As a kind of iterative decoding algorithms based on belief propagation (BP), sum-product algorithm (SPA) is capacity-approaching but scale-variant. Currently, however, excellent performance and great stability are still incompatible for min-sum algorithm (MSA) and its improvements. To remedy above problems, we firstly introduce the concept of rescaling, and disclose the causes of the performance loss of MSA. Then introduce the asymptotic optimal decoding algorithm, as well as the suboptimal one. Finally, a more practicable algorithm is proposed, called check-ratio-aided layer-based rescaling adaptive normalized MSA (CL-RANMSA). Compared with the first two algorithms, CL-RANMSA is both capacity-approaching and characteristically robust. In particular, the practicably optimal factors for base graph 2 in 5G new radio low-density parity check (5G-NR LDPC) codes are provided. Simulations are also carried out whose results depict its scale-invariance and excellent performance.

Low‐density spreading multiple access (LDSMA) was proposed for supporting massive connectivity and low latency scenarios in 5th generation systems owing to its low implementation complexity. In this study, the authors analyse the performance loss of conventional LDSMA systems from two aspects: the one from low‐density spreading constellations, and the other from the mismatch between multi‐user detector and channel decoders. For the first aspect, they propose a new multi‐dimensional constellation designed via simple symbol remapping with the same constellation as in LDSMA in each dimension. And for the second aspect, newly designed Raptor‐like low‐density parity‐check codes are proposed to help retrieve the performance loss. Simulation results show that the performance of the proposed enhanced system is 1.4 dB and above superior to the conventional LDSMA system and less than 1.2 dB away from Shannon limit of Gaussian multiple access channel with symmetric users at sum rate of 1.5 and 2.25 bits per channel use.

To provide massive connectivity of a large number of low-power devices in massive machine-type communications (mMTC), advanced physical layer multiple access technologies are imperative for next generation of cellular networks. In this paper, a physical layer non-orthogonal multiple access (NOMA) scheme, called resource-pattern-aided bit-interleave coded multiple access, BICMA for short, is proposed for mMTC to support high and flexible user load energy-efficiently with low complexity. The rate-compatible length-scalable low-density parity check (LDPC) codes, recommended for 5G new radio in 3GPP work item, are firstly incorporated into NOMA schemes to provide great flexibility and compatibility with other services, like enhanced mobile broadband. In BICMA, instead of spreading, the constellation symbols of users are directly mapped to chips according to resource patterns, of which the density can be adjusted to reduce multi-user interference at each chip. Thanks to the rate-compatible length-scalable LDPC codes and flexible resource pattern, the parameter optimization of BICMA is of low complexity. Simulation results show that the energy-efficiency of BICMA is higher than that of orthogonal multiple access schemes, especially with relatively high sum spectral efficiency. Various user loads, as high as 18 users with spectrum efficiency of 0.25 bits/chip/user, can be supported by BICMA, and the required signal-to-noise ratio of each user only increases slightly while much more users could access the same channel resources simultaneously.

Enhanced mobile broadband (eMBB) supports not only sustained high-data-rate transmission but also contention based grant-free transmission. For efficient radio resource utilization, interleave-division multiple access (IDMA) which is a promising instantiation of non-orthogonal multiple access is adopted for grant-free transmission in this paper. Conventional IDMA schemes mainly focus on low rate channel code which results in limited total spectral efficiency and limited choices of code rates. Moreover, IDMA schemes suffer from performance degradation under severe multi-user interference, since conventional coding scheme in IDMA is mainly optimized for point-to-point transmission. In this work, we propose a rate-compatible LDPC based IDMA solution to support grant-free transmission in eMBB. The raptor-like quasi-cyclic low-density parity-check (RL-QC-LDPC) codes contained in this solution are similar to the channel codes for 5G eMBB data channel in code rate and length, and optimized towards severe multi-user interference with the aid of multi-edge type density evolution based extrinsic information transfer chart tool. According to block error rate simulation results, the proposed rate-compatible LDPC based IDMA scheme is able to provide high throughput with near capacity performance and be robust to flexible number of access users.

