In super-Nyquist wavelength division multiplexed systems, performance of forward error correction (FEC) can be improved by an iterative decoder between a maximum likelihood decoder for polybinary shaping and an FEC decoder. The typical iterative decoder includes not only the iteration between the first and second decoders but also the internal iteration within the FEC decoder. Such two-fold loop configuration would increase the computational complexity for decoding. In this paper, we propose the simplified iterative decoder, where the internal iteration in the FEC decoder is not performed, reducing the computational complexity. We numerically evaluate the bit-error rate performance of polybinary-shaped QPSK signals in the simplified iterative decoder. The numerical results show that the FEC performance can be improved in the simplified scheme, compared with the typical iterative decoder. In addition, the performance of the simplified iterative decoder has been investigated by the extrinsic information transfer (EXIT) chart.

Streaming services that use TCP have increased; however, throughput is unstable due to congestion control caused by packet loss when TCP is used. Thus, TCP control to secure a required transmission rate for streaming communication using Forward Error Correction (FEC) technology (TCP-AFEC) has been proposed. TCP-AFEC can control the appropriate transmission rate according to network conditions using a combination of TCP congestion control and FEC. However, TCP-AFEC was not developed for wireless Local Area Network (LAN) environments; thus, it requires a certain time to set the appropriate redundancy and cannot obtain the required throughput. In this paper, we demonstrate the drawbacks of TCP-AFEC in wireless LAN environments. Then, we propose a redundancy setting method that can secure the required throughput for FEC, i.e., TCP-TFEC. Finally, we show that TCP-TFEC can secure more stable throughput than TCP-AFEC.

This paper proposes novel simplified maximum likelihood detection for XOR physical layer network coding (XOR-PNC) in bi-directional wireless relay systems with Quaternary phase shift keying (QPSK). The proposed detection applies unitary precoding to achieve superior performance without computationally prohibitive exhaustive search. The performance of the XOR employing the proposed simplified MLD with the precoding is analyzed in relay systems with orthogonal frequency division multiplexing (OFDM). The performance of the XOR-PNC with the proposed techniques is also evaluated by computer simulation. The XOR-PNC with the proposed techniques achieves about 7dB better performance than the amplify-and-forward physical layer network coding in the 5-path fading channel at BER=10-4. It is also shown that the XOR-PNC with the proposed techniques achieves better performance than that without precoding.

Stochastic decoding provides ultra-low-complexity hardware for high-throughput parallel low-density parity-check (LDPC) decoders. Asynchronous stochastic decoding was proposed to demonstrate the possibility of low power dissipation and high throughput in stochastic decoders, but decoding might stop before convergence due to “lock-up”, causing error floors that also occur in synchronous stochastic decoding. In this paper, we introduce a wire-delay dependent (WDD) scheduling algorithm for asynchronous stochastic decoding in order to reduce the error floors. Instead of assigning the same delay to all computation nodes in the previous work, different computation delay is assigned to each computation node depending on its wire length. The variation of update timing increases switching activities to decrease the possibility of the “lock-up”, lowering the error floors. In addition, the WDD scheduling algorithm is simplified for the hardware implementation in order to eliminate time-averaging and multiplication functions used in the original WDD scheduling algorithm. BER performance using a regular (1024, 512) (3,6) LDPC code is simulated based on our timing model that has computation and wire delay estimated under ASPLA 90nm CMOS technology. It is demonstrated that the proposed asynchronous decoder achieves a 6.4-9.8× smaller latency than that of the synchronous decoder with a 0.25-0.3 dB coding gain.

Nonlinearity compensation algorithm and soft-decision forward error correction (FEC) are considered as key technologies for future high-capacity and long-haul optical transmission system. In this report, we experimentally demonstrate the following three benefits brought by low complexity perturbation back-propagation nonlinear compensation algorithm in 224Gb/s DP-16QAM transmission over large-Aeff pure silica core fiber; (1) improvement of pre-FEC bit error ratio, (2) reshaping noise distribution to more Gaussian, and (3) reduction of cycle slip probability.

