Code Rate

This paper concerns to blind identification of a convolutional code with desired rate in a noiseless transmission scenario. To the best of our knowledge, blind estimation of convolutional code based on only the received bitstream doesn't lead to a unique solution. Hence, without loss of generality, we will assume that the transmitter employs a non-catastrophic encoder. Moreover, we consider a complete synchronous scenario in which one can extract separate codewords from received sequence. This assumption is valid in many practical communication systems because, the frame preambles allow us to identify the beginning of each codewords. In this paper, we examine the blind identification problem for rate and rate convolutional codes, respectively. For rate , we propose an iterative method that uses three steps in each iteration to test the validity of a possible value of . We show that this method can identify the parameters of a rate convolutional code from only two different noiseless received codewords. Afterwards, we generalize this method for a rate convolutional code in which each iteration is composed of seven successive steps. We show that this method requires at least different codewords to identify all parameters of a rate code.

This paper concerns to blind identification of a convolutional code with desired rate in a noiseless transmission scenario. To the best of our knowledge, blind estimation of convolutional code based on only the received bitstream doesn’t lead to a unique solution. Hence, without loss of generality, we will assume that the transmitter employs a non-catastrophic encoder. Moreover, we consider a complete synchronous scenario in which one can extract separate codewords from received sequence. This assumption is valid in many practical communication systems because, the frame preambles allow us to identify the beginning of each codewords.In this paper, we examine the blind identification problem for rate1/n and rate k/n convolutional codes, respectively. For rate 1/n, we propose an iterative method that uses three steps in each iteration to test the validity of a possible value of n. We show that this method can identify the parameters of a rate 1/n convolutional code from only two diffe...

This paper devotes to comparison of different coding schemes (various constructions of Polar and LDPC codes, Product codes and BCH codes) for the case when information is transmitted over AWGN channel with quantization with lowest possible complexity and resolution: 1-bit. We examine performance (in terms of Frame-error-rate -FER) for schemes mentioned above and give some reasoning for results we obtained. Also we give some recommendations for choosing coding schemes for a given code rate and code length.

To improve the performance of polar code parameter recognition in the fields of intelligent communication, communication detection, and network countermeasures, we propose a new recognition scheme for the additive white Gaussian noise (AWGN) channel. The scheme turns parameter recognition problems into hypothetical tests and is effective due to the check relationship between the received codeword and the dual space determined by the correct parameters. First, a sub-matrix is obtained by removing the frozen-bit-index rows of the polar code generator matrix, and then its dual matrix is calculated. To check the relationship of the dual matrix and codewords, the average likelihood ratio of codewords is introduced as a test statistic, and then the corresponding decision threshold is deduced. Next, the degree of conformity of polar code recognition is defined, and the minimum code length and code rate corresponding to the highest degree of conformity are chosen to calculate the index of information-bit positions by a Gaussian approximation (GA) construction algorithm. Finally, the chosen code length, code rate, and corresponding index are provided as the recognition results. Simulation results show that the algorithm can achieve effective parameter recognition in both high-signal-to-noise (SNR) and low-SNR environments, and that the algorithm's recognition performance increases with decreasing code rate and code length. The parameter recognition rate with a code length of 128 and code rate of 1/5 is close to 100% when the SNR is 4 dB, and the algorithm complexity increases almost linearly with the decreasing code rate and the increasing length of the intercepted data.

In this paper, we present the impact of the multi-diagonal (MD) code for high data rate in spectral amplitude coding optical code division multiple-access (SAC-OCDMA) networks. An MD code structure for SAC-OCDMA system is presented. The MD code can be flexibly generated for any weight and number of users by using a combination of diagonal matrices. Several advantages of the MD code family are: the cross-correlation value remains zero even in case of an increased number of users for any weight value, supporting large number of users, and easy code construction. Thus, these features show the MD code family be a solution to enhance the data rate of SAC-OCDMA networks. Based on the theoretical and simulation evaluation, MD code is shown to provide a much better performance compared to existing SAC-OCDMA codes. The ability of MD code to support simultaneous transmissions at different high data rate has been successfully established through the simulated results of the 0.24 Tbps (12 Gbps ...

I present evidence that fundamental physics constants exhibit network coherence properties exceeding quantum error correction thresholds by approximately 60×. Using adversarial corruption testing on the HoloCell architectural manifold, I demonstrate that when 7 of 12 physics constants are replaced with random values, the remaining 5 true constants pull evolved integer sets back to the architectural manifold {1,

The embedded block coding with optimized truncation (EBCOT) algorithm has been adopted by the JPEG2000 standard. In which, the post compression rate distortion (PCRD)optimization needs a large memory space to store all the code streams of code blocks; however, code blocks of less importance might not be used for the optimal decoded image at a bit rate. To avoid waste of computational power and memory space, a simple context based rate distortion estimation (CBRDE) is proposed to arrange the scanning order of code blocks in an adaptive manner. CBRDE is based on the MQ table, which is available at both encoder and decode. Thus, there is no need to store and transmit the rate distortion information of code blocks. Experimental results show that the rate distortion curves are almost convex; this demonstrates the benefit of the proposed CBRDE.

