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STiMi is a mobile multimedia broadcasting trade standard that was released by the Chinese State Administration of Radio Film and Television in 2006. This paper details the algorithm research and development work on the acquisition for STiMi. First, the STiMi physical layer is briefly reviewed. secondly, individual algorithms, including time slot detection, fractional frequency offset estimation, integer frequency offset estimation and fine timing synchronization are discussed with simulation results presented. An implemented STiMi receiver prototype is also introduced with a focus on the acquisition algorithms implementation. Finally, the whole acquisition procedure is summarized. Acquisition algorithms and structures presented in this paper are adopted for prototyping and perform well. [ABSTRACT FROM AUTHOR]

This paper describes a channel estimation algorithm for the advanced terrestrial digital multimedia broadcasting (T-DMB) transmission system which is fully backward compatible with the conventional T-DMB standard. The advanced T-DMB has enhanced features which provide higher transmission data rate for increasing program channel capacity and better multimedia A/V service quality than the conventional T-DMB system by adopting a hierarchical modulation and scalable audio/video CODEC. To support the higher data rate transmission for the advanced T-DMB, high performance channel estimation and equalization techniques should be adopted into the advanced T-DMB receivers. In this paper, we propose a new channel estimation technique for the Advanced T-DMB to compensate channel distortion under dynamic multipath environment. [ABSTRACT FROM AUTHOR]

When adopting transmit diversity in the Chinese digital television terrestrial broadcasting (DTTB) system, there exists two kinds of interference: training sequences (TSs) from different transmit antennas are overlapped; and the TSs are interfered with OFDM data due to multipath fading. In order to distinguish different TSs, this paper proposes a method to design orthogonal TS set, which also have good correlation characteristic. Subsequently, an iterative algorithm to eliminate the interference between TSs and data is proposed and theoretically analyzed. Finally, simulations are performed and the results show that the system with the proposed scheme can achieve great diversity gain under acceptable computational complexity. [ABSTRACT FROM PUBLISHER]

Single carrier frequency division multiple access (SC-FDMA) is one well-known scheme, which has recently become a preferred choice for uplink channels. Due to the usage of single carrier, the performance of SC-FDMA systems degrades in deep frequency selective fading channels. In this paper, we propose a structure of equalizer based on frequency domain decision feedback which could be used for multi-user SC-FDMA systems. Specific parameters of the equalizer are analyzed as well. This algorithm is applicable to various carrier allocations in multi-user systems such as localized allocation, distributed allocation, and frequency-hopping (FH) allocation. To reduce the complexity, it is not necessary to derive the inversion of matrix, which is required in the traditional decision feedback equalizer for single carrier frequency domain equalization (SC-FDE-DFE). It is shown by the simulation results that perfect performance could be achieved under all kinds of situations for multi-user systems. This structure can be used in the broadcasting uplink channels with SC-FDMA scheme. [ABSTRACT FROM AUTHOR]

LDPC codes are state-of-art error correcting codes, included in several standards for broadcast transmissions. Iterative soft-decision decoding algorithms for LDPC codes reach excellent error correction capability; their performance, however, is strongly affected by finite-precision issues in the representation of inner variables. Great attention has been paid, in recent literature, to the topic of quantization for LDPC decoders, but mostly focusing on binary modulations and analysing finite precision effects in a disaggregrated manner, i.e., considering separately each block of the receiver. Modern telecommunication standards, instead, often adopt high order modulation schemes, e.g. M-QAM, with the aim to achieve large spectral efficiency. This puts additional quantization problems, that have been poorly debated in previous literature. This paper discusses the choice of suitable quantization characteristics for both the decoder messages and the received samples in LDPC-coded systems using M-QAM schemes. The analysis involves also the demapper block, that provides initial likelihood values for the decoder, by relating its quantization strategy with that of the decoder. A new demapper version, based on approximate expressions, is also presented, that introduces a slight deviation from the ideal case but yields a low complexity hardware implementation. [ABSTRACT FROM AUTHOR]

The frequency assignment problem involves the assignment of discrete frequencies to the transmitters of a radio network, such as a radio broadcasting network. Frequency separation is necessary to avoid interference by other transmitters to the signal received from the wanted transmitter at the reception region. Here, it is of major importance to minimize the interference while at the same time using the spectrum efficiently. In this paper we present two original distributed algorithms implemented on clusters of PCs used to solve the frequency assignment problem in the field of radio broadcasting. The first one is based on the island distributed [1] implementation of our hybrid genetic algorithm [2]. The second one uses a distributed cooperative Tabu Search. Experimental results show that our algorithms, applied to several instances given by TDF-C2R, lead to important time performance improvements. [ABSTRACT FROM AUTHOR]

In this paper we present a Base-matrix based decoder architecture for multi-rate QC-LDPC codes proposed in broadband broadcasting system. We use the Modified Min-Sum Algorithm (MMSA) as the decoding algorithm in this architecture, which lowers the complexity of the LDPC decoder while keeping almost the same performance or even better. Based on this algorithm, we designed a novel check node processing unit to reduce the complexity of the decoder and facilitate the multiplex of the processing units. The decoder designed with hardware constraints is not only scalable in throughput, but also easily configurable to support different QC-LDPC codes flexible in code rate and code length. [ABSTRACT FROM AUTHOR]

This paper presents a simple but effective IPTV channel control algorithm that keeps channel zapping time in the tolerable range with high network utilization. The proposed algorithm controls channel zapping time by adjusting the number of broadcasting channels that are located close to users over IP networks and the number of additional I-frames inserted into each channel, based on the user's channel preference information. Finally, experimental results are provided to show the performance of the proposed algorithm. [ABSTRACT FROM AUTHOR]