Sparse code multiple access (SCMA) is a newly proposed non-orthogonal multiple access technique for the 5th generation (5G) of wireless communication systems. It improves low-density signature multiple access (LDSMA) by replacing the inherent repetition code with multi-dimensional constellations. In this paper, the Gaussian-input channel capacities as well as the constellation constrained (CC) capacities of SCMA and LDSMA are analyzed and compared for the cases of Gaussian multiple access channel (GMAC) with random phase rotation and GMAC with fast fading, which disclose the performance potential of SCMA over LDSMA.

In this paper, we analyze asymptotic outage behavior of the uplink non-orthogonal multiple access (NOMA) over blockfading Gaussian multiple access channel (BF-GMAC) without CSIT when the number of active users tends to infinity. A closedform lower bound of the asymptotic outage probability is derived, and the accurate outage probability is numerically calculated and discussed for BF-GMAC with Nakagami-m fading. The numeral results disclose threshold effects for NOMA over such fading channels.

Uplink grant-free non-orthogonal multiple access (NOMA) is promising for machine-to-machine (M2M) communications to meet the requirement of sporadic short packets transmission of massive devices. In this paper, successive interference cancellation (SIC) and joint decoding (JD), as two well-known decoding techniques for NOMA, are analyzed for M2M random access. Due to the sporadic small packet transmission, a block-fading multiple access channel (BF-MAC) is considered, and the frequency-selectivity is also introduced. The achievable individual outage rate regions of SIC and JD, with deterministic active users, are first obtained based on individual outage events, then the analysis is extended to random access and the achievable individual outage rates and throughputs are derived. Numerical results show that SIC can perform as well as JD with relative low user load and transmission powers over frequency-flat BF-MAC. But with higher transmission powers or frequency-selectivity, the outage-limited maximum throughputs of JD are significantly higher than that of SIC.

In this paper, we investigate the impact of channel coding on two kinds of non-orthogonal multiple access (NOMA) schemes, either with a successive interference cancellation (SIC) receiver or with a joint decoding (JD) receiver. Firstly, the information-theoretic individual achievable signal-to-noise ratio (SNR) regions of SIC and JD are analyzed. Then single-user-oriented (SU) code and multi-users-oriented (MU) code are considered for SIC and JD, and the practical individual achievable SNR regions of SIC and JD with these two kinds of channel codes are obtained, which provides an intuitive way to evaluate system performance under different channel conditions and transmission rates. Simulation results show that, the practical boundaries of individual achievable SNR region of SIC generally lie inside the boundaries of JD, as predicted by the theoretical analysis. Moreover, SU code, which is capacity-approaching over point-to-point channel, performs better than MU code for SIC, while the choice of channel codes for JD depends on transmission rates and channel conditions.

Single frequency network (SFN) is a promising technology for broadcasting systems with the advantages of better coverage and higher spectral efficiency. However, transmitting local services with unequal contents in SFN remains to be a problem. Bit division multiplexing (BDM) enables local service insertion (LSI) by dividing the channel resources into two sub-channels at bit level, which can approach the theoretical bound of total capacity than traditional time division multiplexing (TDM) method. With BDM, the transmission of global services enjoys all SFN advantages, at the same time each local content uses the same frequency layer. The coverage ratio of global service area (GSA) and local service areas (LSAs) are limited due to interference and reception thresholds. This paper deals with the option of BDM strategy and the adaption of transmitter parameters to optimize the overall coverage performance. A metaheuristic optimization approach using genetic algorithm (GA) is proposed. The simulation and discussion are based on a realistic scenario.

Non-orthogonal multiple access (NOMA) scheme is a key technique towards the 5th generation (5G) of wireless communication networks due to its higher spectral efficiency comparing to its orthogonal multiple access (OMA) counterpart. However, the complexity of conventional NOMA scheme is extremely high in massive-access scenes. In this paper, we propose a user-pairing based suboptimal scheme for uplink multiple-access by combining NOMA and OMA to facilitate the massive-access scenes. Furthermore, a user-pairing algorithm with O(nlogn) complexity is also derived. According to the Monte Carlo simulation, the proposed scheme shows only a little throughput loss comparing to the optimal transmission scheme in information-theoretic perspective.

