网络数字会议电话回波抵消技术研究
|
摘要
回波是由于2线或2-4线混合的电话线路中阻抗的不匹配产生的。传统的PCM(脉冲编码调制)数字电话信号长途传输是由SDH(同步数字系列)和PDH(准同步数字系列)实现的。目前,消除这种数字电话中产生的回波的方法主要有最小均方(LMS)自适应算法、归一化最小均方(NLMS)自适应算法以及近几年来研究的基于这些算法的改进的自适应算法,这些算法都是符合国际电信联盟标准(ITU-T)G.165和G.168的。
数字网络会议电话不仅可以实现传统视频会议的音视频交流,还具备功能先进的数据共享(各种文档或软件)和协同办公等功能。由于网络会议电话中存在着多方同时讲话和路由变化的问题,国际电话联盟技术标准(ITU-T G.165和G.168)是针对SDH和PDH网络制定,应用于网络会议电话时现有的回波抵消算法不能满足要求。
本文将信息隐藏技术中的数字音频水印技术应用于数字网络会议电话的回波抵消中。把具有低功率谱密度的伪随机序列嵌入发送端的语音信号中,在接收端解调出伪随机序列,从而测量出本段话音数据包的网络延迟时间,将回波消除。在多端通话时,加在每一用户语音信号中的伪随机序列不同,根据码分多址的工作原理,可以辨认出是哪一方在讲话,从而消除回波。
通过理论分析及计算机仿真可以看出:本研究为减少网络数字会议电话回波提供了一种可行的技术方法,可以解决会议电话多方通话回波抵消中身份识别、语音数据包分段时传输网络延迟不同、改进传统自适应算法的收敛速度问题。数字音频水印技术的应用也使伪随机序列的嵌入几乎不影响通话的质量。应用本技术的数字网络会议电话回波抵消完整解决方案有待于进一步研究, ITU-T G.165和G.168中关于数字网络会议电话的技术指标和测试方法有必要探讨。
Echo, in any 2-wire or combination 2- and 4- telephone circuit, is caused by impedance mismatches. Traditional PCM (Pulse Code Modulation) digital telephone, is the implementation of SDH (Synchronous Digital Hierarchy) and PDH (Plesiochronous Digital Hierarchy). At present, the methods of eliminating the echo which generates from the digital telephone are mainly Least Mean Squares (LMS) adaptive algorithm, Naturalization Least Mean Squares (NLMS) adaptive algorithm and some kinds of improved adaptive algorithms based on these algorithms recent years. All above algorithms accord with the telecommunication standardization sector of the International Telecommunication Union (ITU-T) G.165 and G.168.
Not only digital network conference telephone can complete the implementation of audio and video communication on traditional video conference, but also it can complete the implementation of shared data (various documents and software) and cooperative office. Because of the change of routing and multipart talking simultaneously in the network conference telephone, the technical standard of the International Telecommunication Union (ITU-T G.165/G.168) is established according to SDH and PDH, the algorithms available for echo cancellation can’t meet the demand when they are applied to network conference telephone.
According the digital audio watermarking technology of information hiding technique, this paper applies it to the echo cancellation of network conference telephone. The pasedo-random sequence with low power spectrum density is embedded into audio signal which is in the sender, and demodulated in the receiver, and then the network delays of the audio packet is measured, the echo is eliminated. During the multipart talking, with different PN sequences in every user’s audio signal and the theory of CDMA (Code Division Multiple Access), which one is talking can be identified, and then the echo can be eliminated.
Through theoretical analysis and computer simulation, the research provides a feasible technical method, it can solve the personal identification of echo cancellation during multipart talking in conference telephone, and the different network delays during the segmentation of audio packets, modify the problem of convergent rate in traditional adaptive algorithms. Owing to the application of the digital audio watermarking technology, it is hardly to influence the talking quality that embedding the PN sequence.
The integrated scheme about echo cancellation of digital network conference telephone which applies the technology remains to further research, it is necessary to investigate the technical indexes and testing methods which are included in ITU-T G.165 and G.168 about digital network conference telephone.