A two-level, hybrid-optimization scheme is proposed for rate allocation of JPEG2000(J2K) transmission over noisy channels. It combines forward error correction (FEC) and J2K error resilience modes to minimize the expected end-to-end image distortion. Subject to a total target bit rate, it forms fixed-length source packets, and can be adapted to fixed-length channel packets in a straightforward manner. The proposed scheme utilizes the source distortionrate characteristics and the BER statistics of the available channel codes. It has complexity less than ¢¡ ¤£ ¦¥ ¨ § for a specific J2K codestream, where £ is the number of available channel code rates. Performance results with both RCPC and turbo codes for fixed-length source packets are provided to show the effectiveness of the scheme.

In this paper we summarize and expound upon the choices made for the BICM (Bit-Interleaved Coded Modulation) part of the next-generation terrestrial broadcast standard known as ATSC 3.0. The structure of the ATSC 3.0 BICM consists of a forward error correcting (FEC) code, bit interleaver and constellation mapper. In order to achieve high efficiency over a wide range of reception conditions and Carrier-to-Noise (C/N) ratio values, several notable new elements have been standardized. First, twenty four original Low-Density Parity Check (LDPC) codes have been designed, with coding rates from 2/15 (0.13) up to 13/15 (0.87) for two code sizes: 16200 bits and 64800 bits. Two different LDPC structures have been adopted; one structure more suited to medium and high coding rates and another structure suited to very low coding rates. Second, in addition to QPSK, Non-Uniform Constellations (NUCs) have been chosen for constellation sizes from 16QAM to 4096QAM to bridge the gap to the Shannon theoretical limit. Two different types of NUCs have been proposed: one-dimensional NUCs (1D-NUC) for 1024-and 4096-point constellations, and two-dimensional NUCs (2D-NUCs) for 16-, 64-and 256-point constellations. 2D-NUCs achieve a better performance than 1D-NUCs but with a higher complexity since they cannot be separated into two independent I/Q components. NUCs have been optimized for each coding rate for the 64800 bits LPDCs. The same constellations are used for 16200 bits LDPCs, although they have been limited up to 256QAM. Finally, a bit interleaver, optimized for each NUC/coding rate combination, has been designed to maximize the performance. The result is a BICM that provides the largest operating range (more than 30 dB, with the most robust mode operating below -5 dB C/N) and the highest spectral efficiency compared to any digital terrestrial broadcast system today, outperforming the current state-of-the-art DVB-T2 standard BICM by as much as 1 dB in some cases. ATSC 3.0 will also provide a considerable increase in the maximum transmission capacity when using the high-order non-uniform constellations such as 1024QAM and 4096QAM, which will represent a major milestone for terrestrial broadcasting since the highest order constellation currently available is uniform 256QAM. This paper describes the coding, modulation and bit interleaving modules of the BICM block of ATSC 3.0, and compares its performance with other DTT standards such as ATSC A/53 and DVB-T2.

Adaptive coding, modulation and power allocation are well‐known techniques for enhancing the performance of systems transmitting over quasi‐static fading channels. Especially their application in orthogonal frequency division multiplexing (OFDM)‐based systems promises gains in spectral efficiency. However, an open issue is the joint optimisation of code‐rate, modulation formats and power allocation if codewords are mapped onto sub‐symbols with heterogeneous channel quality and modulation formats. In this paper, we present an adaptive coding, modulation and, optionally, power allocation scheme that maximises the throughput under a codeword error rate constraint for bit‐interleaved coded OFDM systems using rate‐compatible punctured block‐circulant low‐density parity‐check codes (RCP‐BLDPC). The average mutual‐information per code bit is used as single parameter for the codeword error probability estimation. The algorithms are evaluated within the context of the OFDMA/TDMA system desig...

Abstract—Within the EU FP6 Integrated Project WINNER, adaptive transmission in a TDMA/OFDMA system is investigated as a key technology for boosting the spectral efficiency of a new radio interface for 4G systems. Adaptive allocation and linkadaptation of ...