Bit-interleaved coded modulation (BICM) systems with high order constellations and signal space diversity (SSD) are deployed in DVB-T2 and recommended for potential use in 3GPP-LTE standards for its high spectral efficiency and diversity gain. However, in such systems, previous soft demapping algorithms show either extremely high computational complexity or poor performance under imbalanced In-phase/Quadrature (IQ) fading channel due to SSD. In this paper, a simplified universal soft demapping algorithm for Gray-mapped Mary constellation with O(logM) computational complexity is proposed. According to the average mutual information analysis and bit error rate (BER) simulation, the proposed algorithm shows almost the same performance as the Max-Log-MAP algorithm under conventional channel and a little SNR loss under imbalanced IQ fading channel.

In 5th generation (5G) wireless communication networks, non-orthogonal multiple access (NOMA) is a key technique for supporting massive connectivity and high spectral efficiency. In NOMA systems, the signals sent by multiple users are superimposed at the receiver, which is a multiple access channel (MAC) in information-theoretic perspective. In this paper, we propose near-capacity multi-user bit-interleaved coded modulation with iterative demapping (MU-BICM-ID) scheme for two-user Gaussian multiple-access channel (GMAC), especially for high spectral efficiency scenarios. To retrieve the shaping loss, we suggest to use amplitude phase shift keying (APSK) constellation instead of conventional quadrature amplitude modulation (QAM) constellation. Furthermore, we disclose the fact that the low-density parity-check (LDPC) code optimized for point-to-point is not suitable for GMAC, and therefore we propose and optimize Raptor-like LDPC code for GMAC. Via extrinsic information transfer (EXIT) chart analysis as well as bit-error rate (BER) simulation, we illustrate the near-capacity performance of the proposed scheme.

Scalable video transmission (SVT) system, which is able to transmit video programs with multiple qualities to different users with different channel and reception conditions, is well designed and implemented on FPGA in this paper. The designed SVT system is based on bit division multiplexing (BDM) technique and H.264/SVC. Capacity region analysis, bit error ratio (BER) simulations and field trial are performed in this paper. The results show that the proposed SVT system has higher spectrum efficiency than the traditional SVT system with redundant source bit streams and time division multiplexing technique. Field trial results show that, with the proposed SVT system, video services with different qualities have different coverage areas.

This paper proposes an algorithm for the decoding of low-density parity check (LDPC) codes, aiming to retrieve the performance degradation of conventional normalized min-sum algorithm, especially for irregular LDPC code. In our proposed algorithm, normalization factors are adjusted dynamically and adaptively during iterative decoding process according to look-up tables obtained by training and simulation in our proposal. The look-up tables construct the relationship between normalization factor a and the failed parity-check equation proportion p, which could be calculated in each decoding process. Simulation results demonstrate that the proposed algorithm can achieve a performance closer to the sum-product algorithm and a coding gain of around 0.2dB compared to the conventional normalized min-sum algorithm for rate 2/3 irregular LDPC codes over the AWGN channel. Our proposal is independent from the channel characteristics and retains low complexity. The implementation facility and the excellent performance make the proposed algorithm expect a wide application.

Digital television terrestrial multimedia broadcasting (DTMB) is the digital television terrestrial broadcasting (DTTB) standard apllied in the People's Republic of China. The performance degradation of each transmission step in DTMB system is analyzed thoroughly in this paper. The overall degradation includes two main parts: the time-frequency bandwidth loss and imperfect coded modulation scheme. This analysis provides the guideline for the potential improvement in each step. Aiming to retrieve degradations in each process, we propose some alternative and effective proposals which could be adopted in future DTTB system.

