自适应回波抵消器和语音编解码器的设计实现
摘要
本文的课题来自通信学院140教研室的预研课题——IP电话系统信号处理关键技术研究。主要工作是设计并实现电话线路自适应回波抵消器和基于多带激励模型的语音编解码器。
对于IP电话系统而言，由于采用了分组交换技术，由2/4线转换器阻抗不匹配而产生的回波在系统中传输的时延较长，这将加重回波对语音通信的影响。所以，自适应回波抵消技术是IP电话系统不可或缺的技术之一。本文采用TMS320C5402 DSP实现了基于NLMS 算法和Geigle算法的回波抵消器。为了提高性能，本文引入了NLMS算法的扩展精度实现和Geigle 算法的高效实现。测试结果表明，回波抵消器的性能超出或达到了G.165建议标准的要求。
语音压缩编码技术对提高IP电话系统的系统容量有重要的意义。基于多带激励模型的语音压缩编码算法以其较好的合成语音质量而受到广泛的重视。本文依据多带激励模型的定义，完成了编解码器算法设计，并用C语言实现。本文引入了一种新的基音周期计算方法，静音帧判决算法，清音帧判决算法，清浊音信息接收端重建等新算法，提高了合成语音的质量，降低了算法的总计算量。非正式编解码测试表明，本文的声码器能在2.4kbps的编码速率上合成出较好的语音。
This work is supported by the pre-research project being conducted in 140 lab: " Researches on Key Signal Processing Technologies in IP Telephone Systems". And the main focus is the design, implementation and testing of an adaptive line echo canceller and a vocoder based on Multi-band Excitation (MBE) model.
Because IP telephone systems employ the packet switching technology, the echo signals generated by the Hybrid due to the impedance mismatch usually experience longer transmitting delay, which significantly decrease the speech communication quality. Therefore, echo cancellers are indispensable devices for any IP telephone system. In this paper, an echo canceller based on NLMS algorithm and Geigle algorithm is implemented on TMS320C5402 Digital Signal Processor (DSP). In order to improve its performance, extended precision techniques and a high-efficiency implementation scheme of Geigle algorithm are employed. The testing results show that the performance of the echo canceller exceeds the requirements specified by ITU recommendation G.165.
Speech compression technologies are of great importance in increasing the capacities of IP telephone systems. Vocoders based on the MBE model are popular due to their good synthesis speech quality. In this paper, a vocoder based on the MBE model is designed, its algorithms are detailed and it is implemented using C programming language. And a new pitch extraction algorithm, an active/inactive frame decision algorithm and a voiced/unvoiced frame decision algorithm are developed with the aims to improve the quality of the vocoder and reduce its overall computation load. The testing results show that the vocoder can synthesize high-quality speech when transmission rate is set to be 2.4kbps.
对于IP电话系统而言，由于采用了分组交换技术，由2/4线转换器阻抗不匹配而产生的回波在系统中传输的时延较长，这将加重回波对语音通信的影响。所以，自适应回波抵消技术是IP电话系统不可或缺的技术之一。本文采用TMS320C5402 DSP实现了基于NLMS 算法和Geigle算法的回波抵消器。为了提高性能，本文引入了NLMS算法的扩展精度实现和Geigle 算法的高效实现。测试结果表明，回波抵消器的性能超出或达到了G.165建议标准的要求。
语音压缩编码技术对提高IP电话系统的系统容量有重要的意义。基于多带激励模型的语音压缩编码算法以其较好的合成语音质量而受到广泛的重视。本文依据多带激励模型的定义，完成了编解码器算法设计，并用C语言实现。本文引入了一种新的基音周期计算方法，静音帧判决算法，清音帧判决算法，清浊音信息接收端重建等新算法，提高了合成语音的质量，降低了算法的总计算量。非正式编解码测试表明，本文的声码器能在2.4kbps的编码速率上合成出较好的语音。
This work is supported by the pre-research project being conducted in 140 lab: " Researches on Key Signal Processing Technologies in IP Telephone Systems". And the main focus is the design, implementation and testing of an adaptive line echo canceller and a vocoder based on Multi-band Excitation (MBE) model.
Because IP telephone systems employ the packet switching technology, the echo signals generated by the Hybrid due to the impedance mismatch usually experience longer transmitting delay, which significantly decrease the speech communication quality. Therefore, echo cancellers are indispensable devices for any IP telephone system. In this paper, an echo canceller based on NLMS algorithm and Geigle algorithm is implemented on TMS320C5402 Digital Signal Processor (DSP). In order to improve its performance, extended precision techniques and a high-efficiency implementation scheme of Geigle algorithm are employed. The testing results show that the performance of the echo canceller exceeds the requirements specified by ITU recommendation G.165.