An important function of any modern digital communication system is error control coding (ECC). Such coding is the field of communications that deals with techniques for detecting and correcting errors in a signal. ECC is especially useful in wireless communication systems. RS codes are the most powerful in the family of linear block codes and are arguably the most widely used type of error control codes. RS code is a type of Forward Error Correction (FEC) code and it is a nonbinary, linear and cyclic block error correcting code. In this paper, the proposed work is to implement the encoder and decoder of Reed-Solomon (RS) coding scheme on the platform of VLSI using the Euclid’s algorithm. Implementation will be done on VLSI Hardware Description Language (VHDL) and the operation and results can be seen on Field Programmable

In a packet radio network that employs frequency‐hop transmission and error control coding, the use of side information permits identification and erasure of symbols that have been corrupted by multiple‐access interference, and thus greatly enhances multiple‐access capability. In one approach, side information is generated from an error detecting random code. We investigate the trade‐off between the reliability of side information and the rate of the code, and derive the maximum allowable code rate to achieve a certain reliability of side information. An achievable region and the channel throughput are examined for an unslotted frequency‐hop packet radio network that employs Reed‐Solomon codes and side information generated within the radio by an error detecting code.

This paper explores the design of multiservice code (MS) in spectral, time spreading and wavelength domains (3-Dimensional). The proposed 3D code is designed in such a manner that follows the ideal in phase unit cross correlation in spectral, time spreading and wavelength domains. The proposed encoder generates the MS code effectively in 3D domain and decoder suppresses the multi-user interferences successfully. Performance analyses are carried out by considering the all noises such as phase induce intensity noise(PIIN), shot noise and thermal noise. The analysis of 3D-MS code for variable code length is also performed for measurement of variation in bit error rate against the received power and number of active users in the design. The observed performances are also compared with existing codes such as 2D-MDW, 3D-PDC and 1D-MS code and the proposed code shows the better performance. The analysis of bit error rate of proposed code is carried out against the number of users at 0.622Gbits/s data rate and 0 dBm received power and comparison is also formed with the existing code such as 3D PD(M = 21, N = 3, P = 3), 2DMDW(M = 63, P = 3), PDC(M = 57, P = 3) and 1D MS Code. It is noted that 3D MS code shows the better performance than the existing methods. Analysis is also evaluated for variable weight and variable length codes in order to observe the variation in bit error rate with variation in received power and number of simultaneous users. Keywords: optical code division multiple access (OCDMA), multiple access interference (MAI), modified double code (MDW), multiservice code (MS), perfect difference code (PDC), three dimensional (3D)

Turbo-coded burst-by-burst adaptive orthogonal frequency division multiplex (AOFDM) wide-band speech transceivers are proposed. A constant throughput adaptive OFDM transceiver was designed and benchmarked against a time-variant rate scheme. The proposed joint adaptation of source-codec, channel-codec, and modulation regime results in attractive, robust, high-quality audio systems, capable of conveying near-unimpaired wide-band audio signals over fading dispersive channels for signal-to-noise ratios (SNR) in excess of about 5 dB. Index Terms-Adaptive OFDM, audio compression, transmission, wide-band speech compression. I. BACKGROUND AND MOTIVATION B URST-BY-BURST adaptive quadrature amplitude modulation (AQAM) transceivers [1] have recently generated substantial research interests in the wireless communications community [2]-[4]. The transceiver reconfigures itself on a burst-by-burst basis, depending on the instantaneous perceived wireless channel quality. More explicitly, the associated channel quality to be experienced by the next transmission burst is estimated on the basis of the current received burst and the specific modulation mode, which is expected to achieve the required performance target is then selected for the transmission of the next burst. In other words, modulation schemes of different robustness and of different data throughput are invoked [5], [6]. In the event of an anticipated error burst due to a low expected instantaneous channel quality, the transmitter can also be temporarily disabled, while the data are delayed and buffered, until the channel quality improves, provided that the associated delay is not excessive for the service supported. Due to this feature, the distribution of channel errors becomes typically less bursty, than in conjunction with nonadaptive modems. This is an attractive feature in conjunction with channel coding, resulting in potentially increased coding gains [5]. Furthermore, the soft-decision channel codec metrics can also be invoked in estimating the instantaneous channel quality [5]. Recently, block turbo-coded AQAM transceivers have also been proposed for dispersive wide-band channels in conjunction with conventional decision feedback equalizers (DFE) [5], where the mean squared error (MSE) at the DFE's output was used as the channel quality metric, controlling the choice of modem modes. An alternative neural-network radial basis function (RBF) DFE-based AQAM modem design was Manuscript

The goal of modern communication systems is to transmit information from one point to another through a communication channel quickly and effectively. Mobile wireless users will move around while always being connected to the network. This necessitates the transmission of digital material through wireless networks and satellites, which has grown to be a significant problem over time. The main goal of this study is to compare the performance and efficiency of OFDM to that of other coding schemes, such as LDPC, Turbo, and Convolution. In this simulation, multiple forms of modulation, such as 4 QPSK with a coding rate of 0.663 and a BER of 0.001 at 0.59 dB Eb/No and an efficiency of 1.326, were compared with LDPC, Turbo, and Convolution codes. The performance is improved in the following simulated comparison of 16 QAM with a coding rate of 0.479, which has a BER of 0.001, an Eb/No of 3.26 dB, and an efficiency of 1.914. Using unique, not tried before LDPC code rate of 0.313, with effic...