A multi-user access scheme is proposed by combining bit-interleaved coded modulation with iterative demapping and decoding (BICM-ID) and multi-user detection (MUD), which is called MU-BICM-ID in this paper. In the proposed scheme, amplitude-phase shift keying (APSK) constellation and serial-concatenated convolutional code (SCCC) are adopted as coded modulation scheme. Unlike traditional orthogonal multiple access (OMA) or successive interference cancellation (SIC) based superposition coding, as a practical implementation of simultaneous decoding, MU-BICM-ID can approach any point on the upper boundary of the capacity region for multiple access channel (MAC) at relative high spectrum efficiency. The extrinsic information transfer (EXIT) analysis results show that, for the case of two users both with 1/2-rate and 16-APSK over Gaussian MAC (GMAC), the reception signal-to-noise ratio (SNR) thresholds of two users are close to the theoretical limit, which is also verified by BER simulation results.

Multiple-input multiple-output (MIMO) is one of the key technologies for the next generation digital terrestrial television broadcasting (DTTB). This paper proposes a near-capacity MIMO bit-interleaved coded modulation (BICM) scheme for DTTB systems. The amplitude phase shift keying (APSK) constellation with Gray mapping is applied to the BICM scheme. From the perspective of average mutual information, APSK has the advantage of shaping gain compared to its quadrature amplitude modulation (QAM) counterpart. Moreover, a new kind of bit interleaver named bit mapping, which takes into account the unequal error protection of both APSK and low-density parity-check (LDPC) codes, is proposed to further improve the system performance. The tool of extrinsic information transfer chart is utilized to help analyze and design the optimal bit mapping. Bit error rate simulations are carried out to verify the superiority of the proposed MIMO system.

Two-user Single-Input Single-Output degraded Gaussian broadcast channel (GBC) with finite input constellations is addressed in this paper. Multiple downlink multiple access schemes including orthogonal multiple access (OMA) schemes and non-orthogonal multiple access (NOMA) schemes are compared from several aspects. For OMA schemes, power control is not considered and constant power for each symbol is assumed. For NOMA schemes, two typical schemes are considered which are traditional linear superposition coding (SC) and recently proposed bit division multiplexing (BDM). BDM is essentially an instantiation of non-linear SC. Capacity region analysis and simulation results are performed to give an insight into the downlink multiple access schemes. The results show that BDM can increase the system throughput compared with OMA schemes and has lower implementation complexity as well as better practical performance compared with traditional linear SC. Given typical transmission rate requirements of two users, the gaps to the union theoretical SNR thresholds for two users by BDM are both less than 2.5dB with independent demapping employed at receiver, which are far better than those achieved by OMA schemes and traditional linear SC.

Digital video broadcasting-next generation handheld (DVB-NGH) is the first broadcasting standard that incorporates multiple-input multiple-output (MIMO) techniques to overcome the Shannon limit of single antenna systems. This paper contributes to analyze the performance of the DVB-NGH MIMO coded modulation system. The detailed performance degradation from information-theoretic limit to implementation by average mutual information and extrinsic information transfer analysis is presented, which provides an insight guideline for future system improvement. Finally, bit error rate simulations are carried out to verify the analysis.

Simultaneous broadcasting of multiple services in multiple-input and multiple-output (MIMO) broadcasting system is highly desirable. In this paper, the recently proposed technique, bit division multiplexing (BDM), is applied to MIMO broadcasting system, which significantly improves the spectral efficiency of multi-service broadcasting compared with conventional time division multiplexing (TDM) approach. Both average mutual information (AMI) analysis and bit error rate (BER) simulation results demonstrate that BDM can achieve higher overall transmission rate or lower minimum SNR requirement of multiple services. Furthermore BDM in MIMO system may decrease the complexity of MIMO detector accordingly.

Doping technique was employed to remove the high error floor in conventional bit-interleaved coded modulation with iterative demapping and decoding (BICM-ID). This paper proposes a new look of the doping technique-BICM-ID with serial concatenated convolutional code (SCCC-BICM-ID). Unlike conventional doping decoding scheme which deems the doping decoder and the demapper as a whole doped demapper, we combine the doping code and the outer convolutional code together to form an SCCC, so that iterative demapping and decoding is performed between the demapper and the SCCC decoder. Extrinsic information transfer (EXIT) chart is used to analyze these two decoding schemes and reveals that they both can achieve near-capacity performance with no penalty of error floor. According to the results of EXIT analysis and BER simulations, SCCC-BICM-ID system can achieve the same or even better performance compared to the doping BICM-ID system with lower decoding complexity.

