基于aacPlus v2的音频直播系统的设计与实现

【作者】 罗龙智；

【导师】 周南；

【作者基本信息】 电子科技大学 ， 信号与信息处理， 2007， 硕士

【摘要】 随着人民生活水平的逐步提高，人们对信息的需求也日益增加，而音频直播就成为了人们及时获得各种信息的重要手段。目前，广播电台主要使用专业音频直播车或ISDN专线进行直播，这两种方式成本都较高且对直播地点选择有依赖性。因此，研究一种携带方便、低成本、对地点选择适应性强的音频直播系统对电台工作和人民生活质量的提高都有现实意义。基于电路交换方式的PSTN网是当前较廉价和普及的网络，它具有时延小、数据传输可靠和错误率低的优点，满足音频直播对实时性和可靠性的需要，缺点是其提供的模拟话音质量不能达到专业音频节目的质量要求。但随着音频压缩技术的提高，aacPlus v2在低码率下能够提供高音质音频，用aacPlus v2作为信源编解码方式，用Modem通信技术在PSTN网上传输数据能够完成音频直播任务，因此，论文结合这两者的优点给出了音频直播系统的解决方案。首先，论文分析了音频直播系统的功能和性能需要，根据系统可扩展性和可移植性的要求，采用了模块化方法进行系统总体设计，把系统分为录音模块、播放模块、编码模块、解码模块、传输模块、缓冲模块和总体控制模块，然后给出了模块层次关系图。在传输模块的设计中，为了使系统能工作在不同的传输网络上，论文设计了一套简单的实时传输控制协议来向上层提供统一接口而向下层兼容不同的传输方式，还阐述了传输协议中传输控制包的定义内容和传输控制算法的具体设计，从而使系统完成了实时传输、实时监测、实时控制任务，解决了直播系统在传输过程中自适应编码和断线异常处理这两个难点。其次，为了进行实时和低延时传输，必须要做到实时编解码，而未优化前的编解码模块不能达到实时要求，因此，在系统的实现中，论文具体阐述了aacPlus v2的优化方法。在算法优化上，对量化部分常用到的非线性函数整数开平方提出了一种新的定点快速算法；在代码优化上，给出了具体的优化方法及步骤。此外，论文还详细叙述了如何用多线程技术实现系统功能，给出了系统单线和双线应用模式下的实现方法，并对此两种模式做了系统测试，测试结果满足专业音频直播系统的各项指标。最后，论文在系统的嵌入式应用上也进行了初步研究，对aacPlus v2在ADSP-BF533上的实时实现给出了具体方法和初期研究成果。更多还原

【Abstract】 With the development of people living standard, the information demand is alsoincreasing, leading to the audio living broadcast become to be the important way toobtain kinds of information. At present, the methods of audio living broadcast dependon broadcasting place and cost much, therefore, researching a system with low costand portable feature has practical and immediate significance to the work ofbroadcasting station as well as the improvement of people lives.PSTN, based on the circuit switching method, is considered as a suitable mediawhich is low cost and common net to satisfy the real-time and reliable transmissionrequirement of the audio living broadcast, however, the disadvantage is that it can notsupply professional audio quality because of the narrow band which can only supportanalog voice data transmission. Fortunatelly, with the development of the audiocompression technology, a new technique named aacPlus v2 can supply high audioquality in low bitrate. Thus, the association of aacPlus v2 and Modemtelecommunication fulfill the living broadcast mission, and a solution is elaborated inthis thsis.Firstly, by analyzing the function and performance requirement of the audio livingbroadcast system, we use modularized method to divide the system into severalmodules such as recording module, playing module, encoding module, decodingmodule, transmission module, buffer module and controlling module. Besides, themodule relationship graphic is presented at last. In order to make the system work ondifferent network, a simple transimission control protocol is devised to supply uniforminterface for the upper level and downward compatibility in the global system design.According to the definition of the protocol packet and the transmission controlalgorithm, real-time transmission, real-time monitoring and real-time controllingbecome possible, then adaptive encoding and offiine processing are both solved.Secondly, the encoding and decoding module which can not satisfy therequirements of the real-time transmission system, thus, an optimization is given asfollow. At the beginning, a new fast algorithm for square root of integer isdemonstrated to accelerate the quantization module of aacPlus v2. Next, some codeoptimization methods and the procedure are supplied. Furthermore, we discuss how torealize the function of system with multi-thread programming technique and supplysingle line and double line mode realization. At last, the test result satisfies therequirements of the professional audio living broadcast system. Finally, we take a preliminary research on embed system, and propose the detailedmethods and conclude the early studies on the optimization of aacPlus v2 algorithmbased on ADSP-BF533.更多还原