语音密码机中的语音压缩改进算法研究
摘要
现阶段,为了满足高速发展的数字通信和其他商业应用需求,语音压缩编码技术得到了良好发展。语音密码机的实现的核心技术是语音压缩编解码技术,在本论文中选择了MELP语音压缩算法来实现。
在语音编码算法中,MELP算法由于能更好的模拟自然语音的特征,能在低速率上合成高质量的语音,因此成为现在低速率语音编码中很有潜力的算法。它在原有的LPC(线性预测)编码的基础上,采用了混合激励、非周期脉冲、自适应谱增强、脉冲整形滤波和傅氏级数幅度值等五项新技术,使得合成语音能更好地拟合自然语音。从而较好的实现了低码率的语音编码。
本论文深入分析了MELP的语音编解码算法的原理,对它的编解码过程中的关键技术进行了归纳总结,其中的一些公式进行了理论推导,并在这基础上对MELP算法在LSF量化方面提出了改进,并做畸变比较分析。数据结果表明,改进后的MELP在2.4kb/s上得到更好的语音合成,更好的拟合成自然语音。
最后基于MATLAB进行了语音信号频谱分析,同时本文提出了DSP上语音数据的获取和存储的实现方法,以及使用CCS软件进行优化代码的方法。最终实现了语音密码机中的MELP算法的改进和优化。
In order to satisfy demands of the digital communication and other commercial applications,the speech compression technology has been developed rapidly. The realization of the voice cipher core technology is speech compression encoding &decoding technology, in this thesis chose MELP speech compress algorithm to realize.
In speech coding algorithm, because MELP algorithm can better simulating natural speech characteristics and can synthesize a high quality speech in the low bite rate, thus became now low speed speech coding has great potential algorithm. It in the original LPC rules (linear forecast) code, and on the basis of using hybrid excitation, the periodic impulsive, the adaptive enhancement, pulse shaping filter and Fourier series amplitude value five other new techniques, making synthesized speech can better fitting natural voice. Thus better realize low bit rate speech coding.
This paper deeply analyzed the MELP speech decoding algorithm of principle, to its key technology in the decoding process has summarized, and some of these formulas for the theoretical derivation, and on this basis to MELP algorithm in LSF quantification and puts forward the improvement and distortion comparison analysis. Experimental results show that the modified MELP in 2.4 kb/s on get better speech synthesis, better fitting natural voice synthesis.
Finally, based on MATLAB for the speech signal spectrum analysis, the paper proposed DSP on speech data acquisition and storage, and the realization method of using CCS optimized method of software code. Finally achieved the voice cipher MELP algorithm is improved and optimized.
在语音编码算法中,MELP算法由于能更好的模拟自然语音的特征,能在低速率上合成高质量的语音,因此成为现在低速率语音编码中很有潜力的算法。它在原有的LPC(线性预测)编码的基础上,采用了混合激励、非周期脉冲、自适应谱增强、脉冲整形滤波和傅氏级数幅度值等五项新技术,使得合成语音能更好地拟合自然语音。从而较好的实现了低码率的语音编码。
本论文深入分析了MELP的语音编解码算法的原理,对它的编解码过程中的关键技术进行了归纳总结,其中的一些公式进行了理论推导,并在这基础上对MELP算法在LSF量化方面提出了改进,并做畸变比较分析。数据结果表明,改进后的MELP在2.4kb/s上得到更好的语音合成,更好的拟合成自然语音。
最后基于MATLAB进行了语音信号频谱分析,同时本文提出了DSP上语音数据的获取和存储的实现方法,以及使用CCS软件进行优化代码的方法。最终实现了语音密码机中的MELP算法的改进和优化。
In order to satisfy demands of the digital communication and other commercial applications,the speech compression technology has been developed rapidly. The realization of the voice cipher core technology is speech compression encoding &decoding technology, in this thesis chose MELP speech compress algorithm to realize.
In speech coding algorithm, because MELP algorithm can better simulating natural speech characteristics and can synthesize a high quality speech in the low bite rate, thus became now low speed speech coding has great potential algorithm. It in the original LPC rules (linear forecast) code, and on the basis of using hybrid excitation, the periodic impulsive, the adaptive enhancement, pulse shaping filter and Fourier series amplitude value five other new techniques, making synthesized speech can better fitting natural voice. Thus better realize low bit rate speech coding.
This paper deeply analyzed the MELP speech decoding algorithm of principle, to its key technology in the decoding process has summarized, and some of these formulas for the theoretical derivation, and on this basis to MELP algorithm in LSF quantification and puts forward the improvement and distortion comparison analysis. Experimental results show that the modified MELP in 2.4 kb/s on get better speech synthesis, better fitting natural voice synthesis.
Finally, based on MATLAB for the speech signal spectrum analysis, the paper proposed DSP on speech data acquisition and storage, and the realization method of using CCS optimized method of software code. Finally achieved the voice cipher MELP algorithm is improved and optimized.
引文
[1]易克初,田斌,付强.语音信号处理[M].北京:国防工业出版社, 2000.
[2]王炳锡.语音编码[M].西安:西安电子科技大学出版社, 2002.6.
[3]杨行峻,迟惠生等.语音信号数字处理[M].北京:电子工业出版社, 1995.
[4]胡光锐.语音处理与识别[M].上海:上海科学技术文献出版社,1994.
[5] Paulo S.R.Diniz, Eduardo A.B.da Silva, SergioL.Netto. Digital Signal Processing-System Analysis and Design[M]. Cambridge University Press, 2002.
