G.723.1标准在TMS320VC5402上的实现
摘要
在现代各种通信系统中，话音通信一直是一项重要的、必不可少的业务。随着通信网络用户数量的增加、网络业务更加综合化、多样化，系统容量、服务质量与网络带宽的矛盾也日益突出，如何在不牺牲语音通话质量的前提下尽可能降低话音信号传输的比特速率是摆在研究者面前的重要课题。同时，由于用数字化的方法进行语音信号处理是目前发展最为迅速的信息学研究领域之一，由此诞生了许多种语音压缩处理方法。ITU-T也在此基础之上提出了一系列关于低速率语音编码的解决方案，G.723.1标准就是其中之一。
本文首先从语音产生的离散数字模型出发，简要叙述了低速率语音编码的基本原理。然后文章对获得广泛应用(并且也是G.723.1标准的基础)的线性预测编码声码器的一些关键技术进行了详细的论述。
在G.723.1标准的实现过程中，我们首先建立了语音编码硬件基础，基于TI公司的TMS320C54xx系列DSP平台，对此做了比较详细的介绍。然后，从系统级的角度剖析了G.723.1标准的结构，并提出了在TMS320C 5402上实现该标准的一个解决方案。最后讨论了该方案实现过程中所遇到的一些问题及其解决方法。
最后，考虑到算法标准本身的不断改进，以及具体实现中需要重点考虑的降低算法复杂度的问题，在论文的最后一章，就如何进一步降低编解码算法的速率及复杂度等问题作了一些探索。
Speech Communication is an important and indispensable service in different modern communication systems. With the increasing of network users > the integrating and diversifying of network service, the contradictions among bandwidth, system capacity and service quality are more and more obvious. How to reduce the bit-rate of speech without remarkably degrading its perceptual quality is a question placed at the front of the researchers. At the same time, the process of speech signal by digital is developed rapidly in information science. There are many ways been brought forward in voice compres^. ITU-T has also put forward a series of ones in low rate speech coder, including recommendation G.723.1.
At first, from the discrete digital model of the speech generate, this paper briefly recount some basic terms and principles in low rate speech processing. Then focus on the research of the linear predictive coding, which is used widely (and is the base of G.723.1 too), and narrate in detail some key techniques in this area.
During the concretely realize of the recommendation G.723.1, we set up the hardware elements of the speech coding, which is the TMS320C54xx DSP platform of TI Inc. We introduce it in this paper too., In the following we discuss the framework of the G.723.1 from the systematic view, and bring out a blue print of implementation this recommendation. Combined to my works, we even introduce the problems we met and the solutions we found.
At last, considering the algorism is mended, some innovative improvement thoughts on further reducing the code-rate and decreasing the complexity of vocoders are included in this dissertation.
本文首先从语音产生的离散数字模型出发，简要叙述了低速率语音编码的基本原理。然后文章对获得广泛应用(并且也是G.723.1标准的基础)的线性预测编码声码器的一些关键技术进行了详细的论述。
在G.723.1标准的实现过程中，我们首先建立了语音编码硬件基础，基于TI公司的TMS320C54xx系列DSP平台，对此做了比较详细的介绍。然后，从系统级的角度剖析了G.723.1标准的结构，并提出了在TMS320C 5402上实现该标准的一个解决方案。最后讨论了该方案实现过程中所遇到的一些问题及其解决方法。
最后，考虑到算法标准本身的不断改进，以及具体实现中需要重点考虑的降低算法复杂度的问题，在论文的最后一章，就如何进一步降低编解码算法的速率及复杂度等问题作了一些探索。
Speech Communication is an important and indispensable service in different modern communication systems. With the increasing of network users > the integrating and diversifying of network service, the contradictions among bandwidth, system capacity and service quality are more and more obvious. How to reduce the bit-rate of speech without remarkably degrading its perceptual quality is a question placed at the front of the researchers. At the same time, the process of speech signal by digital is developed rapidly in information science. There are many ways been brought forward in voice compres^. ITU-T has also put forward a series of ones in low rate speech coder, including recommendation G.723.1.
At first, from the discrete digital model of the speech generate, this paper briefly recount some basic terms and principles in low rate speech processing. Then focus on the research of the linear predictive coding, which is used widely (and is the base of G.723.1 too), and narrate in detail some key techniques in this area.