数字网络会议电话不仅可以实现传统视频会议的音视频交流,还具备功能先进的数据共享(各种文档或软件)和协同办公等功能。由于网络会议电话中存在着多方同时讲话和路由变化的问题,国际电话联盟技术标准(ITU-T G.165和G.168)是针对SDH和PDH网络制定,应用于网络会议电话时现有的回波抵消算法不能满足要求。
本文将信息隐藏技术中的数字音频水印技术应用于数字网络会议电话的回波抵消中。把具有低功率谱密度的伪随机序列嵌入发送端的语音信号中,在接收端解调出伪随机序列,从而测量出本段话音数据包的网络延迟时间,将回波消除。在多端通话时,加在每一用户语音信号中的伪随机序列不同,根据码分多址的工作原理,可以辨认出是哪一方在讲话,从而消除回波。
通过理论分析及计算机仿真可以看出:本研究为减少网络数字会议电话回波提供了一种可行的技术方法,可以解决会议电话多方通话回波抵消中身份识别、语音数据包分段时传输网络延迟不同、改进传统自适应算法的收敛速度问题。数字音频水印技术的应用也使伪随机序列的嵌入几乎不影响通话的质量。应用本技术的数字网络会议电话回波抵消完整解决方案有待于进一步研究, ITU-T G.165和G.168中关于数字网络会议电话的技术指标和测试方法有必要探讨。
Echo, in any 2-wire or combination 2- and 4- telephone circuit, is caused by impedance mismatches. Traditional PCM (Pulse Code Modulation) digital telephone, is the implementation of SDH (Synchronous Digital Hierarchy) and PDH (Plesiochronous Digital Hierarchy). At present, the methods of eliminating the echo which generates from the digital telephone are mainly Least Mean Squares (LMS) adaptive algorithm, Naturalization Least Mean Squares (NLMS) adaptive algorithm and some kinds of improved adaptive algorithms based on these algorithms recent years. All above algorithms accord with the telecommunication standardization sector of the International Telecommunication Union (ITU-T) G.165 and G.168.
Not only digital network conference telephone can complete the implementation of audio and video communication on traditional video conference, but also it can complete the implementation of shared data (various documents and software) and cooperative office. Because of the change of routing and multipart talking simultaneously in the network conference telephone, the technical standard of the International Telecommunication Union (ITU-T G.165/G.168) is established according to SDH and PDH, the algorithms available for echo cancellation can’t meet the demand when they are applied to network conference telephone.
According the digital audio watermarking technology of information hiding technique, this paper applies it to the echo cancellation of network conference telephone. The pasedo-random sequence with low power spectrum density is embedded into audio signal which is in the sender, and demodulated in the receiver, and then the network delays of the audio packet is measured, the echo is eliminated. During the multipart talking, with different PN sequences in every user’s audio signal and the theory of CDMA (Code Division Multiple Access), which one is talking can be identified, and then the echo can be eliminated.
Through theoretical analysis and computer simulation, the research provides a feasible technical method, it can solve the personal identification of echo cancellation during multipart talking in conference telephone, and the different network delays during the segmentation of audio packets, modify the problem of convergent rate in traditional adaptive algorithms. Owing to the application of the digital audio watermarking technology, it is hardly to influence the talking quality that embedding the PN sequence.
The integrated scheme about echo cancellation of digital network conference telephone which applies the technology remains to further research, it is necessary to investigate the technical indexes and testing methods which are included in ITU-T G.165 and G.168 about digital network conference telephone.