This paper investigates the use of a spread spectrum OFDM modulation to provide a variable rate communications link that is tolerant of multipath fading. The system adaptively varies the amount of spreading in response to changing channel conditions in an attempt to optimize throughput while maintaining acceptable error rate performance. System performance results for AWGN and Rayleigh fading channels, both frequency selective and frequency non-selective, show that a desired BER performance can be maintained, A trade-off analysis between Forwa,rd Error Correction (F'EC) coding and spreadhg for a fixed bandwidth expansion factor is performed. For convolutional coding the results shown tha,t a low-rate code provides better performimce while, for a Reed-Solomon code, a coding rate of approximately 1/2 is best.

When the fading channels are encountered during data communication, errors are likely to occur at the receiving end due to multipath propagation. Researchers have been consistently striving to develop Error Correction Schemes that can effectively handle these errors and ensure error-free data reception at the receiver end. Of particular interest are the Forward Error Correction Schemes that can be implemented at the transmitter end itself. However, the implementation of error correction coding through these schemes incurs additional costs in terms of bandwidth expansion, as extra bits need to be added to facilitate error correction. Fortunately, there exists one coding scheme called Trellis Coded Modulation (TCM), which addresses this issue. TCM selects a modulation scheme based on the rate of the convolutional coding scheme. However, this coding technique has limitations in correcting the number of errors, leading to the development of Turbo Coding. This scheme utilizes two coders at the transmitter, arranged in either serial or parallel configuration, and a suitable decoder at the receiver. A design of Turbo Coding scheme has been presented in this paper, that employs convolutional coders having rate 2/3, in a serially concatenated configuration, providing an effective rate of 4/9. This turbo coding scheme is then applied to TCM scheme in order to preserve the bandwidth. Therefore, if using the convolutional coding scheme of rate 2/3, the modulation scheme is 8-QAM and in order to preserve bandwidth after coding, using the Turbo coding scheme of rate 4/9, then the modulation scheme will be 512-QAM. The simulations have been conducted in MATLAB and the error correcting capabilities of the designed scheme in comparison with convolutional coding scheme using the constituent convolutional encoder have also been compared. It has been observed that in the Rician fading channel conditions, the Turbo Trellis Coding Modulation Scheme provides approximately 5 dB gain compared to the convolutional coding scheme.

This paper investigates the outage behavior of transmitting correlated binary sources with arbitrary code rate over quasi-static Rayleigh fading multiple access channel (MAC). The sufficient condition for lossless communication with arbitrary code rate is obtained by assuming separate source and channel coding, which indicates the two rate regions specified by the Slepian-Wolf theorem and the MAC capacity region intersect. An upper bound of the outage probability is derived based on the sufficient condition. Asymptotic property shows the second order diversity gain can be achieved only when the tow sources are fully correlated. Numerical results demonstrate the accuracy of the derived upper bound. The-outage achievable rate, which is the largest rate of transmission such that the outage probability is smaller than a predefined value, is also analyzed. It is found the-outage achievable rate becomes lager with higher transmit power and/or stronger source correlation.

This letter investigates bit-interleaved coded multiple beamforming (BICMB) with perfect coding in millimeter-wave (mm-wave) massive multiple-input multipleoutput (MIMO) systems to achieve both maximum diversity gain and multiplexing gain. Using perfect coding with BICMB enables us to do this. We show that by using BICMB and perfect coding, the diversity gain becomes independent from the number of transmitted data streams and the number of antennas in each remote antenna unit (RAU) at the transmitter and the receiver. The assumption is that the perfect channel state information (CSI) is known at both the transmitter and the receiver and the number of antennas goes to infinity. This latter assumption can be relaxed by a large number of antennas in each RAU, similar to the case for all massive MIMO research. Simulation results show that when the perfect channel state information assumption is satisfied, the use of BICMB with perfect coding results in the diversity gain values predicted by the analysis.

