel distortion computing model

In recent years, IP (Internet Protocol)-based video surveillance has widely been useful for post-event analysis and assisting the work of privacy protection and public safety. To support high-quality IP video surveillance, error-resilience techniques are important for surveillance system design, because video has more stringent requirements than general video transmission for packet loss, latency, and jitter. The optimal FEC (forward error correction) code rate decision is a crucial procedure to determine the optimal source and channel coding rates to minimize the overall picture distortion when transporting video packets over packet loss channels. The conventional FEC code rate decision schemes using an analytical source-coding distortion model and a channel-induced distortion model are usually complex and typically employ the process of model parameter training, which involves potentially high computational complexity and implementation cost. To avoid the complex modeling procedure, we propose a simple but accurate joint source-channel distortion model to estimate the channel-loss threshold set for optimal FEC code rate decision. Since the proposed model is expressed as a simple closed form and has a small number of scene-dependent model parameters, a video sender of the surveillance system using the model can be easily implemented. For training the scene-dependent model parameters in real time, we propose a practical test-run procedure. This method accelerates the test-run while maintaining its accuracy for training the scene-dependent model parameters. Using the proposed simple model and practical test-run method, the video sender can find the optimal code rate for on-the-fly joint source-channel coding whenever there is a change in the packet-loss condition in the channel. Simulations show that the proposed method can accurately estimate the channel loss threshold set, resulting in an optimal FEC code rate with low computational complexity.

Adaptive error protection coding for wireless transmission of motion JPEG 2000 video

The delivery of video over wireless, error-prone transmission channels requires careful allocation of channel and source code rates, given the available bandwidth. In this paper, we present a theoretical framework to find an optimal joint channel and source code rate allocation, by considering an intra-coded video compression standard such as Motion JPEG 2000 and an error-prone wireless transmission channel. Lagrangian optimization is used to find the optimal code rate allocation, from a PSNR perspective, starting from commonly available source coding outputs, such as intermediate rate-distortion traces. The algorithm is simple and adaptive both on the available bandwidth and on the transmission channel conditions, and it has a low computational complexity. Simulation results, using Reed-Solomon (R-S) coding, show that the achieved performance, in terms of PSNR and MSSIM, is comparable with that of other methods reported in literature. In addition, a simplified and sub-optimal expression for determining the channel code assignment is also provided.

Arithmetic-BICM for Seamless Rate Adaptation for Wireless Communication Systems

Bit-interleaved coded modulation (BICM) has been widely used in capacity-achieving communication systems. In this paper, we propose an enhanced BICM, namely, arithmetic-BICM (A-BICM), which is able to achieve high channel throughput and seamless rate adaptation in time-varying wireless channels. The A-BICM generates modulation symbols through arithmetic weighted sum operations, and it adapts to data rate via adjusting the number of transmitted symbols. Our contributions in the design of A-BICM are twofold. First, we use an extrinsic information transfer chart to analyze a demapping procedure, and this facilitates the selection of the weighting multiset, which is a key design parameter in A-BICM mapping. Second, we employ a systematic channel code in A-BICM and design an adaptation scheme to strike a balance between the performances in both high- and low-SNR regions. Simulation results show that the A-BICM outperforms the conventional BICM with length-864 low-density parity check coding in both additive white Gaussian noise and fading channels. The experiments in real WLAN environments reveal that A-BICM achieves 42.1% and 20.9% throughput gain over conventional BICM and hybrid automatic repeat request rate adaptation schemes. Furthermore, the channel coding rate and its constellation in A-BICM do not need to change within a fairly wide SNR region from 1 to 30 dB to reduce communication overhead.

Performance Analysis of Energy Harvesting Wireless Communication System With Finite Blocklength

This letter investigates an achievable channel coding rate and mean delay of point-to-point energy harvesting wireless communication system with finite blocklength, which has vital practical interest. We get the achievable channel coding rate's exact formula and find it increases along with blocklength. Furthermore, the mean delay's lower-bound is also obtained.

Distortion-Power Trade-Offs in Quasi-Stationary Source Transmission over Delay and Buffer Limited Block Fading Channels

This paper investigates distortion-power tradeoffs in transmission of quasi-stationary sources over delay and buffer limited block fading channels by studying encoder and decoder buffering techniques to smooth out the source and channel variations. Four source and channel coding schemes that consider buffer and power constraints are presented to minimize the reconstructed source distortion. The first one is a high performance scheme, which benefits from optimized source and channel rate adaptation. In the second scheme, the channel coding rate is fixed and optimized along with transmission power with respect to channel and source variations; hence this scheme enjoys simplicity of implementation. The two last schemes have fixed transmission power with optimized adaptive or fixed channel coding rate. For all the proposed schemes, closed form solutions for mean distortion, optimized rate, and power are provided and in the high SNR regime, the mean distortion exponent and the asymptotic mean power gains are derived. The proposed schemes with buffering exploit the diversity due to source and channel variations. Specifically, when the buffer size is limited, fixed channel rate adaptive power scheme outperforms an adaptive rate fixed power scheme. Furthermore, analytical and numerical results demonstrate that with limited buffer size, the system performance in terms of reconstructed signal SNR saturates as transmission power increases, suggesting that appropriate buffer size selection is important to achieve a desired reconstruction quality.