During the concretely realize of the recommendation G.723.1, we set up the hardware elements of the speech coding, which is the TMS320C54xx DSP platform of TI Inc. We introduce it in this paper too., In the following we discuss the framework of the G.723.1 from the systematic view, and bring out a blue print of implementation this recommendation. Combined to my works, we even introduce the problems we met and the solutions we found.
At last, considering the algorism is mended, some innovative improvement thoughts on further reducing the code-rate and decreasing the complexity of vocoders are included in this dissertation.
引文
【1】 杨行峻 迟惠生，语音信号数字处理，电子工业出版社，1995．8
【2】 胡航著，语音信号处理，哈尔滨工业大学出版社，2000年5月第一版
【3】 精英工作室著，IP电话完全手册，中国电力出版社，2000年9月第一版
【4】 [美]Jerry D．Gibson Toby Berger等，多媒体数字压缩原理与标准，李煜晖 朱山风等译，电子工业出版社，2000年8月第一版
【5】 胡广书 数字信号处理—理论、算法与实现 清华大学出版社 1997年8月
【6】 曹志刚 钱亚生 现代通信原理 清华大学出版社 1997年
【9】 彭启琮 主编 TMS320C54X实用教程 电子科技大学出版社 2000年1月
【10】 彭启琮 李玉柏 DSP技术 电子科技大学出版社 1997年
【11】 华刚，崔慧娟等 一种高质量语音编解码专用芯片的设计，清华大学学报，2001．1
【12】 唐昆，崔慧娟等 高质量4kbit／s码速率语音编码算法研究
【13】 马金明 裘正定等，低比特率语声编码器的新发展，北方交大学报，1998年6月第3期
【14】 倪维桢 语音编码技术综述 数字通信 1998年第2期p3—5
【15】 汪润生 关于IP电话的若干问题的分析与探讨 数据通信，1999．4，
【16】 粱翎 李爱齐 C语言用户接口编程技巧 学苑出版社 1994年
【17】 ITU-T.Draft Recommendation G.723. 1 Dual rate speech coder for multimedia telecommunications transmitting at 5.3&6.3 kbit/s. Ocotober, 1995
【18】 ITU-T. Annex A to Recommendation G.723.1 Silence compression scheme for dual rate communications transmitting at 5.3 & 6.3 kbit/s. May, 1996
【19】 N.Sugamura and N.Farvardin, "Quantizer Design in LSP Speech Analysis_synthesis", IEEE Journal on Selected Areas in Communications, Vol 6 No.2 February 1988.
【20】 F.K. Soong and B-H Juang, "Line Spectrum Pair(LSP) and Speech Data Compression "Proc. ICASSP 84, pp. 1.10.1-1.10.4.
【21】 Kleijn, W., D. Krasinski and R. Ketchum, "An Efficient Stochastically Excited Linear Predictive Coding Algorithm for High Quality Low Bit Rate Transmission of Speech," Speech Communication, October 1988, p, 305-316.
[22] Peter Kroon. Ed F. Deprettere, "A Class of Analysis-by-Synthesis Predictive Coders for High Qualit Speech Coding at Rates Between 4． 8 and 16 kbits/s", IEEE Jounnal on Selected Areas in Communications, Vol.6, No.2, Feb 1988
[23 ]Linde Y, A.Buzo and R.M.Gray, An Algorithm for Vector Quantization Design. IEEE Trans, Vol.Com-28, no. 1, pp.84 ~ 95, January 1980
[24] H Peens and EC Botha Automatic detection of human stress in speech based on pitch extraction, PRASA '97
[25] Spanias A S, Speech Coding: A Tatorial Review, Proc. IEEE, 1994
[26] Texas Instruments, G723． 1 Dual Rate Speech Coder-Multichannel TMS320C6000 Implementation 2000
[27 ] Texas Instruments, G.723． 1 Speech Codec SIGNAL PROCESSING ASSOCIATES SPECIFICATION SHEET 2000
[28] Texas Instruments, Implementation of G.729 on TMS320C54x 2000
[29] Texas Instruments, TMS320C54X Reference Guide Volume 1 CPU & Peripheral Dsp CPU and Peripherals 1997
[30] Texas Instruments, TMS320C54X Reference Guide Volume 2 Assembly Language Tools 1997
[31] Texas Instruments, TMS320C54X Reference Guide Volume 4 Applecations Guide 1997