引文
[1]杨雯静.基于DSP的会议电话语音改善技术的研究与实现.[华东师范大学硕士学位论文],2006.1-2
[2]胡黄水.NP-23数字电话会议系统研究与开发.[吉林大学硕士学位论文],2005.1-2
[3] Paulo S.R.Diniz. Adaptive Filtering:Algorithms and Practical Implementation. Second Edition. Publishing House of Electronics Industry,2004. 35-155
[4] J. C. Jenq,S. F. Hsieh. Acoustic Echo Cancellation Using Iterative Maximal Length Correlation and Double-Talk Detection. IEEE Transactions on Speech and Audio Processing. 2001,9(8):932-942
[5] Hawwa S. Audio mixing for centralized Conferences in a SIP Environment. ICME’02 Proceedings. 2002.(02):269-272
[6] Yuan Zhang. SIP based VOIP Network and its Interworking with the PSTN. Journal of Electronics and Communication Engineering. 2002.14(6):273-282
[7]李茂全,徐正全,胡瑞敏.用于回波抵消的最大长度序列相关近端语音检测算法研究.计算机与数字工程. 2002. 30(04):8-14
[8]崔丽珍,杜普选.基于DSP的回波抵消技术.包头钢铁学院学报.2005.24(02):168-170
[9]袁江南.自适应回波抵消器的算法与MATLAB仿真.厦门科技.2003.24(03):53-55
[10]谢智波.自适应回波抵消器的算法分析与比较.浙江万里学院学报.2007.20(05):39-42
[11]桑林,郝建军,刘丹谱.数字通信.北京:北京邮电大学出版社,2002.14-15
[12]何振亚.自适应信号处理.北京:科学出版社,2002.22-30
[13] Ang, W.P.and B.Farhang Boroujeny. A New Class of Gradient AdaptiveStep Size LMS Algorithms. IEEE Transactions on Signal Processing. 2001.49(4):805-810
[14]龚耀寰.自适应滤波.第二版.北京:电子工业出版社,2003. 36-50,89-95
[15]杨海燕,黄辉,张雄伟.一种改进的自适应回波抵消器的设计与实现.电子质量.2006.(12):17-19
[16] Simon Haykin.自适应滤波原理.郑宝玉.第四版.北京:电子工业出版社,2003.23-90
[17] C.Y.Chang and K.K.Shyu. Active Noise Cancellation with A Fuzzy Adaptive Filtered-x Algorithm. IEE Proc. Circuits Devices Syst. 2003.150(5):416-422
[18]耿洁,陈家福.自适应算法LMS和RLS的仿真分析.山西电子技术.2007.(05):78-79
[19]陈海兰,胡晓毅,许茹,刘慧,蔡彦.LMS算法和RLS算法在水声信道通信系统中应用的比较.现代电子技术.2007.(03):18-19
[20]王炳锡,彭天强.信息隐藏技术.北京:国防工业出版社,2007.128-133
[21] KIM.H.J, CHOI.Y.H. A novel echo-hiding scheme with backward and forward kernels. Circuits and Systems for Video Technology. 2003.13:885-889
[22]曾兴雯,刘乃安,孙献璞.扩展频谱通信及其多址技术.西安:西安电子科技大学出版社,2004. 217-220
[23]王月清,柴远波,吴桂生.宽带CDMA移动通信原理.北京:电子工业出版社,2001.104-128
[24]樊昌信,张甫翊,徐炳祥,吴成柯.通信原理.第五版.北京:国防工业出版社,2004. 326-366
[25]高凯,王世练,张尔扬.基于FPGA的伪码测距电路的设计与实现.电子技术应用.2002.28(06):44-45
[26]何世彪,谭晓衡.扩频技术及其实现.2007.204-205
[27]及燕丽,王友村,沈其聪等.现代通信系统.北京:电子工业出版社,2001.247-254
[28]孙屹,吴磊.Simulink通信仿真开发手册.2004.31-380
[29] Darko Kirovski, Henrique Malvar. Robust Covert Communication over a Public Audio Channel Using Spread Spectrum. IEEE Journal on Select Ares in Communications. 2003.16(4):561-572
[30] Sviatoslav Voloshynovskiy, Oleksiy Kovall, Thierry Pun. Information-Theoretic Data Hiding for Public Network Security, Services Control and Secure Communications. TELSIKS. 2003.21(7):1-14
[31]岳军巧,钮心忻,杨义先.语音保密通信中的信息隐藏技术.北京邮电大学学报.2002.25(01):79-83
[32]戴跃伟,茅耀斌,王执铨.音频信息隐藏技术及其发展方向.信息与控制.2001.30(05):385-391
[33]戴跃伟,杨洋,茅耀斌,王执铨.变换域音频水印技术初步研究.东南大学学报.2000.30(05):22-27
[34]汪小帆,戴跃伟,茅耀斌.信息隐藏技术.北京:机械工业出版社,2001. 156-164
[35]伯晓晨,李涛,刘路.Matlab工具箱应用指南——信息工程篇.北京:电子工业出版社,2000.256-259
[36]曾兴雯,刘乃安,陈健.高频电路原理与分析.第三版.西安:西安电子科技大学出版社,2002.281-297
[37]周伯扬,耿淑芬.CDMA网络技术.北京:国防工业出版社,2006.86-90
[38] Zhaohong Zhang,Gunter Schmer. Analysis of Filter Coefficient Precision on LMS Algorithm Performance for G.165/G.168 Echo Cancellation. Texas Instruments. 2000:1-8