On the Evaluation of the Polyanskiy-Poor–Verdú Converse Bound for Finite Block-Length Coding in AWGN

A tight converse bound to channel coding rate in the finite block-length regime and under AWGN conditions was recently proposed by Polyanskiy, Poor, and Verdu (PPV). The bound is a generalization of a number of other classical results, and it was also claimed to be equivalent to Shannon's 1959 cone packing bound. Unfortunately, its numerical evaluation is troublesome even for not too large values of the block-length n. In this paper we tackle the numerical evaluation by compactly expressing the PPV converse bound in terms of non-central chi-squared distributions, and by evaluating those through a an integral expression and a corresponding series expansion which exploit a method proposed by Temme. As a result, a robust evaluation method and new insights on the bound's asymptotics, as well as new approximate expressions, are given.

Iterative Quantum-Assisted Multi-User Detection for Multi-Carrier Interleave Division Multiple Access Systems

With the proliferation of smart-phones and tablet PCs, the data rates of wireless communications have been soaring. Hence, the need for power-efficient communications relying on low-complexity multiple-stream detectors has become more pressing than ever. As a remedy, in this paper we design low-complexity soft-input soft-output quantum-assisted multi-user detectors (QMUD), which may be conveniently incorporated into state-of-the-art iterative receivers. Our design relies on extrinsic information transfer charts. Our QMUDs are then employed in multi-carrier interleave-division multiple-access (MC-IDMA) systems, which are investigated in the context of different channel code rate and spreading factor pairs, whilst fixing the total bandwidth requirement. One of our QMUDs is found to operate within 0.5 dB of the classical maximum a posteriori probability MUD after three iterations between the MUD and the decoders, while requiring only half its complexity, at a BER of $10^{-5}$ in the uplink of a rank-deficient MC-IDMA system relying on realistic imperfect channel estimation at the receiver, while supporting 14 users transmitting QPSK symbols.

On the Dispersions of the Gel’fand–Pinsker Channel and Dirty Paper Coding

This paper studies the second-order coding rates for memoryless channels with a state sequence known non-causally at the encoder. In the case of finite alphabets, an achievability result is obtained using constant-composition random coding, and by using a small fraction of the block to transmit the empirical distribution of the state sequence. For error probabilities less than 0.5, it is shown that the second-order rate improves on an existing one based on independent and identically distributed random coding. In the Gaussian case (dirty paper coding) with an almost-sure power constraint, an achievability result is obtained using random coding over the surface of a sphere, and using a small fraction of the block to transmit a quantized description of the state power. It is shown that the second-order asymptotics are identical to the single-user Gaussian channel of the same input power without a state.

Optimum Power Control at Finite Blocklength

This paper investigates the maximal channel coding rate achievable at a given blocklength $n$ and error probability $\epsilon$, when the codewords are subject to a long-term (i.e., averaged-over-all-codeword) power constraint. The second-order term in the large-$n$ expansion of the maximal channel coding rate is characterized both for additive white Gaussian noise (AWGN) channels and for quasi-static fading channels with perfect channel state information available at both the transmitter and the receiver. It is shown that in both cases the second-order term is proportional to $\sqrt{n^{-1}\ln n}$. For the quasi-static fading case, this second-order term is achieved by truncated channel inversion, namely, by concatenating a dispersion-optimal code for an AWGN channel subject to a short-term power constraint, with a power controller that inverts the channel whenever the fading gain is above a certain threshold. Easy-to-evaluate approximations of the maximal channel coding rate are developed for both the AWGN and the quasi-static fading case.

Skalabilitas Signal to Noise Ratio (SNR) pada Pengkodean Video dengan Derau Gaussian

In video transmission, there is a possibility of packet lost an d a large load variation on the bandwidth. These are the source of network congestion, which can interfere the communication data rate. This study discusses a system to overcome the congestion with Signal-to-noise ratio (SNR) scalability-based approach, for the video sequence encoding method into two layers, which is a solution to decrease encoding mode for each packet and channel coding rate. The goal is to minimize any distortion from the source to the destination. The coding system used is a video coding standards that is MPEG-2 or H.263 with SNR scalability. The algorithm used for motion compensation, temporal redundancy and spatial redundancy is the Discrete Cosine Transform (DCT) and quantization. The transmission error is simulated by adding Gaussian noise (error) on motion vectors. From the simulation results, the SNR and Peak Signal to Noise Ratio (PSNR) in the noisy video frames decline with averages of 3dB and 4dB respectively.