[ 32 ] Texas Instruments, TMS320C54XX Reference Guide Volume 5 DSP Enhanced Peripherals 1999
[ 33 ] Texas Instruments, TMS320C54x DSP Optimizing C Compiler 1999
[34] Texas Instruments, TMS320C5402 Datasheet 1999
[35] Texas Instruments, eXpressDSP Algorithm Standard Demonstration Application 1999
【2】 胡航著，语音信号处理，哈尔滨工业大学出版社，2000年5月第一版
【3】 精英工作室著，IP电话完全手册，中国电力出版社，2000年9月第一版
【4】 [美]Jerry D．Gibson Toby Berger等，多媒体数字压缩原理与标准，李煜晖 朱山风等译，电子工业出版社，2000年8月第一版
【5】 胡广书 数字信号处理—理论、算法与实现 清华大学出版社 1997年8月
【6】 曹志刚 钱亚生 现代通信原理 清华大学出版社 1997年
【9】 彭启琮 主编 TMS320C54X实用教程 电子科技大学出版社 2000年1月
【10】 彭启琮 李玉柏 DSP技术 电子科技大学出版社 1997年
【11】 华刚，崔慧娟等 一种高质量语音编解码专用芯片的设计，清华大学学报，2001．1
【12】 唐昆，崔慧娟等 高质量4kbit／s码速率语音编码算法研究
【13】 马金明 裘正定等，低比特率语声编码器的新发展，北方交大学报，1998年6月第3期
【14】 倪维桢 语音编码技术综述 数字通信 1998年第2期p3—5
【15】 汪润生 关于IP电话的若干问题的分析与探讨 数据通信，1999．4，
【16】 粱翎 李爱齐 C语言用户接口编程技巧 学苑出版社 1994年
【17】 ITU-T.Draft Recommendation G.723. 1 Dual rate speech coder for multimedia telecommunications transmitting at 5.3&6.3 kbit/s. Ocotober, 1995
【18】 ITU-T. Annex A to Recommendation G.723.1 Silence compression scheme for dual rate communications transmitting at 5.3 & 6.3 kbit/s. May, 1996
【19】 N.Sugamura and N.Farvardin, "Quantizer Design in LSP Speech Analysis_synthesis", IEEE Journal on Selected Areas in Communications, Vol 6 No.2 February 1988.
【20】 F.K. Soong and B-H Juang, "Line Spectrum Pair(LSP) and Speech Data Compression "Proc. ICASSP 84, pp. 1.10.1-1.10.4.
【21】 Kleijn, W., D. Krasinski and R. Ketchum, "An Efficient Stochastically Excited Linear Predictive Coding Algorithm for High Quality Low Bit Rate Transmission of Speech," Speech Communication, October 1988, p, 305-316.
[22] Peter Kroon. Ed F. Deprettere, "A Class of Analysis-by-Synthesis Predictive Coders for High Qualit Speech Coding at Rates Between 4． 8 and 16 kbits/s", IEEE Jounnal on Selected Areas in Communications, Vol.6, No.2, Feb 1988
[23 ]Linde Y, A.Buzo and R.M.Gray, An Algorithm for Vector Quantization Design. IEEE Trans, Vol.Com-28, no. 1, pp.84 ~ 95, January 1980
[24] H Peens and EC Botha Automatic detection of human stress in speech based on pitch extraction, PRASA '97
[25] Spanias A S, Speech Coding: A Tatorial Review, Proc. IEEE, 1994
[26] Texas Instruments, G723． 1 Dual Rate Speech Coder-Multichannel TMS320C6000 Implementation 2000
[27 ] Texas Instruments, G.723． 1 Speech Codec SIGNAL PROCESSING ASSOCIATES SPECIFICATION SHEET 2000
[28] Texas Instruments, Implementation of G.729 on TMS320C54x 2000
[29] Texas Instruments, TMS320C54X Reference Guide Volume 1 CPU & Peripheral Dsp CPU and Peripherals 1997
[30] Texas Instruments, TMS320C54X Reference Guide Volume 2 Assembly Language Tools 1997
[31] Texas Instruments, TMS320C54X Reference Guide Volume 4 Applecations Guide 1997
[ 32 ] Texas Instruments, TMS320C54XX Reference Guide Volume 5 DSP Enhanced Peripherals 1999
[ 33 ] Texas Instruments, TMS320C54x DSP Optimizing C Compiler 1999
[34] Texas Instruments, TMS320C5402 Datasheet 1999
[35] Texas Instruments, eXpressDSP Algorithm Standard Demonstration Application 1999