This paper proposes an adaptive normalized min-sum algorithm for the decoding of low-density parity check (LDPC) codes, which utilizes an adaptive normalization factor to improve the accuracy of the soft information transferred during the iterative decoding process, and provides superior performance accordingly. This adaptive normalization factor can be adjusted dynamically and adaptively at each decoding iteration according to a look-up table obtained via training and simulation. Its implementation facility and independence from the channel characteristics make the proposed adaptive normalized min-sum algorithm expect a wide application. Simulation results show that the proposed algorithm can achieve performance much closer to the sum-product algorithm and a coding gain of around 0.2dB compared to the conventional normalized min-sum algorithm for DVB-S2's rate 2/5 and 3/5 LDPC codes over the additive white Gaussian noise (AWGN) channel.

This paper proposes a "column extension" design method for rate-compatible quasi-cyclic (QC) low-density parity-check (LDPC) codes, which can provide efficient multi-rate coding schemes for communication systems with flexible spectrum efficiency. The optimization of degree distributions among multiple code rates makes the designed QC-LDPC codes achieve excellent performance at each code rate, while the nested parity-check matrix structure of the designed codes facilitates the implementation of multi-rate encoder/decoder and offers the advantage of low-complexity. A comparison with the multi-rate LDPC codes in DVB-S2 specification is addressed. Extrinsic information transfer (EXIT) chart analysis predicts the superiorities of the optimized degree distributions in Eb/N0thresholds, and the bit error rate (BER) simulation results show that the designed codes perform 0.03-0.1 dB better than the DVB-S2 64K codes at each typical code rate, with even shorter code length.

Due to the unequal error protection (UEP) property of both the irregular low-density parity-check (LDPC) code and the higher-order modulation, bit mapping is proposed as a joint optimization of the LDPC coded-modulation scheme. In this paper, we discuss the bit mapping optimization performed by employing an additional DEMUX following the given bit interleaver, and then apply it to the second generation digital video broadcasting system for cable (DVB-C2). The extrinsic information transfer (EXIT) chart technique is adopted to analyze the influence of bit mapping on the iterative decoding process. Based on the EXIT-chart analysis, an optimization method is proposed to design the DEMUX for performance improvement. As an example, we re-design the DEMUX for the DVB-C2 system with reduced DEMUX width. The superior performance of the optimized DEMUX is verified by the bit error rate (BER) simulation.

3GPP long term evolution (LTE) Turbo coded modulation is well-known for its excellent error-control capability. However, its decoding threshold is still a few decibels away from the constellation-constrained capacity. In this paper, an irregular mapping scheme involving labeling search is proposed to improve the performance of the LTE system. The scheme first predetermines one labeling, and then searches the other one via adaptive binary switch algorithm (ABSA), which considerably increases the design freedom and can provide near-optimal match to a given channel code. By applying the scheme to the LTE specification (Release 8), a practical bit-interleaved Turbo coded modulation with iterative demapping and decoding (Turbo-BICM-ID) system is proposed. With a slight modification on the constellation mapping, the scheme can lower the signal-to-noise ratio (SNR) threshold by 0.43~0.48dB at the block error rate (BLER) of 10-2under different channel conditions.

Signal space diversity (SSD) is a well-known power- and bandwidth-efficient diversity technique with constellation rotation and coordinate interleaving. This paper investigates issues of combining SSD with coded modulation systems (CMSs), e.g., bit-interleaved coded modulation (BICM) and its iterative version, BICM-ID. It demonstrates that the average mutual information (AMI) between the signal after rotated constellation mapping and the signal before or after the soft demapper varies with the rotation angle. A new criterion for determining the optimal rotation angle by maximizing such AMI is therefore proposed. Specific considerations of combining SSD with BICM (BICM-SSD) and BICM-ID (BICM-ID-SSD) are addressed. The demapper's extrinsic information transfer (EXIT) curve in traditional BICM-ID systems exhibits different slopes under different channels, which consequently prevents traditional BICM-ID systems from having simultaneously excellent performance under different channels. It is shown that such a different-slope problem can be simply mitigated by a proper use of SSD. Analysis and simulation show that the proposed BICM-ID-SSD systems hold a near-capacity performance under both additive white Gaussian noise (AWGN) and Rayleigh fading channels simultaneously.

