G729B编解码与回声消除在VoIP网关中的应用
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摘要
随着IP(Internet Protocol)技术的不断发展,能够降低通信成本的VoIP(Voice over Internet Protocol)技术得到了广泛的应用。VoIP的一个关键问题是如何在有限的带宽上实时地传输高质量的语音。本文就此问题研究了VoIP电话中的两个核心技术:语音编解码和回声消除。
ITU-T G.729语音压缩编解码标准是目前应用最广泛的VoIP语音数字信号处理标准之一。G.729B标准在G.729的基础上增加了静音检测技术(Voice Active Detection,简称VAD)。本文首先阐述了G.729B的算法结构,重点分析了CS-ACELP(Conjugate Structure-Algebraic Code Excited Linear Prediction)算法中的线性预测分析、线性预测系数与线谱对系数的转换、基本延迟分析、自适应码本和固定码本的搜索、VAD技术等。
其次,设计了基于双端通话检测的自适应回声消除系统,该系统根据双端通话的状态来控制NLMS自适应滤波器系数的调整。通过对输入信号X N( n )进行区域划分,同时结合G.729B编码器的静音检测结果,提出了一种改进的Geigel算法,使得Geigel算法的检测结果更加稳定。
最后,开发出了大容量的语音处理板。该板以TMS320C64X DSP为主要器件来实现G.729B编解码与回声消除。硬件方面主要包括DSP小阵列系统的电源、时钟、外扩SDRAM、EMIFB、McBSP和HPI口的设计。软件方面,分别从C语言级、文件级、汇编级实现回声消除与G.729B代码的编写与优化。
实验结果表明,优化后的G.729B程序在内存占用和运算复杂度方面都达到了预期目标,语音信号解码之后失真很小;改进的Geigel检测技术使得回声消除器对电话语音回声消除效果更加明显,有效地提高了语音的质量。
With the development of IP (Internet Protocol) technology, the VoIP (Voice over Internet protocol) technology which can reduce the cost of communication has been widely applied all over the world. One of the key problems of VoIP is how to transmit the high quality speech signal in the limited bandwidth. For solving the problem, this paper studies and does research on two core technologies of VoIP: speech codec and echo cancellation.
ITU-T G.729 is one of the most efficiently and broadly used standards that can process speech signal. Based on G.729, G.729B presents voice active detection (VAD) technology. Firstly, this paper introduces the algorithm structure of G.729B, focusing on the LP analysis, transformation of LP to LSP coefficients, pitch detection, searching of adaptive and fixed codebooks, VAD technology, etc.
Secondly, the design of Echo Cancellation system based on double talk detection is presented. The coefficients of NLMS adaptive filter are adjusted according to the double talk status. This paper presents an improved Geigel algorithm by dividing the input signal X N( n ). The improved algorithm is more stable than normal Geigel algorithm. Combined with G.729B coder, the improved algorithm uses the results of VAD of past frames to improve the effect of echo cancellation.
Finally, the design and realization of the high capacity Voice Trans Codec Board are described. This board uses TMS320C64X DSP as the major apparatus to apply G.729B codec and echo cancellation. In the aspect of hardware, the device of Power, Clock, outer-expanded SDRAM, EMIFB, McBSP and HPI of DSP are designed. In the aspect of software, echo cancellation and G.729B codes are programmed and optimized with multiple methods.
The experimental results show that the optimized programs have reached the expectation in data storage and operational complexity. The distortion of the decoded speech is rather small. With the improved Geigel algorithm, the echo canceller can do better than classical Geigel algorithm and improve the quality of the speech.
