新一代语音编码标准——G.729

介绍了国际电联电信标准部门本研究期在语音编码标准化方面取得的最大成果－Ｇ．７２９语音编码的算法原理及测试结果，减少复杂度的附件Ａ及测试结果，今后的工作等。     

基于SMTP邮件系统的邮件寻呼的实现

介绍了基于常见的SMTP邮件系统的邮件寻呼实现,是传统的电信寻呼业和现有的计算机网络技术的良好结合。系统在客户端实现了POP3协议,另外本文还给出先进的MD5数字签名算法的流程。     

G.729标准与G.723.1标准编码方法区别的研究

介绍了G．729，G．723．1两种语音编码标准的主要用途和它们各自使用的算法。并就两种标准的编码算法的码率、线性预测分析、延时以及编码方法等几个方面进行了比较。在线性预测分析时主要是对两种算法在加窗时的不同进行了陈述；在编码方法的比较中又涉及到预处理时由于使用的高通滤波器的差别而导致G．729编码话音质量优于G．723．1的原因。     

基于G.729A和G.729D的宽带语音编码实现

根据7kHz声频编码标准G.722,采用正交镜像滤波器(Quadrature Mirror Fihers,QMF)法将宽带语音划分为等频带的子带,而后采用高质量的语音编码标准G．729,对低频带使用简化的G．729A,高频带使用较低比特的G．729D,实现了低时延的14．4Kb／s的宽带语音编码。     

基于DSP的G.729语音编解码器设计

设计了基于DSP的G.729语音编解码器,并针对G.729算法标准源码代码效率低、执行时间长的不足,从算法精简、代码优化等方面进行了优化。优化后的算法在保证了高质量语音输出的同时,提高了编码效率,实现了对语音信号的实时处理。最后对系统性能进行了测试,结果满足设计要求。     

语音压缩算法G.729的定点化

Ｇ．７２９是国际电信联盟（ＩＴＵ）于１９９６年推出的语音压缩标准,文中主要介绍了将该算法定点化时所采用的一些原则和方法,以及用ＡＤＳＰ２１８１实现定点化时的一些技术。     

基于G.729A建议的语音管理系统

首先介绍了ITU-TG.729A建议的编解码算法,然后根据此算法的特点,并结合实际系统的应用要求,介绍了基于此建议的语音管理系统的实现。整个系统完全采用软件编制,并达到了编解码的实时性,有较高的实用价值。     

G.729语音编码算法研究及其DSP实现

简单介绍G．729编解码算法原理,介绍如何用TMS320VC5402 DSP实现该算法。给出硬件的实现框图并提出了一些在软件设计中的优化方法及技巧。结合实验结果并对其进行了分析。     

单片DSP实现G.729A语音编解码器

国际电信联盟（ＩＴＵ）于１９９５年１１月推出的建议Ｇ．７２９称作“共轭结构代数码本激励线性预测编码方案”（ＣＳ－ＡＣＥＬＰ）。它将６４ｋｂｐｓ的ＰＣＭ信号压缩到８ｋｂｐｓ,是当前较新的语音压缩国际标准。其简化方案Ｇ．７２９Ａ进一步降低了计算的复杂度。Ｇ．７２９和Ｇ．７２９Ａ具有非常好的性能,在良好的信道条件下达到了长话质量,在有随机比特误码、发生帧丢失和多次转接等情况下有很好的稳健性。因此,该标准有广泛的应用前景。Ｇ．７２９编解码原理Ｇ．７２９算法以码本激励线性预测（ＣＥＬＰ）方法为基础。ＣＥＬ…     

基于G.729算法的语音处理系统设计

介绍了G.729算法的编解码原理,详细说明了在PXA255平台上使用WM9705音频编解码芯片构建语音采集与处理系统的软硬件方案,并利用INTELIPP多媒体库实现G.729编解码程序。     

基于DSP平台及G.729A算法的系统设计

应用TMS320VC5509A芯片,通过硬件和软件的设计与调试,实现了对G.729A标准的实时编解码,并应用最新的语音芯片TLV320AIC32,通过和5509A的无缝连接,简化了硬件设计,方便了软件的定点实现。     

基于ARM SoC的G.729A编码实现

G.729A是国际电信联盟(ITU)于1996年提出的一种语音压缩标准,在G.729基础上进行了必要的简化,该算法是一种共轭结构代数码激励线性预测(CS-ACELP)的8 kbits语音编码算法.文中主要介绍一种基于ARM处理器的系统架构和G.729A编码的过程,以及将该编码算法应用于此系统需要注意的一些原则和方法.针对算法本身采用了一些快速算法,并且针对ARM处理器的特性进行了一些优化,以便能降低编码复杂度,做到实时编码.     