In this paper, we present the diversity order analysis of bit-interleaved coded multiple beamforming (BICMB) combined with the constellation precoding scheme. Multiple beamforming is realized by singular value decomposition of the channel matrix which is assumed to be perfectly known to the transmitter as well as the receiver. Previously, BICMB is known to have a diversity order bound related with the product of the code rate and the number of parallel subchannels, losing the full diversity order in some cases. In this paper, we show that BICMB combined with the constellation precoder and maximum likelihood detection achieves the full diversity order. We also provide simulation results that match the analysis.

When the fading channels are encountered during data communication, errors are likely to occur at the receiving end due to multipath propagation. Researchers have been consistently striving to develop Error Correction Schemes that can effectively handle these errors and ensure error-free data reception at the receiver end. Of particular interest are the Forward Error Correction Schemes that can be implemented at the transmitter end itself. However, the implementation of error correction coding through these schemes incurs additional costs in terms of bandwidth expansion, as extra bits need to be added to facilitate error correction. Fortunately, there exists one coding scheme called Trellis Coded Modulation (TCM), which addresses this issue. TCM selects a modulation scheme based on the rate of the convolutional coding scheme. However, this coding technique has limitations in correcting the number of errors, leading to the development of Turbo Coding. This scheme utilizes two coders at the transmitter, arranged in either serial or parallel configuration, and a suitable decoder at the receiver. A design of Turbo Coding scheme has been presented in this paper, that employs convolutional coders having rate 2/3, in a serially concatenated configuration, providing an effective rate of 4/9. This turbo coding scheme is then applied to TCM scheme in order to preserve the bandwidth. Therefore, if using the convolutional coding scheme of rate 2/3, the modulation scheme is 8-QAM and in order to preserve bandwidth after coding, using the Turbo coding scheme of rate 4/9, then the modulation scheme will be 512-QAM. The simulations have been conducted in MATLAB and the error correcting capabilities of the designed scheme in comparison with convolutional coding scheme using the constituent convolutional encoder have also been compared. It has been observed that in the Rician fading channel conditions, the Turbo Trellis Coding Modulation Scheme provides approximately 5 dB gain compared to the convolutional coding scheme.

When the fading channels are encountered during data communication, errors are likely to occur at the receiving end due to multipath propagation. Researchers have been consistently striving to develop Error Correction Schemes that can effectively handle these errors and ensure error-free data reception at the receiver end. Of particular interest are the Forward Error Correction Schemes that can be implemented at the transmitter end itself. However, the implementation of error correction coding through these schemes incurs additional costs in terms of bandwidth expansion, as extra bits need to be added to facilitate error correction. Fortunately, there exists one coding scheme called Trellis Coded Modulation (TCM), which addresses this issue. TCM selects a modulation scheme based on the rate of the convolutional coding scheme. However, this coding technique has limitations in correcting the number of errors, leading to the development of Turbo Coding. This scheme utilizes two coders at the transmitter, arranged in either serial or parallel configuration, and a suitable decoder at the receiver. A design of Turbo Coding scheme has been presented in this paper, that employs convolutional coders having rate 2/3, in a serially concatenated configuration, providing an effective rate of 4/9. This turbo coding scheme is then applied to TCM scheme in order to preserve the bandwidth. Therefore, if using the convolutional coding scheme of rate 2/3, the modulation scheme is 8-QAM and in order to preserve bandwidth after coding, using the Turbo coding scheme of rate 4/9, then the modulation scheme will be 512-QAM. The simulations have been conducted in MATLAB and the error correcting capabilities of the designed scheme in comparison with convolutional coding scheme using the constituent convolutional encoder have also been compared. It has been observed that in the Rician fading channel conditions, the Turbo Trellis Coding Modulation Scheme provides approximately 5 dB gain compared to the convolutional coding scheme.

When the fading channels are encountered during data communication, errors are likely to occur at the receiving end due to multipath propagation. Researchers have been consistently striving to develop Error Correction Schemes that can effectively handle these errors and ensure error-free data reception at the receiver end. Of particular interest are the Forward Error Correction Schemes that can be implemented at the transmitter end itself. However, the implementation of error correction coding through these schemes incurs additional costs in terms of bandwidth expansion, as extra bits need to be added to facilitate error correction. Fortunately, there exists one coding scheme called Trellis Coded Modulation (TCM), which addresses this issue. TCM selects a modulation scheme based on the rate of the convolutional coding scheme. However, this coding technique has limitations in correcting the number of errors, leading to the development of Turbo Coding. This scheme utilizes two coders at the transmitter, arranged in either serial or parallel configuration, and a suitable decoder at the receiver. A design of Turbo Coding scheme has been presented in this paper, that employs convolutional coders having rate 2/3, in a serially concatenated configuration, providing an effective rate of 4/9. This turbo coding scheme is then applied to TCM scheme in order to preserve the bandwidth. Therefore, if using the convolutional coding scheme of rate 2/3, the modulation scheme is 8-QAM and in order to preserve bandwidth after coding, using the Turbo coding scheme of rate 4/9, then the modulation scheme will be 512-QAM. The simulations have been conducted in MATLAB and the error correcting capabilities of the designed scheme in comparison with convolutional coding scheme using the constituent convolutional encoder have also been compared. It has been observed that in the Rician fading channel conditions, the Turbo Trellis Coding Modulation Scheme provides approximately 5 dB gain compared to the convolutional coding scheme.

