基于迭代结构的联合信源信道GSM语音解码

由于算法复杂度和延迟的限制,语音编码参数仍然保留了一定的冗余信息,这些冗余量表现为参数的非等概分布和时间相关性。该文在比特级上利用信源编码参数的冗余信息在软输入、软输出信道解码和信源参数估计之间进行迭代运算,并将此联合解码算法运用于GSM EFR系统。仿真结果表明此算法降低了编码参数的BER,提高了解码语音的信噪比,具有一定的可行性和有效性。     

序列图像的多码率混合编码及其性能分析

本文针对序列图像提出了一种可称为MRHC的多码率编码方法。该方法把信源和信道编码组成一个整体,利用3D小波图像能量分布的特点,在完成压缩编解码的同时完成纠错编解码。从而达到码率优化分配,提高了序列图像在噪声信道中的传输的质量。     

信源反馈条件下Raptor码的一种解码算法

在Raptor码普通译码方法的基础上,结合信源信道联合译码方法,提出一种利用信源解码器反馈信息作为输入的改进译码的算法,并对此算法进行试验仿真以及参数研究。测试结果显示,在信源信道联合编码条件下,使用这种双信息来源的译码的算法,可以使得信道解码的成功率大幅提高,残留误码率大幅降低。最后,将这种算法与JPEG2000信源编码结合测试,图像的PSNR值比普通译码方法最高提高了6dB。在信道条件恶劣的无线传输中具有重要意义。     

一种频选衰落信道下的Turbo多用户检测算法

联合MAP多用户检测与信道解码的迭代多用户检测(MUD)技术可显著提高宽带移动CDMA系统的容量和性能.在多径时变衰落的编码信道下,提出一种迭代实现干扰抑制、符号估计、信道解码的Turbo多用户检测算法.在每次迭代中,MUD自适应地实现干扰抑制并输出符号估计的软信息,软输入软输出的信道解码器使用LOG MAP方法实现信道解码并反馈符号估计的软信息作为下一次TurboMUD迭代的先验信息.仿真结果证实了该算法在频选衰落信道下经两次迭代就能逼近单用户编码CDMA系统的接收性能.     

基于LDPC码的小波图像传输码率优化

LDPC码是一种具有低复杂度,强纠错能力的信道分组编码结构,SPIHT算法是嵌入式小波图像压缩编码中性能较好的一种。文中充分考虑SPIHT算法和LDPC码的特性,提出了根据信源编码后的数据在解码重建时的重要程度进行不等纠错保护的信源信道联合编码方案。实验表明,该方案有利于压缩图像在噪声信道上的可靠传输,提高系统整体纠错性能。     

未知噪声信息多径衰落信道的软判决度量生成

信道解码中的软判决技术需要利用信道噪声方差信息来产生软判决度量,而通常情况下,这一信息是未知和时变的。本文首先简要分析了信道噪声方差对软判决度量的影响,进而提出一种新的在多径衰落信道中估计时变噪声方差的方法,并利用该方法的估计结果为信道解码器生成软判决度量。将该算法应用于中国地面数字电视广播传输系统外接收机的仿真结果表明：与已有的信道噪声方差估计算法相比,本文算法具有收敛速度快、估计结果准确的优点,生成的软判决度量逼近于多径衰落信道的最优软判决度量。     

基于MS估计和迭代结构的信源信道联合解码系统性能分析

该文对基于均方MS(Mean Square)误差最小准则估计的软输入信源解码和基于信源参数特性及迭代结构的维特比解码两种信源信道联合解码算法进行分析,分别给出系统性能提高的影响因子,同时与信源信道独立解码算法相比较,并推导了联合算法中的接收参数信噪比的提高幅度以及比特误码率的上限值,证明了联合解码算法的可行性和有效性。实验仿真结果表明了分析方法的正确性。     