Luby Transform (LT) codes are the first practical implementation of digital fountain codes. In LT codes, encoding symbols are independently generated so as to realize the universal property which means that performance is independent of channel parameters. The universal property makes LT codes able to provide reliable delivery simultaneously via channels of different quality while it may also limit the flexibility of LT codes. In certain application scenarios, such as real-time multimedia transmission, most receivers have tolerable channels whose erasure rates are not fixed, and channels of high erasure rate are outside the design box. In this paper, Connection Choice (CC) codes are proposed to trade the universal property for better performance. The key to CC codes is replacement of random selection with tournament selection. Tournament selection can equalize the frequency of input symbols to join encoding and change the degree distribution of input symbols. Our study indicates that CC codes with appropriate degree distributions provide better performance than the best known LT code when channels of high erasure rate can be ignored. CC codes enable system designers to customize digital fountain codes by taking into account the distribution of the erasure rate and create a new possibility for setting trade-offs between performance and erasure rate.

LT codes are the first practical rateless codes whose reception overhead totally depends on the degree distribution adopted. The capability of LT codes with a particular degree distribution named robust soliton has been theoretically analyzed; it asymptotically approaches the optimum when the message length approaches infinity. However, real applications making use of LT codes have finite number of input symbols. It is quite important to refine degree distributions because there are distributions whose performance can exceed that of the robust soliton distribution for short message length. In this work, a practical framework that employs evolutionary algorithms is proposed to search for better degree distributions. Our experiments empirically prove that the proposed framework is robust and can customize degree distributions for LT codes with different message length. The decoding error probabilities of the distributions found in the experiments compare well with those of robust soliton distributions. The significant improvement of LT codes with the optimized degree distributions is demonstrated in the paper.

In this paper, we propose a novel layered scalable- multiple description coding (LS-MDC) which offers the benefits of both scalable video coding and multiple description coding for robust video transmission over packet lossy networks. In the proposed LS-MDC method, multiple descriptions including base layer, enhancement layers, and their corresponding FEC parity data are allocated into two network paths of a path diversity system. Unlike the conventional approaches, the source base/enhancement data and their own parities in the proposed method are not transmitted together but are transferred over different paths. Therefore, the effect of burst packet losses can be effectively reduced for the descriptions. Furthermore, in order to minimize the overall distortion for the LS-MDC system and exploit the benefits of path diversity, we also propose an optimal rate allocation scheme that can adaptively control the transmission rate as well as the channel coding rate for media senders. Experiments show that the proposed method provides much better peak signal-to-noise ratio (PSNR) than conventional MDC techniques.

To establish seamless and highly robust content distribution, we proposed the new concept of Inter-Stream Forward Error Correction (FEC), an efficient data recovery method leveraging several video streams. Our previous research showed that Inter-Stream FEC had significant recovery capability compared with the conventional FEC method under ideal modeling conditions and assumptions. In this paper, we design the Inter-Stream FEC architecture in detail with a view to practical application. The functional requirements for practical feasibility are investigated, such as simplicity and flexibility. Further, the investigation clarifies a challenging problem: the increase in processing delay created by the asynchronous arrival of packets. To solve this problem, we propose a pragmatic parity stream construction method. We implement and evaluate experimentally a prototype system with Inter-Stream FEC. The results demonstrate that the proposed system could achieve high recovery performance in our experimental environment.

In this paper, we present a novel class of long quasi-cyclic low-density parity-check (QC-LDPC) codes. Each of the codes in this class has a structure formed by concatenating single-parity-check codes and QC-LDPC codes of shorter lengths, which allows for efficient, high throughput encoder/decoder implementations. Using a code in this class, we design a forward error correction (FEC) scheme for optical transmission systems and present its high throughput encoder/decoder architecture. In order to demonstrate its feasibility, we implement the architecture on a field programmable gate array (FPGA) platform. We show by both FPGA-based simulations and measurements of an optical transmission system that the FEC scheme can achieve excellent error performance and that there is no significant performance degradation due to the constraint on its structure while getting an efficient, high throughput implementation is feasible.