ITU-T G.729语音压缩编解码标准是目前应用最广泛的VoIP语音数字信号处理标准之一。G.729B标准在G.729的基础上增加了静音检测技术(Voice Active Detection,简称VAD)。本文首先阐述了G.729B的算法结构,重点分析了CS-ACELP(Conjugate Structure-Algebraic Code Excited Linear Prediction)算法中的线性预测分析、线性预测系数与线谱对系数的转换、基本延迟分析、自适应码本和固定码本的搜索、VAD技术等。
其次,设计了基于双端通话检测的自适应回声消除系统,该系统根据双端通话的状态来控制NLMS自适应滤波器系数的调整。通过对输入信号X N( n )进行区域划分,同时结合G.729B编码器的静音检测结果,提出了一种改进的Geigel算法,使得Geigel算法的检测结果更加稳定。
最后,开发出了大容量的语音处理板。该板以TMS320C64X DSP为主要器件来实现G.729B编解码与回声消除。硬件方面主要包括DSP小阵列系统的电源、时钟、外扩SDRAM、EMIFB、McBSP和HPI口的设计。软件方面,分别从C语言级、文件级、汇编级实现回声消除与G.729B代码的编写与优化。
实验结果表明,优化后的G.729B程序在内存占用和运算复杂度方面都达到了预期目标,语音信号解码之后失真很小;改进的Geigel检测技术使得回声消除器对电话语音回声消除效果更加明显,有效地提高了语音的质量。
With the development of IP (Internet Protocol) technology, the VoIP (Voice over Internet protocol) technology which can reduce the cost of communication has been widely applied all over the world. One of the key problems of VoIP is how to transmit the high quality speech signal in the limited bandwidth. For solving the problem, this paper studies and does research on two core technologies of VoIP: speech codec and echo cancellation.
ITU-T G.729 is one of the most efficiently and broadly used standards that can process speech signal. Based on G.729, G.729B presents voice active detection (VAD) technology. Firstly, this paper introduces the algorithm structure of G.729B, focusing on the LP analysis, transformation of LP to LSP coefficients, pitch detection, searching of adaptive and fixed codebooks, VAD technology, etc.
Secondly, the design of Echo Cancellation system based on double talk detection is presented. The coefficients of NLMS adaptive filter are adjusted according to the double talk status. This paper presents an improved Geigel algorithm by dividing the input signal X N( n ). The improved algorithm is more stable than normal Geigel algorithm. Combined with G.729B coder, the improved algorithm uses the results of VAD of past frames to improve the effect of echo cancellation.
Finally, the design and realization of the high capacity Voice Trans Codec Board are described. This board uses TMS320C64X DSP as the major apparatus to apply G.729B codec and echo cancellation. In the aspect of hardware, the device of Power, Clock, outer-expanded SDRAM, EMIFB, McBSP and HPI of DSP are designed. In the aspect of software, echo cancellation and G.729B codes are programmed and optimized with multiple methods.
The experimental results show that the optimized programs have reached the expectation in data storage and operational complexity. The distortion of the decoded speech is rather small. With the improved Geigel algorithm, the echo canceller can do better than classical Geigel algorithm and improve the quality of the speech.