This paper focus on performance of code Zero Correlation-Correlation (ZCC) in free space optical communication. The ZCC code has a superior characteristic which eliminate the overlapping code between any users. Due to this high class characteristic, the code improves the performance of the conventional code in free space optical environment. In this paper the analysis performance of bit error rate is considering the avalanche (APD) noise, thermal noise and multiuser interference. The result shows that ZCC code improve the performance of conventional code in term of number of users, power received and data bit rate.

This paper aims at designing a Seamless Rate Adaptation for wireless networking which achieves smooth rate adjustment in a broad dynamic range of channel conditions. Conventional rate adaptation can only achieve a staircase rate adjustment. Even when combining with hybrid ARQ, it suffers from an irreconcilable conflict between throughput and dynamic range. We tackle this problem from a new perspective by relying on modulation, instead of channel coding, for rate adaptation. We propose rate compatible modulation (RCM), in which modulation signals are incrementally generated from information bits through weighted mapping. Rate adaptation is achieved through varying the number of modulated signals. As more signals are transmitted, information bits gradually accumulate energy. The weights in bit-to-symbol mapping are delicately designed to ensure fine-grained energy accumulation so that smoothness and efficiency can both be achieved. We design and implement a rate adaptation system, called SRA and evaluate its performance through a software radio testbed. Results show that, under highly dynamic channel conditions, SRA achieves over 80% throughput gain over 802.11a adaptive modulation and coding, and achieves 28.8% and 43.8% gain over HARQ systems implemented with Turbo code and Raptor code. We believe that the concept of rate compatible modulation opens up a fresh research avenue toward the wireless rate adaptation problem.

This paper discusses the application of adaptive modulation and adaptive rate turbo coding to orthogonal frequency-division multiplexing (OFDM), to increase throughput on the time and frequency selective channel. The adaptive turbo code scheme is based on a subband adaptive method, and compares two adaptive systems: a conventional approach where a separate turbo code is used for each subband, and a single turbo code adaptive system which uses a single turbo code over all subbands. Five modulation schemes (BPSK, QPSK, 8AMPM, 16QAM, and 64QAM) are employed and turbo code rates considered are 1/2 and 1/3. The performances of both systems with high (10 −2) and low (10 −4) BER targets are compared. Simulation results for throughput and BER show that the single turbo code adaptive system provides a significant improvement.

Homophonic coding is a framework to reversibly convert a message into a sequence with some target distribution. This is a promising tool to generate a codeword with a biased code-symbol distribution, which is required for capacityachieving communication by asymmetric channels. It is known that asymptotically optimal homophonic coding can be realized by a Fixed-to-Variable (FV) length code using an interval algorithm similar to a random number generator. However, FV codes are not preferable as a component of channel codes since a decoding error propagates to all subsequent codewords. As a solution for this problem an asymptotically optimal Variable-to-Fixed (VF) length homophonic code, dual Shannon-Fano-Elias-Gray (dual SFEG) code, is proposed in this paper. This code can be interpreted as a dual of a modified Shannon-Fano-Elias (SFE) code based on Gray code. It is also shown as a by-product that the modified SFE code, named SFEG code, achieves a better coding rate than the original SFE code in lossless source coding.

Homophonic coding is a framework to reversibly convert a message into a sequence with some target distribution. This is a promising tool to generate a codeword with a biased code-symbol distribution, which is required for capacityachieving communication by asymmetric channels. It is known that asymptotically optimal homophonic coding can be realized by a Fixed-to-Variable (FV) length code using an interval algorithm similar to a random number generator. However, FV codes are not preferable as a component of channel codes since a decoding error propagates to all subsequent codewords. As a solution for this problem an asymptotically optimal Variable-to-Fixed (VF) length homophonic code, dual Shannon-Fano-Elias-Gray (dual SFEG) code, is proposed in this paper. This code can be interpreted as a dual of a modified Shannon-Fano-Elias (SFE) code based on Gray code. It is also shown as a by-product that the modified SFE code, named SFEG code, achieves a better coding rate than the original SFE code in lossless source coding.