A high-speed low-complexity time-multiplexing Reed-Solomon-based forward error correction architecture based on the pipelined truncated inversionless Berlekamp-Massey algorithm is presented in this paper. The proposed architecture has very high speed and very low hardware complexity compared with conventional Reed-Solomon-based forward error correction architectures. Hardware complexity is improved by employing a truncated inverse Berlekamp-Massey algorithm. A high-speed and high-throughput data rate is facilitated by employing a three-parallel processing pipelining technique and modified syndrome computation block. The time-multiplexing method for pipelined truncated inversionless Berlekamp-Massey architecture is used in the parallel Reed-Solomon decoder to reduce hardware complexity. The proposed architecture has been designed and implemented with 90-nm CMOS technology. Synthesis results show that the proposed 16-channel Reed-Solomon-based forward error correction architecture requires 417,600 gates and can operate at 640 MHz to achieve a throughput of 240 Gb/s. The proposed architecture can be readily applied to Reed-Solomon-based forward error correction devices for next-generation short-reach optical communications.

This letter proposes a practical algorithm for video transmission of the scalable extension of H.264/AVC (SVC) over limited bit-rate and varying channel signal-to-noise ratio (SNR). The proposal consists of SVC source-layer dropping and layered FEC using LDPC codes to maximize the video quality. The experimental results show that the proposed method realizes better video quality than the compared unequal error protection (UEP) without source-layer dropping. This implies that the dropping of a certain number of source-layers and using the resultant bit-budget for channel coding is more effective than the other UEP case which uses all possible source-layers.

Forward error correction (FEC) is one of important and heavy tasks for wireless communication. Leading edge mobile embedded systems usually support not only one FEC standard, but multiple FEC standards in order to adapt to various wireless communication standards. In this paper, we propose two-stage configurable decoder model (2-Stage CDM) for multiple FEC standards for Viterbi and Turbo coding which have a variation under the constraint length, coding rate, etc. Proposed decoder model realizes a decoder instance which supports dedicated multiple FEC standards, and rapid design for domain specific decoder is realized. Proposed decoder model is configurable in two stages: at hardware generation time and at runtime, and designers can easily specify these specifications by various design parameters. Experimental results show proposed two-stage configurable decoder model supports various domain specific FEC decoder including existing decoder, and the decoder instances based on proposed 2-Stage CDM have sufficient throughput for each communication standard and reasonable area overhead compared with existing decoder.

This paper presents a high-throughput low-complexity four-parallel Reed-Solomon (RS) decoder for high-rate WPAN systems. Four-parallel processing is used to achieve 12-Gbps data throughput and low hardware complexity. Also, the proposed pipelined folded Degree-Computationless Modified Euclidean (fDCME) algorithm is used to implement the key equation solver (KES) block, which provides low hardware complexity for the RS decoder. The proposed four-parallel RS decoder is implemented 90-nm CMOS technology optimized for a 1.2 V supply voltage. The implementation result shows that the proposed RS decoder can be operated at a clock frequency of 400 MHz and has a data throughput 12.8-Gbps. The proposed four-parallel RS decoder architecture has high data processing rate and low hardware complexity. Therefore it can be applied in the FEC devices for next-generation high-rate WPAN systems with data rate of 10-Gbps and beyond.

Joint source-channel coding (JSCC) is a method to jointly allocate the given total transmission bitrate to the source coding and channel coding to maximize the video quality at the receiving end. In this paper, we propose a practical model for efficiently determining a near-optimal code rate for JSCC in real-time video communications. The conventional code rate decision schemes using analytical source coding distortion model and 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 introduce a very simple video quality model based on the playable bitrate which is defined as the total bit amount per unit time that is not affected by the channel loss during transmission including correctly recovered bits by the channel decoder. Because the video quality at the receiving end is clearly commensurate with the playable bitrate, we can easily determine the quality-oriented near-optimal code rate by finding the code rate that maximizes the playable bitrate at the sender side. The proposed playable bitrate model is very simple because it does not require the complex training procedure for obtaining model parameters, which is usually required in the conventional code rate decision method. It is shown by simulations that the proposed code rate decision scheme based on the playable bitrate model can efficiently determine the near-optimal code rate for JSCC in terms of high accuracy on the optimal code rate.