[39]陈桂明,张明照,戚红雨,张宝俊等.应用Matlab建模与仿真.北京:科学出版社,2001.17-85
[2]胡黄水.NP-23数字电话会议系统研究与开发.[吉林大学硕士学位论文],2005.1-2
[3] Paulo S.R.Diniz. Adaptive Filtering:Algorithms and Practical Implementation. Second Edition. Publishing House of Electronics Industry,2004. 35-155
[4] J. C. Jenq,S. F. Hsieh. Acoustic Echo Cancellation Using Iterative Maximal Length Correlation and Double-Talk Detection. IEEE Transactions on Speech and Audio Processing. 2001,9(8):932-942
[5] Hawwa S. Audio mixing for centralized Conferences in a SIP Environment. ICME’02 Proceedings. 2002.(02):269-272
[6] Yuan Zhang. SIP based VOIP Network and its Interworking with the PSTN. Journal of Electronics and Communication Engineering. 2002.14(6):273-282
[7]李茂全,徐正全,胡瑞敏.用于回波抵消的最大长度序列相关近端语音检测算法研究.计算机与数字工程. 2002. 30(04):8-14
[8]崔丽珍,杜普选.基于DSP的回波抵消技术.包头钢铁学院学报.2005.24(02):168-170
[9]袁江南.自适应回波抵消器的算法与MATLAB仿真.厦门科技.2003.24(03):53-55
[10]谢智波.自适应回波抵消器的算法分析与比较.浙江万里学院学报.2007.20(05):39-42
[11]桑林,郝建军,刘丹谱.数字通信.北京:北京邮电大学出版社,2002.14-15
[12]何振亚.自适应信号处理.北京:科学出版社,2002.22-30
[13] Ang, W.P.and B.Farhang Boroujeny. A New Class of Gradient AdaptiveStep Size LMS Algorithms. IEEE Transactions on Signal Processing. 2001.49(4):805-810
[14]龚耀寰.自适应滤波.第二版.北京:电子工业出版社,2003. 36-50,89-95
[15]杨海燕,黄辉,张雄伟.一种改进的自适应回波抵消器的设计与实现.电子质量.2006.(12):17-19
[16] Simon Haykin.自适应滤波原理.郑宝玉.第四版.北京:电子工业出版社,2003.23-90
[17] C.Y.Chang and K.K.Shyu. Active Noise Cancellation with A Fuzzy Adaptive Filtered-x Algorithm. IEE Proc. Circuits Devices Syst. 2003.150(5):416-422
[18]耿洁,陈家福.自适应算法LMS和RLS的仿真分析.山西电子技术.2007.(05):78-79
[19]陈海兰,胡晓毅,许茹,刘慧,蔡彦.LMS算法和RLS算法在水声信道通信系统中应用的比较.现代电子技术.2007.(03):18-19
[20]王炳锡,彭天强.信息隐藏技术.北京:国防工业出版社,2007.128-133
[21] KIM.H.J, CHOI.Y.H. A novel echo-hiding scheme with backward and forward kernels. Circuits and Systems for Video Technology. 2003.13:885-889
[22]曾兴雯,刘乃安,孙献璞.扩展频谱通信及其多址技术.西安:西安电子科技大学出版社,2004. 217-220
[23]王月清,柴远波,吴桂生.宽带CDMA移动通信原理.北京:电子工业出版社,2001.104-128
[24]樊昌信,张甫翊,徐炳祥,吴成柯.通信原理.第五版.北京:国防工业出版社,2004. 326-366
[25]高凯,王世练,张尔扬.基于FPGA的伪码测距电路的设计与实现.电子技术应用.2002.28(06):44-45
[26]何世彪,谭晓衡.扩频技术及其实现.2007.204-205
[27]及燕丽,王友村,沈其聪等.现代通信系统.北京:电子工业出版社,2001.247-254
[28]孙屹,吴磊.Simulink通信仿真开发手册.2004.31-380
[29] Darko Kirovski, Henrique Malvar. Robust Covert Communication over a Public Audio Channel Using Spread Spectrum. IEEE Journal on Select Ares in Communications. 2003.16(4):561-572
[30] Sviatoslav Voloshynovskiy, Oleksiy Kovall, Thierry Pun. Information-Theoretic Data Hiding for Public Network Security, Services Control and Secure Communications. TELSIKS. 2003.21(7):1-14
[31]岳军巧,钮心忻,杨义先.语音保密通信中的信息隐藏技术.北京邮电大学学报.2002.25(01):79-83
[32]戴跃伟,茅耀斌,王执铨.音频信息隐藏技术及其发展方向.信息与控制.2001.30(05):385-391
[33]戴跃伟,杨洋,茅耀斌,王执铨.变换域音频水印技术初步研究.东南大学学报.2000.30(05):22-27
[34]汪小帆,戴跃伟,茅耀斌.信息隐藏技术.北京:机械工业出版社,2001. 156-164
[35]伯晓晨,李涛,刘路.Matlab工具箱应用指南——信息工程篇.北京:电子工业出版社,2000.256-259
[36]曾兴雯,刘乃安,陈健.高频电路原理与分析.第三版.西安:西安电子科技大学出版社,2002.281-297
[37]周伯扬,耿淑芬.CDMA网络技术.北京:国防工业出版社,2006.86-90
[38] Zhaohong Zhang,Gunter Schmer. Analysis of Filter Coefficient Precision on LMS Algorithm Performance for G.165/G.168 Echo Cancellation. Texas Instruments. 2000:1-8
[39]陈桂明,张明照,戚红雨,张宝俊等.应用Matlab建模与仿真.北京:科学出版社,2001.17-85