引文
[1] 赵慧玲,NGN和较交换,中兴通信技术,2005,11(3):30~34
[2] ITU-T, G.729,Coding of Speech at 8kb/s Using Conjugate Structure Algebraic-Code-Excited Linear-Prediction(CS-ACELP),1996
[3] R.D.Gitlin, S.D.Weinstein. On the design of gradient algorithms for digitally implemented adaptive filters.IEEE Trans on CT,1973,2:125~136
[4] H.Yasukawa, S.Shimada, I.Furukrawa, et al, Acoustic echo canceller with high speech quality, ICASSP, 1987: 2125~2128
[5] R.D.Gitlin, S.D.Weinstein. The effects of large interference on the tracking capability of digitally implemented echo cancellers,IEEE Trans on COM, 1978, 6:833~839
[6] 叶华,吴伯修,变步长自适应滤波算法的研究,电子学报,1990,18(4):63~69
[7] 高鹰,谢胜利,一种变步长LMS自适应滤波算法及分析,电子学报,2001,29(8):1094~1097
[8] E.Nurgun, S.Filiz, Wavelet transform based adaptive filter: analysis and new results,IEEE Trans on SP, 1996, 44(9):2163~2168
[9] J.H.Cho,D.R.Morgan,J.Benesty,An Objective Technique for Evaluating Doubletalk Detectors in Acoustic Echo Cancelers,IEEE Transaction On Speech And Audio Processing,1999,7(6):718~724
[10] T.Gansler,M Hansson,C.J.Ivarsson,A doubletalk detectors based on coherence,IEEE Trans on Commun,1996,44(11):1421~1427
[11] J.Benesty,R.M.Dennis,H.C.Jun,A New Class of Doubletalk Detectors Based on Cross-Correlation,IEEE Trans On Speech And Audio Processing,2000,8(2):168~172
[12] 蒙淑艳,自适应回声抵消和噪声消除算法的研究,[硕士学位论文],吉林长春,吉林大学,2004
[13] 罗夙,G.729语音编码标准及其应用,电声技术,2002,3:11~12
[14] ITU-T, G.729 Annex B,A Silence Compression Scheme for G.729 Optimized for Terminals Conforming to ITU-T V.70,1996
[15] R.Salami,C.Laflamme,J.-P.Adoul,et al,A toll quality 8kb/s speech codec for the Personal Communications Systems (PCS). IEEE Trans on Vehicular Technology,1994,43:808~816
[16] W.C.Chu,Window Optimization in Linear Prediction Analysis,IEEE Trans on Speech and Audio Processing,2003,11(6):626~635
[17] W.C.Chu,T.Miki,Optimization of Window and LSF Interpolation Factor for the ITU-T G.729 Speech Coding Standard,Euro speech Geneva Switzerland,2003:1061~1064。
[18] P.Alku,T.Backstrom,All-Pole Modeling Technique Based on the Weighted Sum of the LSP Polynomials, IEEE International Conference on Acoustics、Speech and Signal Processing,(ICASSP),2002,1:665~668
[19] A.Magrath,M.Sandler,Parametric Encoding of Line Spectral Pairs for Wideband Speech Coding,Applied Signal Processing,1998,5(4):255~269
[20] A.Kataoka,J.Ikedo,S.Hayashi,LSP and Gain Quantization for the Proposed ITU-T 8kb/s Speech Coding Standard,IEEE Speech Coding for Telecommunications Proceedings,1995,9:7~8
[21] 杨俊杰,黎福海,G.729 Annex B 简介,电声技术,2002,9:10~12
[22] 郑洪英,郭东辉,黄国和,等,基于静音检测的ITU-T G.729算法,厦门大学学报,2002,41(4):431~434
[23] ITU-T, G.165,Echo cancellers,1993
[24] ITU-T, G.168,Digital network echo cancellers,2000
[25] M.M.Sondhi,D.A.Berkley,Silencing echo on the telephone network,Proceedings of the IEEE,1980,68(8):948~963