G729 Voice Decoder Design

Embedded digital signal processing (DSP) systems are usually associated with real time constraints and/or high data rates such that fully software implementations are often not satisfactory. In that case, mixed hardware/software implementations are to be investigated. This paper presents the design of a HW/SW G.729 voice decoder dedicated to embedded systems. The decoder has been built around, on the one hand a reconfigurable digital circuit (FPGA) to achieve the so called IP hardware part—the autocorrelation computation—using a linear systolic array, and on the other hand a digital signal processor (DSP) for the remainder of the algorithm. Apart such an implementation is typically driven by the use of reusable component (IP) it is of great interest for new G729-based applications such as Voice over IP (VoIP) for example. It results in an overall reduction of the execution time per frame. Another interesting point is the design of a parameterizable autocorrelation block which can be useful for a wide range of applications such as GSM 13 Kbit/s, APC 9.6 Kbit/s and G723 6.3 Kbit/s and 5.3 Kbit/s. In the G729 context and using a V50 Virtex FPGA, the execution time of this function is 10 times faster than a TMS320C6201 DSP implementation. Fatma Sayadi is Ph.D. student at Faculty of Sciences, Monastir, Tunisia in collaboration with the LESTER Laboratory, University de Bretagne Sud, Lorient, France. She is a member of Laboratory of Electronics and Micro-Electronics. His researches interest, the implementation of Digital Signal, high level design using VHDL language, Hardware/Software Co-design. Emmanuel Casseau received his Ph.D Degree in Electrical Engineering in 1994. He is currently an Associate Professor in the Electronic Department at the University de Bretagne Sud, Lorient, France. He is also in charge of the IP project of the Lester Lab., University de Bretagne Sud. His research interests include system design, high-level synthesis, virtual components and SoCs. Mohamed Atri born in 1971, received his Ph.D. Degree in Micro-electronics from the Science Faculty of Monastir in 2001. He is currently a member of the Laboratory of Electronics & Micro-electronics. His research includes Circuit and System Design, Network Communication, IPs and SoCs. Mehrez Marzougui received the B.Sc. degree from University of Science and Technology (electronic option), Monastir, Tunisia, and the M.Sc. degree in electronic from the same university in 1996 and 1998 respectively. Since 1998, he has been a Ph.D. candidate in Electronic and Micro-electronic laboratory at the University of Sciences and Technology, Monastir, Tunisia. His research interests include hardware/software co-verification and high-level synthesis. Rached Tourki was born in 1948. He received the B.S. degree in Physics (Electronics option) from Tunis University, in 1970; the M.S. and the Doctorat de 3eme cycle in Electronics from Institut d'Electronique d'Orsay, Paris-south University in 1971 and 1973 respectively. From 1973 to 1974 he served as microelectronics engineer in Thomson-CSF. He received the Doctorat d'etat in Physics from Nice University in 1979. Since this date he has been professor in Microelectronics and Microprocessors with the physics department, Faculte des Sciences de Monastir. Eric Martin born in 1961, is a Full Professor at the University of South Brittany in Lorient, France. His interest includes the implementation of Digital Signal and Image Processing and high-level design methods for dedicated circuits.     

G．729标准码本搜索算法分析及优化

G.729用固定码本和自适应码本构成的激励通过合成滤波器恢复出较高质量的语音信号.由于算法复杂耗时过多,不能在DSP上实时实现,其中固定码本搜索和自适应码本搜索是最复杂的模块.介绍了码本搜索方法并进行了改进,使复杂度大大降低.结果表明语音质量没有明显下降.     

G．729中线谱对系数量化算法的硬件实现

首先介绍了基于G．729语音编解码标准的线谱对参数矢量量化的算法步骤及流程。根据ASIC设计方法和要求,用HDL设计硬件电路实现该算法,并将硬件电路仿真结果与C语言算法程序运算结果进行比较,结果显示该硬件设计能很好地实现该算法,结果准确;并比传统DSP编程实现的速度快很多,确保了计算过程的实时性,且所占资源更为节省。     

针对无线语音通信的G.729算法的改进及DSP实现

无线语音通信系统中,由于开放空间的信道衰落特性,使得语音信号在空间传输中受到较大的干扰。针对这一特点,提出了一种基于G.729协议的优化算法。该优化算法粗化了码本搜索,并通过定点溢出保护的去除和宏定义的使用改进了程序运行效率,和原算法相比有较低的计算复杂度和处理延时,利于无线语音的传输。通过DSP硬件仿真实验和人耳听觉辨别表明,在计算复杂度大幅下降的情况下,合成话音质量仍能达到令人满意的效果。