摘要
随着卫星通信、数字移动通信和个人通信网的迅猛发展，日益增加的客户需求量与现有通信信道容量之间的矛盾日益突出。如何在现有的信道资源条件下，通过压缩信源以提高传输效率已成为当前急待解决的问题之一。相继出现的语音信号压缩标准为语音信号的高效传输提供了一种有效方法，其实质就是在相当的语音质量指标下，降低数字化语音的数码率。
G.723算法是ITU-T建议的应用于低速率多媒体服务中语音或其它音频信号的压缩算法，例如：H.323，H.324系统。这种声码器具备两种比特率：5.3kbps，6.3kps。在帧边界处可以在两种速率之间进行切换。本算法提供对无声语音帧的检测以及在无声时进行舒适噪声填充的功能。如果优化系统，有限地提高其复杂度，将会得到更高的语音质量。G723.1算法同样适用于音乐或其它声音信号，但是处理效果不如语音。
G723.1是一种参数编码，是目前制定的一系列采用混合编码技术的语音编码标准中编码效率最高的，它以其卓越的性能被广泛应用到各种领域，与G.711等波形编码标准相比，其复杂度提升了很多，相应地大大增加了实现的难度。G.723.1的高质量和低码率是以其高复杂度的编码算法、较高的延迟以及较大的存储空间换得的，这也大大增加了它的实时实现难度。将算法转化为应用的第一步是准确的理解算法，并编写成代码，转化为可运行的程序，这是先决条件。但是，算法的正确实现并不保证它的性能能满足用户的要求，这就要求根据实际的需求对算法进行改进。同时，在实现过程中，程序的编写还要受到硬件资源的限制，例如内存的大小，CPU的速度等。
本论文首先对语音编码的基础技术进行了详细的描述，同时在深入研究G.723.1语音编码标准的基础上，给出了C语言的实现，并从码本搜索的角度，对ITU-T制定的标准提出了优化策略；同时，在介绍TMS320C541 DSP硬件环境的基础上，给出G.723.1实时实现的系统设计、硬件设计及软件设计的流程。
With the rapid development of satellite communications, digital mobile communications and personal communications, the conflict between user's demand and capability of communication channels is more and more outstanding. How to improve the usage efficiency of resources available is the focus to which people have put much attention. Now, the standards of speech compression coding provide a way of transporting speech signals efficiently. In fact, all of them are to reduce the baud rate of data under definite speech quality.
G723 aerithemetic is a compressing arithemetic that proposed by ITU-T and applied in speech and other audio frequency signals of low velocity multimedia services,such as:H.323,H.324 system.This arithemetic provides inspection to silence speech frames and fills in comfortable noise when it is silence.If optimize system and increase the complexity limitedly,we can get higher quality of speech.G723.1 is also available in music or other voice signals,but the managing effect is not as good as speech's.
Among the standards which are established using mixed coding technology, G723.1 has the highest efficiency, and is widely used in many fields. But, to gain the high quality and low baud rate, G723.1 pays out the costs, such as the high complex of coding arithmetic, high delay and large store space. Each of them brings up the difficulties of its real-time implementation. The first step of use the arithmetic is to understand it accurately and code,then change it into program can be run, this is the precondition, but we can't assure that the capability can meet the demand of customers, so we shoud improve the arithmetic according to the demand of practice. At the same time, in practice, the program also limited by the hardware such as:size of memory,speed of CPU,etc.
Detailed the basic technology of speech coding and based on the in-depth study of G723.1, the paper implements it using C language and provides some optimizing policies with respect to code-book research. At the same time, based on the analysis of TMS320C5410 EVM environment, the paper gives system design, hardware design and the flow of software design.
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