[26] J.D.Gordy,R.A.Goubran,A Low_Complexity doubletalk Detector for Acoustic echo cancellers in packet-based telephony,2005 IEEE Workshop on Application of Signal Processing to Audio and Acoustics,2005,16(19):74~77
[27] 蒋春晖,声学回声抵消算法及其实现,[硕士学位论文],江苏南京,东南大学,2005。
[28] 胡啸,WLAN上VoIP中的自适应回声抵消技术研究,[博士学位论文],江苏南京,东南大学,2004
[29] C.Breining,P.Dreiseitel,E.Hansler,et al,Acoustic Echo Control:An Application of Very-High-Order Adaptive Filters,IEEE Signal ProcessingMagazine,1999,16:42~69
[30] S.HayKin,自适应滤波器原理(第三版),北京,电子工业出版社,1998:183~188
[31] D T M Slock. On the convergence behavior of the LMS and the normalized LMS algorithms,IEEE Trans on Signal Processing,1993,41(9):2811~2825
[32] 吴光弼,祝琳瑜,一种变步长LMS自适应算法,电子学报,1994,22(1):55~66
[33] 黄本雄,王进军,用快速自适应和相关度检测算法的回声抵消器,华中科技大学学报(自然科学版),2004,32(6):4~6
[34] D.L.Duttweiler,A twelve-channel digital echo canceller,IEEE trans on Commun, 1998,26(5):647~653
[35] 曹秀英,双向通话检测器,通信技术,1997,2:45~50
[36] YD/T 1292-2003,基于H.248的媒体网关控制协议技术要求,2003
[37] 郭凯,H.248在软交换中的应用,[硕士学位论文],江苏南京,南京邮电学院,2005
[38] 尹勇,欧光军,关荣锋,DSP集成开发环境CCS开发指南,北京,北京航空航天大学出版社,2003:8~15
[39] 李方慧,王飞,何佩琨等,TMS320C6000系列DSPs原理与应用,北京,电子工业出版社,2003:197~198
[40] 戴明祯,周建江,TMS320C54x DSP、结构、原理及应用,北京,北京航空航天大学出版社,2001:98~103
[41] 醴勇,麻晓园,TMS320C6X DSP的C语言与汇编混合编程技术,计算机测量与控制,2003,11(9):716~718
[42] 余胜生,彭智,周敬利,在TMS320C6x系列DSP上实现G.729算法,计算机工程与科学,2004,26(4):51~54
[2] ITU-T, G.729,Coding of Speech at 8kb/s Using Conjugate Structure Algebraic-Code-Excited Linear-Prediction(CS-ACELP),1996
[3] R.D.Gitlin, S.D.Weinstein. On the design of gradient algorithms for digitally implemented adaptive filters.IEEE Trans on CT,1973,2:125~136
[4] H.Yasukawa, S.Shimada, I.Furukrawa, et al, Acoustic echo canceller with high speech quality, ICASSP, 1987: 2125~2128
[5] R.D.Gitlin, S.D.Weinstein. The effects of large interference on the tracking capability of digitally implemented echo cancellers,IEEE Trans on COM, 1978, 6:833~839
[6] 叶华,吴伯修,变步长自适应滤波算法的研究,电子学报,1990,18(4):63~69
[7] 高鹰,谢胜利,一种变步长LMS自适应滤波算法及分析,电子学报,2001,29(8):1094~1097
[8] E.Nurgun, S.Filiz, Wavelet transform based adaptive filter: analysis and new results,IEEE Trans on SP, 1996, 44(9):2163~2168
[9] J.H.Cho,D.R.Morgan,J.Benesty,An Objective Technique for Evaluating Doubletalk Detectors in Acoustic Echo Cancelers,IEEE Transaction On Speech And Audio Processing,1999,7(6):718~724
[10] T.Gansler,M Hansson,C.J.Ivarsson,A doubletalk detectors based on coherence,IEEE Trans on Commun,1996,44(11):1421~1427
[11] J.Benesty,R.M.Dennis,H.C.Jun,A New Class of Doubletalk Detectors Based on Cross-Correlation,IEEE Trans On Speech And Audio Processing,2000,8(2):168~172
[12] 蒙淑艳,自适应回声抵消和噪声消除算法的研究,[硕士学位论文],吉林长春,吉林大学,2004
[13] 罗夙,G.729语音编码标准及其应用,电声技术,2002,3:11~12
