定点DSP在数字语音通信中的应用

本文介绍了新一代定点DSP芯片——TMS320C54x的体系结构和特点,及应用单片C548实现高质量、低延时的G.729协议的语音编码算法,并分析了定点DSP在实时实现数字语音通信中的一些关键技术。     

基于TMS320 C5416芯片的低时延 G.729A声码器的研制

简要介绍了G.729A声码器的编解码算法和定点数字信号处理芯片TMS320C5416,给出了基于TMS320C5416芯片实现低时延的G.729A声码器的硬件和软件设计,并对软件设计过程中应该注意的若干问题进行了分析和总结,最后给出了实时实现的测试结果.     

基于DSP的语音处理教学实验平台设计

为了帮助学生深入学习DSP实验教学中语音处理这一教学难点,文中介绍了一种基TMS320C5402的通用语音处理教学实验平台,以配合理论和实验教学.该系统采用TI的TMS320C5402作为处理器,选用TLC320AD50作为语音模块,实现对语音信号实时快速处理.     

一种语音压缩处理通用DSP系统的设计与实现

提出了一种基于TMS320VC549定点DSP微处理器芯片的语音信号压缩／解压缩处理系统。该系统采用TLC320AD50C模数、数模转换器,采样率为8kHz,处理能力为100MIPS。系统配置有大容量存储器,具备资源扩展能力,适用于语音信号压缩／解压缩和语音识别、语音合成等其他领域。实验结果表明。系统对语音信号的压缩解压缩处理具有稳定性、灵活性和通用性。     

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

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

Optimization and Realization of G.729 Speech Coding Algorithm

The principles of G.729 algorithm are analyzed.It proposes an optimal approach of adaptive codebook search.Realized on fixed point DSP TMS320VC5410,the searching time of the optimal algorithm is thus significantly decreased,and the result shows that the speech quality is not decreased.     

A Compiler-Friendly RISC-Based Digital Signal Processor Synthesis and Performance Evaluation

As DSP (Digital Signal Processing) applications become more complex, there is also a growing need for new architectures supporting efficient high-level language compilers. We try to synthesize a new DSP processor architecture by adding several DSP processor specific features to a RISC core that has a compiler friendly structure, such as many general-purpose registers and orthogonal instructions. The synthesized digital signal processor supports single-cycle MAC (Multiply-and-ACcumulate), direct memory access, automatic address generation, and hardware looping capabilities in addition to ordinary RISC instructions. The compiler for the new architecture is quickly implemented by developing a code-converter that modifies the assembly codes that are generated by the RISC compiler. The performance effects of adding each of these as well as all the combined features are evaluated using seven DSP-kernel benchmarks, a QCELP vocoder, and an MPEG video decoder. The effects of CPU clock frequency change due to the addition of these features are also considered. Finally, we also compare the performances with several existing DSP processors, such as TMS320C3x, TMS320C54x, and TMS320C5x.     

G.718宽带语音编解码器的DSP实现及优化

G.718是ITU-T最新提出的一种嵌入式可变速率宽带语音和音频编解码标准,该算法将语音信号进行分类编码,算法复杂度大大增加,但可以在窄带和宽带均达到极佳的语音质量.在分析其算法原理和关键技术的基础上,结合TMS320C55x系列DSP平台和G.718算法特点,提出了合理的汇编优化实现方案,在TMS320C5505EVM上完成了实时宽带语音编解码器.实验测试表明,G.718算法的语音质量优于同类型其他算法的宽带语音编解码器.     

车载FSK信号的实时高精度检测与DSP实现

本文基于16位定点运算精度的TMS320C50 DSP芯片,提出了一种通过频谱分析实现时高精度检测FSK信号的方法,并从理论上对主要技术给予了详细的论述,文章最后给出了具体硬件实现及部分实验结果。     

一种可用于CDMA移动通信的变速率语音编码算法

该文提出了一种码率为 0.75-5.4kb/s可变速率的高质量语音编码讲法。该算法对CELP的激励进行了改进,根据语音的特征把语音分成4类,不同类型的语音采用不同的激励码本。特别是对于浊音,提出了一种基于基音同步的嵌入分裂式激励码本,该码本利用浊音具有准周期性的特点,使该算法在很低的码率下就可很好地恢复浊音信号,克服了CELP在4kb/s速率以下因码本尺寸小而导致合成语音质量差的缺点。经非正式听音测试,它的主观质量超过了1～8kb/s的可变速率QCELP系统,并且平均速率大约只有2kb/s,比QCELP的5kb/s平均速率低了很多、非常适用于 CDMA移动通信系统。     

ITU-T G.729.A语音编解码器的实现

ＩＴＵ－Ｔ Ｇ．７２９．Ａ是ＩＰ电话中普遍使用的低速率语音编码器。介绍了 Ａ．７２Ａ．Ａ在ＴＭＳ３２０Ｃ６２０１ ＤＳＰ上的实现方法，研究了提高 Ｇ．７２９．Ａ的编码器的运行速度的优化方法。由于采用了这些优化方法，最后实现的语音编码器在一块ＴＭＳ３２０Ｃ６２０１支持下能同时处理大约２０路语音通道。     

基于DSP的数字语音压缩系统

给出了基于TMS320C6711 DSP和G.729A语音压缩标准的实时语音压缩和解压系统的设计方案,并给出了系统的软、硬件设计.在TMS320C6711高速平台上实现数字语音信号的实时输入、压缩、解压、输出,只需要移植应用程序,就可以实现其他语音功能的扩展.     