一种利用信源冗余的Turbo均衡新算法

为克服无线衰落信道中严重的符号间干扰( ISI)的影响,提出了一种新的利用信源冗余的Turbo均衡算法。该算法将联合信源信道译码技术与Turbo均衡技术结合起来,在均衡、译码、信源之间建立起软信息交互的环路,有效提高了整体接收的性能。外信息传递( EXIT)图分析与计算机仿真均表明,尽管信源冗余给译码器带来的性能提升较为有限,但是将这部分信息反馈回均衡器后,在严重ISI信道,信源冗余度为70%时,整体接收的性能改善约为9.5 dB,基本达到了理想加性高斯白噪声( AWGN)信道下的误码性能。     

类卷积译码的容错解扰算法

针对自同步扰码系统的联合信源信道译码问题,本文将自同步扰码看作一种特殊的卷积编码,提出了类似卷积译码的软输入软输出(SISO)自同步去扰算法。该算法利用信源冗余更新扰码序列的外信息,在信道译码时作为先验信息进行译码,实现了自同步去扰与信道译码之间的软信息交互,充分利用了信源冗余信息,使得接收系统的性能得到了有效提升。仿真结果表明,在TPC编码条件下,当信源冗余度为70%时,联合信源信道译码的性能增益约为4.1dB。相比于单一纠错编码系统,当通信系统中存在自同步扰码时,联合信源信道译码具有更大的性能增益。      

DTMB中NR码软解码的实现

提出了一种有效的软输入软输出NR解码方法,可应用于地面数字电视广播传输标准(DTMB)中.其算法基于软输入的最大似然概率译码,使用最大相关码字和次最大相关码字产生输出软信息.经仿真验证,引入此方案的NR解码后,P8码率LDPC解码性能提高了约3.6 dB.     

波束空时分组码系统中一种基于ICA的正交检测方案

结合波束形成和空时分组编码的混合系统与传统的单波束发射以及单空时编码发射分集方案相比可以大大提高链路性能;传统的译码方案通常以接收端能够获取精确的信道状态信息为前提;但是对于某些特定的通信环境,这种前提条件通常很难满足.独立分量分析(ICA)技术可以在不进行信道估计的情况下对发射信号实现有效检测.本文针对接收端的信号结构提出了一种基于ICA的正交检测方案;并通过仿真将新方案与传统方案进行了性能比较,仿真结果表明,新方案具有较好的系统适应性和误码率特性.     

Turbo codes for image transmission-a joint channel and sourcedecoding approach

This paper studies an application of turbo codes to compressed image/video transmission and presents an approach to improving error control performance through joint channel and source decoding (JCSD). The proposed approach to JCSD includes error-free source information feedback, error-detected source information feedback, and the use of channel soft values (CSV) for source signal postprocessing. These feedback schemes are based on a modification of the extrinsic information passed between the constituent maximum a posteriori probability (MAP) decoders in a turbo decoder. The modification is made according to the source information obtained from the source signal processor. The CSVs are considered as reliability information on the hard decisions and are further used for error recovery in the reconstructed signals. Applications of this joint decoding technique to different visual source coding schemes, such as spatial vector quantization, JPEG coding, and MPEG coding, are examined. Experimental results show that up to 0.6 dB of channel SNR reduction can be achieved by the joint decoder without increasing computational cost for various channel coding rates     

Medical image transmission over a GSM cellular system

This authors report on an investigation of possible source/channel coders for medical image transmission over a GSM cellular link. Two source coders (JPEG and EZW) and two channel coders (convolutional and turbo codes) are used. Analysis of the results indicates that EZW with convolutional coding gives the best performance, but some kind of automatic repeat on request (ARQ) is also needed     

Distributed video coding based on lossy syndromes generated in hybrid pixel/transform domain

A recently proposed class of distributed source coding based video coders enables low-complexity compression and robust transmission over unreliable channels. These architectures process the video signal either in the pixel or in the transform domain generating some side information that permits a correct decoding of the coded image from a set of possible correlated sources. The approach proposed in this paper processes the video sequence both in the pixel and in the transform domain exploiting the advantages of both schemes and generating a set of lossy syndromes. The resulting video coding scheme requires a lower computational complexity at the decoder with respect to their transform-domain counterparts (like DISCOVER or PRISM) and provides a high compression gain and an increased robustness against channel losses.     