Speech compression technologies are of great importance in increasing the capacities of IP telephone systems. Vocoders based on the MBE model are popular due to their good synthesis speech quality. In this paper, a vocoder based on the MBE model is designed, its algorithms are detailed and it is implemented using C programming language. And a new pitch extraction algorithm, an active/inactive frame decision algorithm and a voiced/unvoiced frame decision algorithm are developed with the aims to improve the quality of the vocoder and reduce its overall computation load. The testing results show that the vocoder can synthesize high-quality speech when transmission rate is set to be 2.4kbps.
引文
[1]. "Definitions and Terms", http://www.iec.org/online/tutorials/echo_cancel/index.html
[2]. Sophocles J. Orfanidis,“Introduction to Signal Processing”，清华大学出版社，1998
[3]. 覃景繁，“电话会议系统中的回波及其消除技术”，通信技术，第2期，1998
[4]. Petzold. "Elementare Raumakustik", Berlin 1967, pp 8.
[5]. H.Decker, "A Delay Conduit for Measuring and Demonstrating transit time effects in telecommunication systems", Elek. Nachr. Tech. Volume 8,1965
[6]. 方文郁，“卫星通信中的回波抑制与回波抵消”，第11卷，第9期，1995
[7]. Geneva, ITU Recommendation, G.165, 1993
[8]. 韦刚等，“回波抵消理论进展及其应用”，电路与系统学报，第三卷，第三期，1998
[9]. Kazuo Murano, et. al, “Echo Cancellation and Applications”，IEEE Communication Magazine, January, 1990
[10]. 沈福民，“自适应信号处理”，西安电子科技大学出版社，2001
[11]. 孙志辉等，“用TMS320C25实现回波消除”，数据通信，第一期，1996
[12]. H. Fan, and W.K.Jenkis, “An Investigation of an Adaptive IIR Echo Canceller: Advantages and Problems”，IEEE Transactions on ASSP, Vol. 36, No. 12, 1988
[13]. 孙炼等，“一种加白化预滤波的回波抵消算法”，南京邮电学院学报，第四期，1997
[14]. Gordon W. Davidson “Reduced Complexity Echo Cancellation Using Orthonormal Functions”，IEEE Transactions on Circuits and Systems, Vol. 38, No.1, January 1991
[15]. Simon Haykin, “Adaptive Filter Theory”, 3rd Edition, Prentice Hall, 1998
[16]. Chamzas.C.,“Adaptive Filtering and Periodic Inputs: Conditions for An Exponential Divergence”，ICECS, Volume 2, 13-16, 96
[17]. 曹秀英，“双向通话检测器”，通信技术，第二期，1997
[18]. D. L. Dutterweiler, “A twelve-channel digital echo canceller”，IEEE transactions on Communication, Vol. COMM-26, pp. 647-653, May 1978
[19]. H. Ye and B. X. Xu, “A new double-talk detection algorithm based on orthogonality theorem”, IEEE Transactions on Communication, Vol.39, pp. 1542-1545, Nov. 1991
[20]. J. Benesty, D. R. Morgan, and J. H. Cho, “A new class of double-talk detector based on cross-corelation”，IEEE Transactions on Speech and Audio Signal Processing, Vol. 8, pp 168-173, March 2000
[21]. J. H. Cho, D. R. Morgan and J. Benesty, “An objective technique for evaluating double talk detectors in acoustic echo cancellors”, IEEE Transactions on Speech Audio Processing, Vol. 7, pp.718-724, Nov. 1999
[22]. Texas Instruments, “Echo Cancellation Software for TMS320C54x”，BPRA054, 1997
[23]. Texas Instruments,“Extended Precision IIR Filter Design on the TMS320C54x DSP”，BPRA454, June, 1998
[24]. 彭启琮主编，“TMS320C54x实用教程”, 电子科技大学出版社，1999年7月
[25]. TI Instruments,“Code Composer Studio: Getting started Guide”，SPRU509, May, 2001
[26]. 刘福声等，“统计信号处理”，国防科技大学出版社，1999
[27]. 曹志刚，钱亚生，“现代通信原理”，清华大学出版社，1996
[28]. 彭启琮，李玉柏，管庆等，“DSP技术的发展与应用”，高等教育出版社，2002
[29]. 张雄伟，曹铁勇，“DSP芯片的原理与开发应用”，电子工业出版社，2000
[30]. 王沫然，“Matlab 6.0与科学计算”，电子工业出版社，2001
[30]. 苏金升（信息产业部电信管理局局长），“信产部：语音通信仍是现阶段主流”，http://www.cnc.ac.cn/news/tech/200211070010.html
[31]. Jayant, N. S. "Signal compression: Technology, Targets, and Research Directions," IEEE Journal on Selected Areas of Communications, Vol,10, No.5, pp. 769 - 815, June 1992
[32]. D.W.Griffin and J.S.Lim, "Multiband Excitation Vocoder", IEEE Transaction on ASSP, Vol. 36, pp1223-1235, August 1988
[33]. Tony Robinson, "A Tutorial to Speech coding", Intel Corporation Survey, 1998.