道路检测算法及其DSP实时实现

基于智能车视觉导航系统平台，在TMS320C6201定点数字信号处理器上实现了一种基于HSV色彩空间的道路检测算法。该算法能够从道路图像序列中成功地提取黄色和白色标志线，达到每秒14帧的处理速度。同时，本文深入探讨了TMSC6000DSP上的实时视频处理技术，如软件流水线以及其它一些DSP编程实现的问题。     

Development of the Test and Evaluation Systems for the CDMA Mobile System

In this paper, we present three kinds of development support systems (test and evaluation systems), which are developed to fulfill the specific needs in accordance with the CDMA Mobile System (CMS) developmental phases. The support systems consist of the mobile switch traffic simulator (MSTS), the mobile call simulator (MCS), and the CDMA mobile system analysis tools (CSAT). The MSTS is used for the simulation of the base station in the stage of development of the mobile exchange, and the MCS is used for the overall performance test of the CMS. The CSAT is used for the CMS performance evaluation. These systems have been used for the CMS development successfully.     

基于 TMS320VC5402实现双路全双工ITU G.728语音编解码

对G．728编解码算法和定点数字信号处理芯片TMS320VC5402作了简要介绍。并讨他在TMS320VC5402上实现双路全双工ITUG7．28语音编解码的关键技术,包括软件设计方法和算法优化技术。结果表明,所运行的编解码程序,既可以保证良好的语音质量,又有良好的实时性。     

The CDMA Mobile System Architecture

The architecture of the CDMA mobile system (CMS) is developed based on three function groups - service resource, service control, and service management groups. In this paper, the CMS architecture is discussed from the point of view of implementing these functions. The variable length packets are used for transmission. The synchronization clock signals are derived from the GPS receiver. The open loop and closed loop techniques are used for the power control. The internationally accepted signaling and network protocols are employed. The call control for the primary services is designed to provide efficient mobile telecommunication services. The softer handoff is implemented in one card. The mobile assisted handoff and the network assisted handoff are employed in the soft and hard handoffs. The authentication is based on the secret data which includes random numbers. The management functions, which include the location management, resource management, cell boundary management and OAM management, are implemented to warrant the system efficiency, maximum capacity and high reliability. The architecture ensures that the CMS is flexible and expandable to provide subscribers with economic and efficient system configuration. The dynamic power control, adaptive channel allocation, and dynamic cell boundary management are recommended for future work.     

基于TMS320C25音频处理系统的C语言编程

这篇文章着重讨论了Ｃ语言程序在以ＴＭＳ３２０Ｃ２５为基础的处理系统中的开发和应用。详细介绍了从Ｃ源程序到适用于Ｃ２５格式的程序的开发过程，同时讨论了在开发过程中可能遇到的问题及其解决办法。同时，文章对两种编程语言在ＴＭＳ３２０Ｃ２５系统中的应用前景作了简单的比较和预测。     

语音信号采集和播放的TLC320AD50的实现

通过基于TMS320VC5402与TLC320AD50芯片硬件电路及软件的设计,实现了语音信号的采集和播放.简述了语音采集和播放系统的用途,详细阐述了硬件电路设计的步骤和基本原理,以及A/D芯片TLC320AD50的工作机制.整个系统根据外围电路和语音处理算法的不同,具有较好的扩展能力和适应性.     

CELP语音编码与TMS320C54x

介绍了码激励线性预测（ＣＥＬＰ）编码的基本原理和新近推出的ＴＭＳ３２０Ｃ５４ｘ定点ＤＳＰ芯片,结合其指令特点探讨了ＴＭＳ３２０Ｃ５４ｘ在实现ＣＥＬＰ类语音编码方案的一些有效的编程方法。     

变码率分类型实时语音声码器

本文提出的声码器将语音分成静音、清音、浊音和混合音四类。用自适应方法进行分频带清浊音判决和有声／无声判决,提高了分类算法的稳定性、准确性和灵活性、准确性和灵活性,还保持了混合语音的音质,且无须对清浊音判决结果进行编码。对清音和浊音的频谱分别采用不同的ＬＳＰ量化表进行编码,从而用标量量化器替代子矢量量化器,降低了复杂度。声码器的码率最高２．４ｋｂｐｓ,最低为１００ｂｐｓ,平均码率１．４ｋｂｐｓ。实时