This paper proposes Twin Turbo (T2) MIMO-OFDM (Multiple Input Multiple Output-Orthogonal Frequency Division Multiplexing). The advanced iterative decoder, called the T2 decoder, decreases the transmission error rate compared to conventional turbo decoders because it uses the correlation information among the bits mapped on an identical symbol of multi-level modulation and updates the channel reliability. When T2 is applied to a MIMO-OFDM, the required symbol energy to noise power density ratio (Es/N0) can be reduced more effectively than when T2 is applied to SISO (Single Input Single Output). This is because T2 can use the correlation among the bits not only mapped on an identical symbol but also transmitted from different antennas. Moreover, T2 achieves good performance in a correlated MIMO channel because the average minimum squared Euclidean distances between symbol replica candidates consisting of signals transmitted from multiple transmitter antennas are reduced. Computer simulations verify that the required Es/N0 of T2 MIMO-OFDM using 16QAM is 1.9 dB lower than that of a conventional turbo decoder when the correlation coefficients of transmitter and receiver antennas are 0.8. A computational complexity analysis clarifies the relation between the increase in computational complexity and the reduction in the required Es/N0.

This letter presents an iterative decision feedback channel estimation scheme for burst mode COFDM transmission. The feature of the proposed scheme is that the channel estimation using metrics comparison is applied to the initial stage of the iterative mechanism, which makes it possible to provide a reliable data stream at the initial stage. Computer simulation results show that the proposed approach provides better BER than the traditional iterative decision feedback channel estimation scheme irrespective of the number of iterations.

Efficient real-time contents distribution services on the Internet are only possible by suppressing the influence of packet losses. One solution for UDP transmission is the use of Forward Error Correction (FEC) based on Reed-Solomon codes. However, a more efficient method is required since this causes the increase of network traffic and includes the weakness to burst packet losses. In this paper, we propose a data recovery method that generates redundant data with the combination of Reed-Solomon codes and convolution of neighboring blocks. We realize the small amount of redundancy and the high reliability in data transmission compared with using only Reed-Solomon codes in the environment that burst packet losses are occurred frequently. We implement proposal method into the network bridge and confirm its efficiency from the viewpoint of data reconstruction from burst packet losses.

Packet loss and delay cause degradation in the quality of real-time, interactive applications such as video conferencing. Forward error correction (FEC) schemes have been proposed to make the applications more resilient to packet loss, because the time required to recover the lost packets is shorter than that required to retransmit the lost packets. On the other hand, the codec generally used in real-time applications like MPEG4 has the feature that the sending bit rate and the packet size of the traffic vary significantly according to the motion of an object in a video. If the traditional FEC coding, which is calculated on the basis of a fixed-size block, is applied to such applications, a waste of bandwidth and a delay variation are caused and the quality is degraded. In this paper, we propose suitable FEC schemes for visual communication systems using variable bit-rate (VBR) codec and evaluate the effectiveness of these schemes using our prototype implementation and experimental network.

The history of forward error correction in optical communications is reviewed. The various types of FEC are classified as belonging to three generations. The first generation FEC represents the first to be successful in submarine systems, when the use of RS(255, 239) became widespread as ITU-T G.975, and also as G.709 for terrestrial systems. As WDM systems matured, a quest began for a stronger second generation FEC. Several types of concatenated code were proposed for this, and were installed in commercial systems. The advent of third-generation FEC opened up new vistas for the next generation of optical communication systems. Thanks to soft decision decoding and block turbo codes, a net coding gain of 10.1 dB has been demonstrated experimentally. That brought us a number of positive impacts on existing systems. Each new generation of FEC was compared in terms of the ultimate coding gain. The Shannon limit was discussed for hard or soft decision decoding. Several functionalities employing the FEC framing were introduced, such as overall wrapping by the FEC frame enabling the asynchronous multiplexing of different clients' data. Fast polarization scrambling with FEC was effective in mitigating polarization mode dispersion, and the error monitor function proved useful for the adaptive equalization of both chromatic dispersion and PMD.