[34]. 王炳锡，“语音编码”，西安电子科技大学出版社，2002
[35]. 俞波，“基于多带激励模型的语音分析和语音合成”，电子科技大学硕士论文，1996年12月
[36]. 杨行俊，迟惠生等，“语音信号数字处理”，电子工业出版社，1995
[37]. George Box, Gwilym Jenkis,“时间序列分析：预测与控制”,中国统计出版社，2001
[38]. 胡广书，“数字信号处理——理论，算法与实现”，清华大学出版社，1996
[39]. P. Stoica, “谱估计导论”，电子科技大学出版社，2000
[40]. L.R.Rabiner， R.W.Schafer， “Digital Processing of Speech Signals”， Prentice Hall， 1979
[41]. 张贤达，“现代信号处理”，清华大学出版社，1996
[42]. 王田，“1.2Kb/s-2.4kb/s低码率语音变码技术研究”，清华大学博士论文，1996
[43]. F.Itakura，“Line Spectrum Representation of Linear Predictive Coefficients of Speech Signals”。J. Acoust. Soc. Am., 57, S35(a), 535, 1975
[44]. 张连海。 “多带激励语音编码算法研究及IMBE算法实现”，解放军信息工程学院硕士论文，2000
[45]. 张连海，王炳锡，“4.15kbs IMBE算法研究与实现”
[46]. 陆奕奕，“AMBE语音压缩算法实现和改进”，通信技术，第四期，1998
[47]. 李强，“基于多带激励模型的低速语音编码方案及计算机仿真”，电子科技大学硕士论文，1999
[48]. P.C. Meuse，“A 2400bps Multi-Band Excitation Vocoder”，Proc. IEEE. Int. Conf. Acoust. Speech Signal Processing, pp. 9-12, April, 1990
[49]. F. K. Soong and B. H. Juang，“Line Spectrum Pair (LSP)and Speech Data Compression”，Proc. ICASSP-84, pp. 1.10-1.10.4, 1984
[50]. Shan Lu and Marion Baraniecki，“A Robust 1200bps Voice Coder Based on Vector Quantization of Line Spectral Frequencies”，Proc. of ICASSP, pp. 522-524, 1997
[51]. J. Malhoul, et al，“Vector Quantization in Speech coding”，Proc. IEEE, Vol.73, No. 11, pp. 1551-1558, November, 1985
[2]. Sophocles J. Orfanidis,“Introduction to Signal Processing”，清华大学出版社，1998
[3]. 覃景繁，“电话会议系统中的回波及其消除技术”，通信技术，第2期，1998
[4]. Petzold. "Elementare Raumakustik", Berlin 1967, pp 8.
[5]. H.Decker, "A Delay Conduit for Measuring and Demonstrating transit time effects in telecommunication systems", Elek. Nachr. Tech. Volume 8,1965
[6]. 方文郁，“卫星通信中的回波抑制与回波抵消”，第11卷，第9期，1995
[7]. Geneva, ITU Recommendation, G.165, 1993
[8]. 韦刚等，“回波抵消理论进展及其应用”，电路与系统学报，第三卷，第三期，1998
[9]. Kazuo Murano, et. al, “Echo Cancellation and Applications”，IEEE Communication Magazine, January, 1990
[10]. 沈福民，“自适应信号处理”，西安电子科技大学出版社，2001
[11]. 孙志辉等，“用TMS320C25实现回波消除”，数据通信，第一期，1996
[12]. H. Fan, and W.K.Jenkis, “An Investigation of an Adaptive IIR Echo Canceller: Advantages and Problems”，IEEE Transactions on ASSP, Vol. 36, No. 12, 1988
[13]. 孙炼等，“一种加白化预滤波的回波抵消算法”，南京邮电学院学报，第四期，1997
[14]. Gordon W. Davidson “Reduced Complexity Echo Cancellation Using Orthonormal Functions”，IEEE Transactions on Circuits and Systems, Vol. 38, No.1, January 1991
[15]. Simon Haykin, “Adaptive Filter Theory”, 3rd Edition, Prentice Hall, 1998
[16]. Chamzas.C.,“Adaptive Filtering and Periodic Inputs: Conditions for An Exponential Divergence”，ICECS, Volume 2, 13-16, 96
[17]. 曹秀英，“双向通话检测器”，通信技术，第二期，1997
[18]. D. L. Dutterweiler, “A twelve-channel digital echo canceller”，IEEE transactions on Communication, Vol. COMM-26, pp. 647-653, May 1978
[19]. H. Ye and B. X. Xu, “A new double-talk detection algorithm based on orthogonality theorem”, IEEE Transactions on Communication, Vol.39, pp. 1542-1545, Nov. 1991