Signal space diversity (SSD) is a well-known technique providing excellent performance gain over fading channels. This paper investigates the system design of bit-interleaved coded modulation with iterative demodulation and SSD (BICM-ID-SSD). It demonstrates that within SSD the average mutual information (AMI) between the signal after rotated constellation mapping and that before the soft demapper varies with the rotation angle. Therefore, a new criterion for searching the optimal rotation angle by maximizing such AMI is proposed. Based on this criterion it is shown that the optimal rotation angle is not relevant to the labeling, and 45-degree is found to be the optimal or near optimal rotation angle for square quadrature amplitude modulation (QAM) at low to moderate code rates. Furthermore, the procedure of BICM-ID-SSD system design could be divided into two independent steps: 1) determining the optimal rotation angle based on the above criterion, and 2) choosing well-fitted labeling and outer channel code with the aid of extrinsic information transfer (EXIT) charts. Analysis and simulation results show that the proposed system exhibits a near-capacity performance, meanwhile being robust over both additive white Gaussian noise (AWGN) and Rayleigh fading channels.

Through the investigation on frame header (FH) structures of DTMB system, i.e. the digital terrestrial television broadcasting standard of China, spectrum sensing methods based on PN cross-correlation (PNCC) are proposed for all FH options. Miss-detection probability, decision threshold and sensing time requirement of the proposed methods are analyzed in detail. Comparisons on spectrum sensing performance, robustness to time dispersive channel and carrier frequency offset are made between the proposed methods and the existing methods based on PN autocorrelation (PNAC). Both theoretical analyses and simulation results demonstrate the considerable advantage of the proposed PNCC methods over the existing PNAC methods.

This paper summarizes some key features for future Chinese digital television terrestrial broadcasting (DTTB) systems including improved frame structure, transmit diversity, advanced channel coding and modulation, multi-service support, broadcasting return channel solution and etc.

To improve the overall performance of time-domain synchronous (TDS) orthogonal frequency division multiplexing (OFDM) systems, novel iterative decoding and channel estimation algorithm with Nordstrom-Robinson (NR) code is proposed in this paper. Compared to the non-iterative case, the system performance can be significantly improved by the proposed iterative algorithm at the cost of a little complexity and latency increase. Simulation results show that, for example, with maximum Doppler spread of 25 Hz under two typical multipath channels at bit error rate (BER) of 10-4, the signal to noise ratio (SNR) performance of the proposed iterative algorithm is only 0.2 to 0.3 dB away from the ideal case, and about t dB or more better than the non-iterative one.

It is well known that the cyclic property of transmitted signal can be used to achieve joint frame and carrier frequency synchronization, while the uncompensated sampling frequency offset (SFO) would still be a problem. Based on the observation that the impact of SFO on the received cyclic block and its replica is equivalent to a relative time shift, a joint synchronization scheme solely based on cyclic property is proposed to achieve frame, sampling frequency and carrier frequency synchronization simultaneously. A two-dimensional sliding-window auto-correlator (2D-SWAC) is presented to implement the joint synchronization with low complexity. The proposed scheme is suitable for any transmission system with similar cyclic property. Meanwhile, it is robust to large sampling and carrier frequency offset and strong channel frequency selectivity, as demonstrated by the analysis and simulations.

