数字音频广播中的音频编码新技术

在广播系统中,由于带宽的限制,传统的信源编码技术已难以满足高质量音频输出的要求,同时由于广播系统传输信道的复杂性,对信道编码也提出更高的要求,为了进一步提高压缩比和信道传输性能,出现了许多信源和信道编码新技术,本文对其中的一些技术进行了介绍。     

DVB概述

１引言数字视频广播（DigitalVideoBroadcasting，DVB）是一种基于MPEG－２的国际标准传输技术，它包含了HDTV在内多种数字电视格式的的广播和传输。欧洲DVB规划的范围可用图１来表示。信源编码和系统复用都遵循MPEG国际标准。信道编码则根据信道不同有卫星传输、有线传输、地面开路广播几种。由于将来的DVB业务趋向于付费电视方式，因此，DVB系统还需要一个高度可靠的有条件接收系统。２DVB中的信源和信道编码方式DVB包含了信源和信道两部分标准。MPEG－２信源编码采用MPEG－２标准，利用了人体视、听觉的生理特性，把图…     

噪声信道中基于残留冗余的联合信源信道编码.

对信源编码中的残留冗余在联合编码中的作用进行了研究，提出了一个在噪声信道中对可变长信源编码码流传输提供有效差错保护的联合信源信道编码方法，该方法利用信源编码器输出中的残留冗余为传输码流提供差错保护。与Sayood K提出的系统相比，该方法是基于改进的联合卷积软解码以及采用非霍夫曼码的通用可变长码，更接近于一般的信源和信道编码方法，并且信源符号集的大小也不受限制。仿真表明，所提出的联合编码方法可获得比传统的分离编码方法更高的性能增益。     

基于多描述的联合信源信道编码

在实时通信系统中,传统的联合信源信道编码方法在分组编码长度有限的情况下,不可避免地会存在残余误差,从而影响了整体性能。提出了一种基于多描述源编码的联合信源信道编码方法,此方法不仅可以利用分组编码的纠错能力,也可以利用多描述源编码的鲁棒性。通过具有代表性的高斯信号编码序列在有损网络信道中的传输仿真,可以看出,此方法相对于传统的联合信源信道编码方法,可提供更好的信噪比性能。     

噪声信道中基于残留冗余的联合信源信道编码

对信源编码中的残留冗余在联合编码中的作用进行了研究,提出了一个在噪声信道中对可变长信源编码码流传输提供有效差错保护的联合信源信道编码方法,该方法利用信源编码器输出中的残留冗余为传输码流提供差错保护。与SayoodK提出的系统相比,该方法是基于改进的联合卷积软解码以及采用非霍夫曼码的通用可变长码,更接近于一般的信源和信道编码方法,并且信源符号集的大小也不受限制。仿真表明,所提出的联合编码方法可获得比传统的分离编码方法更高的性能增益。     

无线信道中的联合信源信道编码

该文提出了一种噪声信道下传输渐进图像的联合信源信道编码方法。该方法根据信道条件的好坏动态的调整信源编码速率和信道编码速率，因此极大地提高了系统的性能和编码效率。同时该方法还具有结构简单，易于实现等优点。试验证明本方法与以前文献中提出的EEP方法以及UEP方法相比，在信噪比低时即信道条件恶劣的情况下，能够明显提高恢复图像的质量。     

DVB系统中的信道结构分析

本从信道角度出发。对DVB两大支撑技术——信源编码和信道编码的结构进行了分析。分析了在不同的传输信道上。DVB所采取的传输方式标准，以及适合该信道的编码方式与调制技术，并对以数字电视为代表的数字音频产业进行了展望。     

渐进图像传输中基于失真的码率分配

该文根据信源编码压缩比、信道码率及信道特征对渐进图像传输失真的影响,提出一种基于最小图像失真的动态码率分配策略。在总的码率限制下采用遍历搜索信道码率的方法,得出可用码率在信源编码和信道编码之间的最优分配,使端对端的失真最小。     

联合信源信道编码的原理及其在无线通信中的应用

根据香农信源和信道分离编码理论进行的分离信源和信道编码在时变信道时不能充分利用系统资源。正是在这种情况下 ,提出了信源信道联合编码 ,可以跟随信道的变化充分利用通信系统的资源 ,达到最好的端对端的通信效果。本文首先阐述了联合信源信道编码的原理 ,然后介绍了常用的实现方法 ,最后 ,提出了在设计联合信源信道编码系统时的改进方法。     

MARC中联合BICM-ID和网络编码设计及性能分析

联合网络信道编码是将网络编码和信道编码联合处理,通过联合处理网络信道编码可以增加带有噪声信道的分集数,从而进一步提高网络容量。结合联合网络信道编码的思想,基于比特交织编码调制迭代译码(BICM-ID)设计了两种应用于多址接入中继信道(MARC)环境中的联合网络信道编码方案,并对这两种方案进行了仿真比较。仿真结果表明:在译码之前对解调后的软信息直接进行网络译码相比于对译码后的外信息进行网络译码能更加有效地利用中继信道传输的冗余信息。     

Robust transmission of MELP-compressed speech: an illustrative example of joint source-channel decoding

A mixed-excitation linear predictive (MELP) speech coder was selected as the US federal standard for 2400 b/s speech compression. This paper examines the quality of MELP-compressed speech when transmitted over noisy communication channels in conjunction with a variety of error-control schemes. The focus is on channel decoders that exploit the "residual redundancy" inherent in the MELP bitstream. This residual redundancy, which is manifested by the correlation in time and the nonuniform distribution of various MELP parameters, can be quantified by modeling the parameters as one-step Markov chains and computing the entropy rate of the Markov chains based on the relative frequencies of transitions. Moreover, this residual redundancy can be exploited by an appropriately "tuned" channel decoder to provide substantial coding gain when compared with decoders that do not exploit it. Channel coding schemes include conventional binary convolutional codes and iteratively-decoded parallel concatenated convolutional (turbo) codes.     