[14] ITU-T, G.729 Annex B,A Silence Compression Scheme for G.729 Optimized for Terminals Conforming to ITU-T V.70,1996
[15] R.Salami,C.Laflamme,J.-P.Adoul,et al,A toll quality 8kb/s speech codec for the Personal Communications Systems (PCS). IEEE Trans on Vehicular Technology,1994,43:808~816
[16] W.C.Chu,Window Optimization in Linear Prediction Analysis,IEEE Trans on Speech and Audio Processing,2003,11(6):626~635
[17] W.C.Chu,T.Miki,Optimization of Window and LSF Interpolation Factor for the ITU-T G.729 Speech Coding Standard,Euro speech Geneva Switzerland,2003:1061~1064。
[18] P.Alku,T.Backstrom,All-Pole Modeling Technique Based on the Weighted Sum of the LSP Polynomials, IEEE International Conference on Acoustics、Speech and Signal Processing,(ICASSP),2002,1:665~668
[19] A.Magrath,M.Sandler,Parametric Encoding of Line Spectral Pairs for Wideband Speech Coding,Applied Signal Processing,1998,5(4):255~269
[20] A.Kataoka,J.Ikedo,S.Hayashi,LSP and Gain Quantization for the Proposed ITU-T 8kb/s Speech Coding Standard,IEEE Speech Coding for Telecommunications Proceedings,1995,9:7~8
[21] 杨俊杰,黎福海,G.729 Annex B 简介,电声技术,2002,9:10~12
[22] 郑洪英,郭东辉,黄国和,等,基于静音检测的ITU-T G.729算法,厦门大学学报,2002,41(4):431~434
[23] ITU-T, G.165,Echo cancellers,1993
[24] ITU-T, G.168,Digital network echo cancellers,2000
[25] M.M.Sondhi,D.A.Berkley,Silencing echo on the telephone network,Proceedings of the IEEE,1980,68(8):948~963
[26] J.D.Gordy,R.A.Goubran,A Low_Complexity doubletalk Detector for Acoustic echo cancellers in packet-based telephony,2005 IEEE Workshop on Application of Signal Processing to Audio and Acoustics,2005,16(19):74~77
[27] 蒋春晖,声学回声抵消算法及其实现,[硕士学位论文],江苏南京,东南大学,2005。
[28] 胡啸,WLAN上VoIP中的自适应回声抵消技术研究,[博士学位论文],江苏南京,东南大学,2004
[29] C.Breining,P.Dreiseitel,E.Hansler,et al,Acoustic Echo Control:An Application of Very-High-Order Adaptive Filters,IEEE Signal ProcessingMagazine,1999,16:42~69
[30] S.HayKin,自适应滤波器原理(第三版),北京,电子工业出版社,1998:183~188
[31] D T M Slock. On the convergence behavior of the LMS and the normalized LMS algorithms,IEEE Trans on Signal Processing,1993,41(9):2811~2825
[32] 吴光弼,祝琳瑜,一种变步长LMS自适应算法,电子学报,1994,22(1):55~66
[33] 黄本雄,王进军,用快速自适应和相关度检测算法的回声抵消器,华中科技大学学报(自然科学版),2004,32(6):4~6
[34] D.L.Duttweiler,A twelve-channel digital echo canceller,IEEE trans on Commun, 1998,26(5):647~653
[35] 曹秀英,双向通话检测器,通信技术,1997,2:45~50
[36] YD/T 1292-2003,基于H.248的媒体网关控制协议技术要求,2003
[37] 郭凯,H.248在软交换中的应用,[硕士学位论文],江苏南京,南京邮电学院,2005
[38] 尹勇,欧光军,关荣锋,DSP集成开发环境CCS开发指南,北京,北京航空航天大学出版社,2003:8~15
[39] 李方慧,王飞,何佩琨等,TMS320C6000系列DSPs原理与应用,北京,电子工业出版社,2003:197~198
[40] 戴明祯,周建江,TMS320C54x DSP、结构、原理及应用,北京,北京航空航天大学出版社,2001:98~103
[41] 醴勇,麻晓园,TMS320C6X DSP的C语言与汇编混合编程技术,计算机测量与控制,2003,11(9):716~718
[42] 余胜生,彭智,周敬利,在TMS320C6x系列DSP上实现G.729算法,计算机工程与科学,2004,26(4):51~54