[20]. J. Benesty, D. R. Morgan, and J. H. Cho, “A new class of double-talk detector based on cross-corelation”，IEEE Transactions on Speech and Audio Signal Processing, Vol. 8, pp 168-173, March 2000
[21]. J. H. Cho, D. R. Morgan and J. Benesty, “An objective technique for evaluating double talk detectors in acoustic echo cancellors”, IEEE Transactions on Speech Audio Processing, Vol. 7, pp.718-724, Nov. 1999
[22]. Texas Instruments, “Echo Cancellation Software for TMS320C54x”，BPRA054, 1997
[23]. Texas Instruments,“Extended Precision IIR Filter Design on the TMS320C54x DSP”，BPRA454, June, 1998
[24]. 彭启琮主编，“TMS320C54x实用教程”, 电子科技大学出版社，1999年7月
[25]. TI Instruments,“Code Composer Studio: Getting started Guide”，SPRU509, May, 2001
[26]. 刘福声等，“统计信号处理”，国防科技大学出版社，1999
[27]. 曹志刚，钱亚生，“现代通信原理”，清华大学出版社，1996
[28]. 彭启琮，李玉柏，管庆等，“DSP技术的发展与应用”，高等教育出版社，2002
[29]. 张雄伟，曹铁勇，“DSP芯片的原理与开发应用”，电子工业出版社，2000
[30]. 王沫然，“Matlab 6.0与科学计算”，电子工业出版社，2001
[30]. 苏金升（信息产业部电信管理局局长），“信产部：语音通信仍是现阶段主流”，http://www.cnc.ac.cn/news/tech/200211070010.html
[31]. Jayant, N. S. "Signal compression: Technology, Targets, and Research Directions," IEEE Journal on Selected Areas of Communications, Vol,10, No.5, pp. 769 - 815, June 1992
[32]. D.W.Griffin and J.S.Lim, "Multiband Excitation Vocoder", IEEE Transaction on ASSP, Vol. 36, pp1223-1235, August 1988
[33]. Tony Robinson, "A Tutorial to Speech coding", Intel Corporation Survey, 1998.
[34]. 王炳锡，“语音编码”，西安电子科技大学出版社，2002
[35]. 俞波，“基于多带激励模型的语音分析和语音合成”，电子科技大学硕士论文，1996年12月
[36]. 杨行俊，迟惠生等，“语音信号数字处理”，电子工业出版社，1995
[37]. George Box, Gwilym Jenkis,“时间序列分析：预测与控制”,中国统计出版社，2001
[38]. 胡广书，“数字信号处理——理论，算法与实现”，清华大学出版社，1996
[39]. P. Stoica, “谱估计导论”，电子科技大学出版社，2000
[40]. L.R.Rabiner， R.W.Schafer， “Digital Processing of Speech Signals”， Prentice Hall， 1979
[41]. 张贤达，“现代信号处理”，清华大学出版社，1996
[42]. 王田，“1.2Kb/s-2.4kb/s低码率语音变码技术研究”，清华大学博士论文，1996
[43]. F.Itakura，“Line Spectrum Representation of Linear Predictive Coefficients of Speech Signals”。J. Acoust. Soc. Am., 57, S35(a), 535, 1975
[44]. 张连海。 “多带激励语音编码算法研究及IMBE算法实现”，解放军信息工程学院硕士论文，2000
[45]. 张连海，王炳锡，“4.15kbs IMBE算法研究与实现”
[46]. 陆奕奕，“AMBE语音压缩算法实现和改进”，通信技术，第四期，1998
[47]. 李强，“基于多带激励模型的低速语音编码方案及计算机仿真”，电子科技大学硕士论文，1999
[48]. P.C. Meuse，“A 2400bps Multi-Band Excitation Vocoder”，Proc. IEEE. Int. Conf. Acoust. Speech Signal Processing, pp. 9-12, April, 1990
[49]. F. K. Soong and B. H. Juang，“Line Spectrum Pair (LSP)and Speech Data Compression”，Proc. ICASSP-84, pp. 1.10-1.10.4, 1984
[50]. Shan Lu and Marion Baraniecki，“A Robust 1200bps Voice Coder Based on Vector Quantization of Line Spectral Frequencies”，Proc. of ICASSP, pp. 522-524, 1997
[51]. J. Malhoul, et al，“Vector Quantization in Speech coding”，Proc. IEEE, Vol.73, No. 11, pp. 1551-1558, November, 1985