Robust Transmission of Multistage Vector Quantized Sources Over Noisy Communication Channels—Applications to MELP Speech Codec

Joint source-channel coding is an effective approach for the design of bandwidth efficient and error resilient communication systems with manageable complexity. An interesting research direction within this framework is the design of source decoders that exploit the residual redundancy for effective signal reconstruction at the receiver. Such source decoders are expected to replace the traditionally heuristic error concealment units that are elements of most multimedia communication systems. In this paper, we consider the reconstruction of signals encoded with a multistage vector quantizer (MSVQ) and transmitted over a noisy communications channel. The MSVQ maintains a moderate complexity and, due to its successive refinement feature, is a suitable choice for the design of layered (progressive) source codes. An approximate minimum mean squared error source decoder for MSVQ is presented, and its application to the reconstruction of the linear predictive coefficient (LPC) parameters in mixed excitation linear prediction (MELP) speech codec is analyzed. MELP is a low-rate standard speech codec suitable for bandwidth-limited communications and wireless applications. Numerical results demonstrate the effectiveness of the proposed schemes     

A joint source/channel coder with block constraints

The assumptions made about the source during source coder design result in a residual redundancy at the output of the source coder. This redundancy can be utilized for error protection without any additional channel coding. Joint source/channel coders obtained using this idea via maximum a posteriori probability decoders tend to fail at low probability of error. In this paper, we propose a modification of the standard approach which provides protection at low error rates as well     

A High-Performance Codec Suitable for the GSM Half-Rate Coding System

1IntroductionTheGSMpanEuropeandigitalradiosystemhas-beendesignedwithaparticularTDMAframestfllcturewhichenablestheusingofeitherfull-rateorhalf-ratechannels.Speechandchannelcodingalgorithmsforfull-ratechannelshavebeenindependentlystandardized,leadingrespectivelytotheRPE-LTPalgorithmandprotectionschemebasedonaconvolutionalcodewithaCRCforerrordetection.StandardiZationofacombinedspeechandchannelhalf-ratecodecataglobalrateofII.4kbpshasstartedunderthecontrolofETSI.Theobjectiveisverychalleng…     

一种面向保密通信的改进型MBE语音编码算法

随着军事通信的应用需求迅速扩展,如何有效地在信源端对语音信号进一步压缩。并且在复杂信道条件下实现高质量的低速率语音编码技术是一个重要研究方向。以MBE语音编码模型为基础,提出了一种改良算法,即在编码端利用信源冗余度,将对语音合成质量影响较大的参数进行检纠错保护,并在解码端采用谐波增强以改善终端语音合成质量。测试数据表明,在1％-3％的信道误码条件下,PESQ评分平均提高了近14％。     

语音编码综述

着重阐明语音编码的分类,质量评定,压缩编码的基本方法。介绍了有关语音编码的标准,并探讨了今后语音编码的研究方向。     

Reliable Speech Transmission Using Combined Adaptive-rate Sampling (ARS) and Channel Coding

A novel speech transmission technique over noisy channel condition based on combined ARS and channel coding schemes is described. The proposed scheme is suitable for time varying channels with some form of ARQ facilities. It is shown that under an AWGN channel condition, up to 3.8 dB gain relative to a conventional 32 kbits/s transmission can be provided using the combined technique.     

Side information creation for efficient Wyner–Ziv video coding: Classifying and reviewing

Video coding technologies have played a major role in the explosion of large market digital video applications and services. In this context, the very popular MPEG-x and H-26x video coding standards adopted a predictive coding paradigm, where complex encoders exploit the data redundancy and irrelevancy to ‘control’ much simpler decoders. This codec paradigm fits well applications and services such as digital television and video storage where the decoder complexity is critical, but does not match well the requirements of emerging applications such as visual sensor networks where the encoder complexity is more critical. The Slepian–Wolf and Wyner–Ziv theorems brought the possibility to develop the so-called Wyner–Ziv video codecs, following a different coding paradigm where it is the task of the decoder, and not anymore of the encoder, to (fully or partly) exploit the video redundancy. Theoretically, Wyner–Ziv video coding does not incur in any compression performance penalty regarding the more traditional predictive coding paradigm (at least for certain conditions). In the context of Wyner–Ziv video codecs, the so-called side information, which is a decoder estimate of the original frame to code, plays a critical role in the overall compression performance. For this reason, much research effort has been invested in the past decade to develop increasingly more efficient side information creation methods. This paper has the main objective to review and evaluate the available side information methods after proposing a classification taxonomy to guide this review, allowing to achieve more solid conclusions and better identify the next relevant research challenges. After classifying the side information creation methods into four classes, notably guess, try, hint and learn, the review of the most important techniques in each class and the evaluation of some of them leads to the important conclusion that the side information creation methods provide better rate-distortion (RD) performance depending on the amount of temporal correlation in each video sequence. It became also clear that the best available Wyner–Ziv video coding solutions are almost systematically based on the learn approach. The best solutions are already able to systematically outperform the H.264/AVC Intra, and also the H.264/AVC zero-motion standard solutions for specific types of content.