The authors face the problem of designing good LDPC codes for applications requiring variable, that is adaptive, rates. More precisely, the object of the paper is twofold. On one hand, we propose a deterministic (not random) procedureto construct good LDPC codes without constraints on the code dimension and rate. The method is based on the analysis and optimization of the local cycles length in the Tanner graph and gives the designer the chance to control complexity of the designed codes. On the other hand, we present a novel puncturing strategy which acts directly on the parity check matrix of the code, starting from the lowest rate needed, in order to allow the design of higher rate codes avoiding additional complexity of the co/decoding hardware. The efficiency of the proposed solution is tested through a number of numerical simulations. In particular, the puncturing strategy is applied for designing codes with rate variable between 0.715 and 0.906. The designed codes are used in...

This paper begins with an investigation of the optimization of binary turbo code through good interleaver design. For this purpose, different types of interleaver have designed here and have evaluated their performance. Basically the performances of the turbo code using block interleaver, helical interleaver, random interleaver and odd-even interleaver have evaluated. From this investigation it has been seen that for small code length the performance of the block interleaver is superior in non-puncturing case and the performance of the odd-even interleaver is superior in puncturing case, but for large code length the performance of the random interleaver is better in both of puncturing and non-puncturing condition. In this paper, it has investigated that the performance of the odd-even interleaver (block interleaver with odd number of rows and columns) significantly increased in puncturing conditions. Index terms— Turbo encoder; Turbo decoder; Interleaver; code rate; Puncturing; Non...

We present a universal coding scheme, based on polar codes, that can achieve the compound capacity of any finite set of binary-input asymmetric channels. The scheme is a hybrid combination of Honda and Yamamoto's polar coding scheme for asymmetric channels and a universal polar coding scheme for symmetric channels proposed by Hassani and Urbanke. In the proposed universal construction for the asymmetric setting, we exploit the staircase structure in the universal scheme for symmetric channels to define a coding strategy that requires neither storage-intensive shared boolean functions nor a sidechannel between encoder and decoder in order to transmit bits corresponding to bit-channels that are not completely polarized.

We present a universal coding scheme, based on polar codes, that can achieve the compound capacity of any finite set of binary-input asymmetric channels. The scheme is a hybrid combination of Honda and Yamamoto's polar coding scheme for asymmetric channels and a universal polar coding scheme for symmetric channels proposed by Hassani and Urbanke. In the proposed universal construction for the asymmetric setting, we exploit the staircase structure in the universal scheme for symmetric channels to define a coding strategy that requires neither storage-intensive shared boolean functions nor a sidechannel between encoder and decoder in order to transmit bits corresponding to bit-channels that are not completely polarized.

Full-rate is very important in any data transmission coding. For transmitting data at low bit rate than full-rate code, higher modulation scheme is required. But it is impossible to design full rate orthogonal designs with complex constellation for more than two transmit antennas. Only Alamouti code provides full-rate for two transmit antennas. In this paper, Bit Error Rate (BER) is calculated for Quasi-Orthogonal Space-time Block Coding (QOSTBC). Here we work with Rayleigh fading channel. We consider the codes which decodes pairs of symbols instead of simple separate decoding like Orthogonal Space-Time Block Coding. In Quasi-Orthogonal Space-time Block Code full-rate is achieved but full-diversity is sacrificed. Diversity is the most important techniques for providing reliable communication over fading channels. One of the diversity techniques that uses multiple transmit and/or receive antennas is space diversity. Multiple antenna technique provides a space diversity to struggle with the fading without necessarily sacrificing bandwidth resources, so the excellent solutions of removing the fading of the channel for broadband wireless communications is using space diversity. Then, with the constellation rotation of the symbol, rotated version of Quasi-Orthogonal Space-Time Block Code is generated. It provides full diversity. We simulate BER for QOSTBC, rotated QOSTBC, orthogonal STBC and for uncoded system. The simulation result shows that QOSTBC and rotated QOSTBC perform better than other systems. It shows that QOSTBC provides a full transmission rate but that rotated QOSTBC provides the full rate with the full diversity.

The error-correcting performance of low-density parity check (LDPC) codes, when decoded using practical iterative decoding algorithms, is known to be close to Shannon limits. In this paper we study the LDPC codes performance when varying code rate, constellation level and the maximum number of iteration, this paper shows better coding gain can be obtained at the cost of higher complexity or higher bit rate. For some cases, due to their inability to reach very low bit error rates (e.g., 10− 12) at low signal-to-noise ratios (SNRs), a consequence the error rate floor phenomenon associated with iterative LDPC decoders is produced. This paper demonstrates that the concatenation system used LDPC as an inner code and the Bose, Chaudhuri, and Hocquenghem codes (BCH) as an outer code can successfully lower the floor. Connecting BCH codes lower the LDPC BER floor by a factor(from 32 to 22) at about SNR (from-1dB to 1.8 dB) with the same overall cod rate(i.e without any reduction in the bandwith efficiency) and restricted few number of iteration (only 5 iterations).