This paper introduces a layered structure of the Korean terrestrial digital multimedia broadcasting (T-DMB) ensemble based on Eureka-147 and suggests a new transmission architecture for flexible T-DMB service. In the suggested transmission architecture, MPEG-2 transport stream (TS) for T-DMB multimedia service is inserted into the ensemble distributed via the ensemble transport interface (ETI). Thus, we propose a novel algorithm remultiplexing the ETI with MPEG-2 TS. The architecture of ETI remultiplexing algorithm is described as a four-step process, and implemented as real-time embedded software in an ETI remultiplexing system. The design and implementation of the interoperable ETI remultiplexing system available for a small-scale transmission system for T-DMB service is described in detail. [ABSTRACT FROM AUTHOR]

It has been demonstrated that turbo codes substantially outperform other codes, e.g., convolutional codes, both in the non-fading additive white Gaussian noise (AWGN) channel as well as multiple-transmit and multiple-receive antenna fading channels. Moreover, it has also been reported that turbo codes perform very well in fast fading channels, but perform somewhat poorly on slow and block fading channels of which the broadband fixed wireless access (FWA) channel is an example. In this paper, we thoroughly compare the performance of turbo-coded and convolutional-coded broadband FWA systems both with and without antenna diversity under the condition of identical complexity for a variety of decoding algorithms. In particular, we derive mathematical expressions to characterize the complexity of turbo decoding based on state-of-the-art Log-MAP and Max-Log-MAP algorithms as well as convolutional decoding based on the Viterbi algorithm in terms of the number of equivalent addition operations. Simulation results show that turbo codes do not offer any performance advantage over convolutional codes in FWA systems without antenna diversity or FWA systems with limited antenna diversity. Indeed, turbo codes only outperform convolutional codes in FWA systems having significant antenna diversity. [ABSTRACT FROM AUTHOR]

It is well known that HPAs (High Power Amplifiers) are inherently nonlinear devices and many researches have focused on the predistortion for memoryless HPAs. However, memory effects of HPAs can no longer be ignored when communication systems have wider bandwidth. Memoryless predistortion techniques proposed previously seldom have satisfactory effectiveness for typical wideband applications such as OFDM (Orthogonal Frequency Division Multiplexing) systems. In this paper, an improved adaptive predistortion method called 2D LUT (2-dimension look-up table) with different accuracy levels is presented to linearize HPAs with memory effects. Simulation and experimental results show that 2D LUT implements excellent performance in mitigating the signal deterioration caused by memory effects, both rectifies the signal constellation distortion and suppresses the spectrum emission. Large scale matrix computation is also avoidable in these adaptive algorithms, which makes them feasible when a real-time system is necessary. [ABSTRACT FROM AUTHOR]

Transmit diversity can be applied to OFDM systems by adopting space time code. Since the received signal is the overlapped signals transmitted from different transmit antennas, channel estimation is a rather challenging task for space time coded OFDM (ST-OFDM) systems. Pilot structure can help the receiver to effectively separate the overlapped signals and perform accurate channel estimation. In this paper, we propose three different channel estimation algorithms based on specially designed comb type pilots inserted in frequency domain. One of our proposed algorithms is performed in frequency domain and the other two are performed in time domain. Such comb type pilot based algorithms can provide higher bandwidth efficiency than common significant-tap-catching algorithm using training block pilots. Numerical analyzes and computational simulation show that our proposed estimation schemes have the same good performance while the time domain methods have relatively simple structure. [ABSTRACT FROM AUTHOR]

In this paper, we proposed a novel frequency synchronization method which has larger estimation range than conventional method proposed by Tufvesson. A two steps frequency offset estimation method is performed in the new method. In the first step the received signal is correlated with local training sequence to eliminate the influence of training sequence which makes the estimation range of frequency offset is independent of period of the repeated data in the training sequence. The received signal is correlated with its delay in the second step to obtain the fine estimation of frequency offset. Precise estimation can be achieved in the proposed method than Tufvesson's method. The period of the repeated data in the training sequence determines the estimation range of the frequency offset in the conventional method, while in the new method they are irrelevant. Tens of subcarriers spacing can be estimated by the new method. The new method can be used in other cases as long as a known sequence is transmitted, which will introduce valuable flexibility in training sequence design. The validity of the algorithm is verified in AWGN channel and multipath fading channel. [ABSTRACT FROM AUTHOR]

Broadcasting has been widely used in mobile Ad hoc networks as a communication means to disseminate information to all reachable nodes. Because radio signals are likely to overlap with others in a geographical area, straightforward broadcasting by flooding becomes very costly and results in serious redundancy, contention and collision, to which we refer as the broadcast storm problem. In this paper we propose the Relative Degree Adaptive flooding Broadcast (RDAB) algorithm for Ad hoc networks to efficiently reduce the broadcast overhead in the network. Based on the current situation of the network and the degree of the nodes, RDAB calculates the relative degree of the nodes, decides which nodes need to re-transmit and which nodes only need to receive. The higher the neighbor node's relative degree, the more uncovered nodes it can cover, hence these nodes can be selected to re-transmit broadcasting packets in the networks. We analyze the reliability and the validity of the RDAB algorithm to prove that the RDAB algorithm is a valid flooding broadcast algorithm. Simulation results show that the RDAB strategy outperforms the Ordinary Flooding Broadcast Method (OBM) and the Multipoint Relaying (MPR) protocol for Ad hoc networks. [ABSTRACT FROM AUTHOR]