Bit-interleaved coded modulation (BICM) is a sub- optimal scheme from the average mutual information (AMI) point of view due to independent demapping. However, this AMI loss can be neglected for signal constellations with Gray mapping at high coding rates. AMI of amplitude-phase shift keying (APSK) constrained additive white Gaussian noise (AWGN) channel is provided in this paper, which shows that the BICM scheme has considerable loss with APSK constellations since no Gray mapping exists. Bit-interleaved low-density parity-check (LDPC) coded modulation (BILCM) is an excellent scheme and has been employed in many broadcasting and communication systems, however, such scheme also suffers from the independent demapping loss. Therefore, BILCM with iterative demapping and decoding (BILCM-ID) is proposed in this paper to overcome such loss. Simulation results show that the BILCM-ID scheme outweighs BILCM scheme by about 0.5 and 0.3 dB over AWGN channel for 32-APSK constellation at coding rates 2/3 and 4/5, respectively.

In order to improve the system performance over wireless fading channel, transmit diversity technique has been proven as an attractive and effective solution especially for broadcasting systems. In this paper, a PN sequence design scheme based on both cyclic-shift m-sequences and space-time block coded (STBC) m-sequences for time domain synchronous orthogonal frequency-division multiplexing (TDS-OFDM) system with transmit diversity is proposed. Based on this novel design, flexible channel estimation which provides a trade-off between maximum delay spread and maximum Doppler spread over double-selective channel is investigated and a radix-2 fast Fourier transform (FFT) approach is adopted for implementation from the viewpoint of computational complexity. Simulations show that the performance of the proposed transmit diversity scheme is better than conventional one and perform well under different propagation conditions.

In order to improve the system performance over both frequency- and time-selective fading channels, orthogonal frequency division multiplexing (OFDM) system with transmit diversity technique is an effective and attractive solution. However, the received signal is the superposition of the signals from different transmit antennas. Hence, to obtain accurate channel state information (CSI) is difficult but very important to recover transmitted OFDM symbols. Conventional channel estimation methods use subcarrier pilots or training sequences as the channel sounding signal to obtain the estimation of channel frequency response (CFR). Actually, decision-directed channel estimation (DDCE) method which takes advantage of the fact that m-QAM constellation subcarriers in OFDM symbol also contain much valuable hidden-information and can be used to perform the channel estimation as well as tracking of time-varying channel. In this paper, a simple and effective partial hard decision method for m-QAM is investigated. Furthermore, a simplified DDCE method based on an accurate initial CFR and partial decision results of OFDM subcarriers for transmit diversity system is proposed. Simulation results demonstrate that the proposed algorithm works well over the slow-varying frequency-selective fading channel.

Pseudo-noise (PN) sequence is widely used in wireless transmission systems for synchronization, which is usually performed via cross-correlation, i.e. PN correlator, between the received signal and a locally generated PN sequence. However, the hardware implementation of such a correlator becomes the inhibiting bottleneck of integration due to the long tapped-delay line, particularly in the case of oversampling systems. Furthermore, few attempts have been made to improve the structure until now attributing to the random but simple coefficients of the PN correlator. This paper proposes a novel grouping and segmentation structure for filters with simple coefficients, such as a PN correlator. Remarkable hardware reduction has been achieved by multiplexing tapped-delay line and further operations skillfully. The arithmetic complexities of the conventional direct form structure and the proposed structure are compared based on a digital terrestrial multimedia broadcasting (DTMB) system. The advantages of the proposed structure are demonstrated by the FPGA implementation results.

Accurate channel estimation is essential for coherent signal transmission. This paper presents a novel channel estimation algorithm for time-domain synchronous orthogonal frequency division multiplexing (TDS-OFDM) systems with identical pseudo-noise (PN) sequence padding. Simple cyclic reconstruction and tail cancellation methods are proposed to estimate the channel impulse response (CIR). Phase equalization and partial hard decision on the received data symbols are used to improve the estimation performance. Simulation results show that the proposed channel estimation algorithm brings good BER performance (approaching the ideal case) with fast convergence (no iteration required) over frequency-selective slowly fading channels.