Turbo decodulation: iterative combined demodulation and source-channel decoding

We propose the combination of iterative demodulation and iterative source-channel decoding as a multiple turbo process. The receiver structures of bit-interleaved coded modulation with iterative decoding (BICM-ID), iterative source-channel decoding (ISCD), and iterative source coded modulation (ISCM) are merged to one novel turbo system, in which in two iterative loops reliability information is exchanged between the three single components, demodulator, channel decoder and (softbit) source decoder. Simulations show quality improvements compared to the different previously known systems, which use iterative processing only for two components of the receiver.     

使用交互式编解码机制的分布式视频编码研究

为了克服Slepian-Wolf编码不存在统一编码以及采用的无记忆信道模型不符合实际等问题,运用交互式编解码(IED)方式,建立了统一编码,构建了有记忆的有限状态信道(FSC)模型,同时设计了一种适合FSC的用于LDPC解码的置信传播算法。实验结果表明,实现后的IED相比非IED具有更低的误码率(BER),且与理论极限码率更加逼近,实际视频序列的测试结果显示,实现的IED分布式视频编码系统较LDPC算法有2dB左右的率失真性能增益。     

Unequally protected multistage vector quantization for time-varyingCDMA channels

We present a source-channel coding system for operation over code-division multiple-access channels with time-varying conditions. The proposed system consists of a multistage vector quantizer (MSVQ) in conjunction with unequal protection against channel errors. The receiver estimates the channel conditions and decodes as many stages of the quantizer as can be reliably decoded. The approach to system design and optimization is first derived and evaluated for a system that employs hard decoding of stage indices. The approach is then extended to the more general case of weighted decoding. Simulation results are given for transmission of Gauss-Markov sources over broadcast and slow fading channels. Consistent and substantial improvement is achieved over the standard MSVQ with equal error protection, and the gains, in terms of source signal-to-noise ratio, are in the range of 3-5 dB     

Joint Source–Channel Turbo Techniques for Discrete-Valued Sources: From Theory to Practice

The principles which have been prevailing so far for designing communication systems rely on Shannon's source and channel coding separation theorem. This theorem states that source and channel optimum performance bounds can be approached as close as desired by designing independently the source and channel coding strategies. However, this theorem holds only under asymptotic conditions, where both codes are allowed infinite length and complexity. If the design of the system is constrained in terms of delay and complexity, if the sources are not stationary, or if the channels are nonergodic, separate design and optimization of the source and channel coders can be largely suboptimal. For practical systems, joint source-channel (de)coding may reduce the end-to-end distortion. It is one of the aspects covered by the term cross-layer design, meaning a rethinking of the layer separation principle. This article focuses on recent developments of joint source-channel turbo coding and decoding techniques, which are described in the framework of normal factor graphs. The scope is restricted to lossless compression and discrete-valued sources. The presented techniques can be applied to the quantized values of a lossy source codec but the quantizer itself and its impact are not considered.     

Channel coding for satellite mobile channels

The deployment of channel coding and interleaving to enhance the bit-error performance of a satellite mobile radio channel is addressed for speech and data transmissions. Different convolutional codes (CC) using Viterbi decoding with soft decision are examined with inter-block interleaving. Reed-Solomon (RS) codes with Berlekamp-Massey hard decision decoding or soft decision trellis decoding combined with block interleaving are also investigated. A concatenated arrangement employing RS and CC coding as the outer and inner coders, respectively, is used for transmissions via minimum shift keying (MSK) over Gaussian and Rayleigh fading channels. For an interblock interleaving period of 2880 bits, a concatenated arrangement of an RS(48,36). over the Galois field GF(256) and punctured PCC(3,1,7) yielding an overall coding rate of 1/2, provides a coding gain of 42dB for a BER of 10^?6, and an uncorrectable error detection probability of 1–10^?9.