Non-orthogonal multiple access (NOMA) is a promising multiple access technique for beyond fifth generation (B5G) cellular wireless networks, where several users can be served on a single time-frequency resource block, using the concepts of superposition coding at the transmitter and selfinterference cancellation (SIC) at the receiver. For terrestrial networks, the achievable performance gains of NOMA over traditional orthogonal multiple access (OMA) are well-known. However, the achievable performance of NOMA in aerial networks, compared to terrestrial networks, is not well-understood. In this paper, we provide a unified analytic framework to characterize the outage probabilities of users considering various network settings, such as i) uplink and downlink NOMA and OMA in aerial networks, and ii) uplink and downlink NOMA and OMA in terrestrial networks. In particular, we derive closed-form rate outage probability expressions for two users, considering line-of-sight (LOS) Rician fading channels. Numerical results validate the derived analytical expressions and demonstrate the difference of outage probabilities of users with OMA and NOMA transmissions. Numerical results unveil that the optimal UAV height increases with the increase in Rice-K factor, which implies strong line-of-sight (LOS) conditions. Index Terms-Non-orthogonal multiple access (NOMA), Unmanned aerial vehicle (UAV), terrestrial/aerial networks, SINR, outage probability.

This paper presents about error minimization in OFDM system. In conventional system, usually using channel coding such as BCH Code or Convolutional Code. But, performance BCH Code or Convolutional Code is not good in implementation of OFDM System. Error bits of OFDM system without channel coding is 5.77%. Then, we used convolutional code with code rate 1/2, it can reduce error bitsonly up to 3.85%. So, we proposed OFDM system with Modified Convolutional Code. In this implementation, we used Software Define Radio (SDR), namely Universal Software Radio Peripheral (USRP) NI 2920 as the transmitter and receiver. The result of OFDM system using Modified Convolutional Code with code rate is able recover all character received so can decrease until 0% error bit. Increasing performance of Modified Convolutional Code is about 1 dB in BER of 10-4 from BCH Code and Convolutional Code. So, performance of Modified Convolutional better than BCH Code or Convolutional Code.Keywords: OFDM, BCH Code,...

For 100 Gb/s optical transport network, researchers are searching a suitable error correction coding scheme that can provide coding gain more than 10 dB at the bit error rate less than 10 −12 , provided that the redundancy does not exceed 20 percent. This paper presents a least complex error correction coding scheme based on iterative threshold decoding called multistage threshold decoding with difference register (MTD) for the 100 Gb/s optical transport network. High rate (code rate 0.8) self-orthogonal convolutional codes are considered. The MTD achieves lower bound error performance of maximum likelihood decoding at higher bit energy to noise density ratio. The codes with orthogonal checking 10 and larger satisfy the requirements of 100 Gb/s optical transport network. The bit error rate of MTD with parity check decoding becomes less than 1/100 times in the error floor region compared to ordinary MTD. The MTD based decoding with parity check decoding for the code with orthogonal checking 12 produces 10.60 dB coding gain at the bit error rate 10 −15. The coding gain further improves 0.25 dB in the waterfall region by using 2-step decoding.

One of major drawbacks of orthogonal frequency division multiplexing (OFDM) systems is the high peak-to-average power ratio (PAPR). A signal with high PAPR leads to nonlinear distortion caused mainly by power amplifiers in wireless transmitters. Partial transmit sequence (PTS) is one of the most attractive methods to reduce the PAPR in OFDM systems. It achieves considerable PAPR reduction without distortion, but it requires an exhaustive search over all the combinations of the given phase factors, which results in a computational complexity that increases exponentially with the number of partitions. For this optimization problem, we propose in this paper a suboptimal PTS method based on the self-adaptive multipopulation differential evolution algorithm (SAMDE). The self adaptation of control parameters and structured population, is able to obtain high quality solutions with low computational cost by evolving each sub-population of individuals over successive generations.

In this letter, we first introduce a stronger notion of the optimistic achievable coding rate and discuss a coding theorem. Next, we give a necessary and sufficient condition under which the coding rates of all the optimal FF codes asymptotically converge to a constant.

This paper deals with a design problem of an observerbased robust stabilizing controller for a class of polytopic uncertain systems. The proposed controller synthesis differs from the conventional quadratic stabilization based on Lyapunov criterion and is based on the computation of the system's trajectory. In this paper, we show a LMI-based design method of the observer-based robust controller. The effectiveness of the proposed controller design approach is presented through a simple numerical example.