This study proposes an iterative channel impulse response (CIR) estimation scheme for multicarrier block transmission with known symbol padding, in the context of the time-domain synchronous OFDM (TDS-OFDM) system. It is shown that the TDS-OFDM still fulfils the circular convolution relationship even if pseudo-noise (PN) sequence is used as guard interval. While conventional methods perform the channel estimation based on the cyclic prefix reconstruction or residual intersymbol interference cancellation in the time domain, by processing the data part and the known symbol padding in one TDS-OFDM block separately, the new proposal uses the one- tap frequency-domain equalization to separate the data part and the known symbol padding and the CIR can be estimated in an iterative manner. In a typical example for 3780-point FFT based TDS-OFDM system with 16QAM constellations, the mean square error (MSE) of of the proposed channel estimator can be reduced to about 10-5/2 after two iterations when the channel signal-to-noise ratio (SNR) is larger than 8 dB, without the initial CIR estimation. The BER performance of a QPSK TDS-OFDM system shows that the signal-to-noise-ratio (SNR) degradation based on the proposed channel estimator is less than 0.3 dB compared with the lower bound for both Brazil A and B channels.

As one of the most common linearization techniques, adaptive digital predistortion minimizes the difference between the input signal of the predistorter and the feedback signal from the power amplifier. The loop delay between these two signals must be controlled precisely so that the convergence of the predistorter should be assured. In this paper a novel two-stage loop delay correction scheme is presented. In the first stage, the integer delay between the input signal and the feedback signal is estimated based on the cross-correlation of them. In the second stage, the fractional delay between these two signals are corrected by a delay lock loop (DLL) circuit, in which a novel timing error detector used as the discriminator is proposed. The proposed loop delay correction scheme is applied to an adaptive digital predistorter in an orthogonal frequency-division multiplexing (OFDM) system. Simulation results show that the proposed loop delay correction method eliminates the loop delay perfectly, so that nearly optimal power spectral density (PSD) performance is achieved and a high degree of linearization is realized.

The method of modeling and ordering in wavelet domain is very important to design a successful algorithm of embedded image compression. In this paper, the modeling is limited to "pixel classification," the relationship between wavelet pixels in significance coding. Similarly, the ordering is limited to "pixel sorting," the coding order of wavelet pixels. We use pixel classification and sorting to provide a better understanding of previous works. The image pixels in wavelet domain are classified and sorted, either explicitly or implicitly, for embedded image compression. A new embedded image code is proposed based on a novel pixel classification and sorting (PCAS) scheme in wavelet domain. In PCAS, pixels to be coded are classified into several quantized contexts based on a large context template and sorted based on their estimated significance probabilities. The purpose of pixel classification is to exploit the intraband correlation in wavelet domain. Pixel sorting employs several fractional bit-plane coding passes to improve the rate-distortion performance. The proposed pixel classification and sorting technique is simple, yet effective, producing an embedded image code with excellent compression performance. In addition, our algorithm is able to provide either spatial or quality scalability with flexible complexity.

We consider the problem of embedded image transmission with feedback of channel state information (CSI) and retransmission request. Adaptive modulation is employed to improve the spectral efficiency. Constrained retransmission is employed to provide reliable packet transmission with limited delay. Performance analysis and numerical results over Nakagami block fading channels are provided for the transmission system with statistical or instantaneous CSI at the transmitter. The results show that CSI at the transmitter is very important for adaptive modulation/coding over fading channels.

A new embedded image compression algorithm is proposed, based on progressive Pixel Classification And Sorting (PCAS) in wavelet domain. To exploit the intraband and interband correlation in wavelet domain, EZW [1] and SPIHT [2] implicitly classify wavelet pixels as zerotree pixels or not, while MRWD[3], SLCCA[4], and EBCOT[5] implicitly classify wavelet pixels as neighbors of significant pixels or not. In this paper, the wavelet pixels to be encoded are explicitly and finely classified based on their predicted probabilities, which is more sophisticated and effective. Furthermore, wavelet pixel sorting is introduced to help improve rate-distortion performance within each bit-plane coding. The technique of pixel classification and sorting is simple, yet effective to produce the image code with excellent compression performance. In addition, our algorithm provides both SNR and resolution scalability.