[6] Salor.O.,Demirekler,M.An An Analysis on LPC Spectrum Spectrum and LPC Residual Spectrum Relationship for Voice Transrormation[J].ICSLP.2004.
[7] F.C.A.Brooks,Lajos Hanzo. A Multiband Excited Waveform-Interpolated 2.35kb/s Speech Coder for Bandlimited Channels.IEEE Transaction Vehicular Technology,2000,vol.49(3):766～772.
[8] O.Gottesman,A.Gersho. Enhanced waveform interpolation coding at low bit-rate Speech And Audio Processing.IEEE Transaction on Speech and Audio Processing,2001,vol.9: 786～798.
[9] International Telecommunication Union-T Recommendation G.723.1 (03/96).
[10] International Telecommunication Union-T Recommendation G.729 (03/96).
[11] International Telecommunication Union-T Recommendation G.729A (11/96).
[12] International Telecommunication Union-T Recommendation G.729B (11/96).
[13] RabinerL.RandSchafer.R,W.Digital Processing of SpeechSignals[M]. Prentice-Hall,Inc,1978.
[14] Changchun Bao. Harmonic excitation LPC(HE-LPC)speech coding at 2.3 kb/s Acoustics,Speech,and Signal Processing,2003 Proceedings.(ICASSP‘03).2003 IEEE International Conference on,Volume:1,6-10 April 2003.
[15] McCree, A, Kwan Truong,George, E.B. A 2.4 kbit/s MELP coder candidate for the new U.S. Federal Standard[J].IEEE.1996.
[16] Harwell, K.,Scheets, G.,Weber, J. A Multilanguage Study of the Quality of Interleaved MELP Voice Traffic Over a Lossy Network[J].IEEE.2009.
[17] Xia Zou,Xiongwei Zhang. High quality 0.6/1.2/2.4kbps multi-band LPC speech coding algorithm[J].Wireless,Mobile and Multimedia Networks.2006.
[18] Khalili, F,Ardebilipour, M.Sameti, H. Design and implementation of VectorQuantizer for a 600 bps cocoder Based on MELP[J].ICACT 2009.
[19] Tian Wang Koishida, K. Cuperman, V. Gersho, A. Collura, J.S. A 1200/2400 bps coding suite based on MELP[J].IEEE.2002.
[20] Lu Gao,Li Guo,Canxing Lu. Acceleration of MELP Algorithm Using DSP Coprocessor with Extended Registers[J]. Digital System Design, Architectures, Methods and Tools, 2009. DSD '09.
[21] Takahiro Unno, Thomas P.Barnwell, KwanTruong .An improved Mixed Excitation Linear Prediction(MELP) coder[J].IEEE IC ASSP,2003.
[22] Ronald P.Cohn and John S.Collura.Incorporating perception into LSF quantization–some experiments[J].IEEE.1997.
[23] Wei Ran Lin,Soo Ngee Koh,Xiao Lin.Widerband speech coding using MELP model[J].School of IEEE.1999.
[24]Chaogang Wu,Hongxu Jiang,Bo Li.An improved MELP speech coder.International Conference on Information Technology and Computer Science.2009.
[25] Qzgiil Salor,Miibeccel Demirekler.Spectral modification for context-free voice conversion using MELP speech coding framework[J].International Symposium on Intelligent Multimedia.2004.
[26] M.Djamah and D.O‘Shaughnessy.Low-complexity encoding of speech LSF parameters using multistage tree-structured vector quantization:application to the MELP coder[J].IEEE.2009.
[27] Selma Ozaydin,Buyurman Baykal.A 1200 bps speech coder with LSF matrix quzatization[J].IEEE.2001.
[28]顾堃.单芯片语音密码机中的语音压缩算法研究[D].西安电子科技大学.2007.
[29]贾亮,赵鹏飞,危国腾.混合激励线性预测低速率语音编码研究[J].电子工程设计.2010.3.25-28.
[30]戚银城,张巍,苑津莎.超低速率MELP语音编码算法研究[J].声学技术.2007.12.1196-1200.
[31]郭莉,郭丽红,王炳锡.极低速率语音编码的新发展与应用[J].电声技术.2005.6.38-41.
[32] S.Haykin.Nonlinear Methods of Spectral Analysis[A]. Springer Verlag, Berlin,Heidelberg, 1979.
[33] A.V.Oppenheim, R.w.Schafer. Digital Signal Processing[M]. Prentice Hall, 1975.
[34] J. Makhoul. Linear Prediction: A Tutorial Review[A]. Proc. IEEE, 1975, 63:561-580.
[35] Thomas E. Tremain. The Government Standard Linear Predictive Coding Algorithm: LPC-10[J]. Speech Technology, 1982, April, 40-49.
[36] Chu, W.C. A scalable MELP coder based on embedded quantization of line spectral frequencies[J].ISPACS.2005.
[37] Guo Li,Zhu Wenyu,Wang Bingxi. An improved 1.2 kb/s speech coder based on MELP[J].ICSP.2004.
[38] McCree.A,De Martin, J.C. A 1.7 kb/s MELP coder with improved analysis and quantization[J].IEEE.1998.
[39] J. makhou, IEEE Trans. Acousties, speech and signal proc[A]. 1997.
[40] GershoA.Advances in speech and audio compression IEEE Proc . Data Compression, 1994.
[41] Texas Instruments. TLV320 AIC23 Stereo Audio Coded, 8 to 96KHz[J], With Integrated Headphone Amplifier Data Manual.July, 2001.
[42] WANG Ping,GAO Jun,TIAN Pei-gen. Parameter compensation of UV communication system of speech signal and simulation by Matlab[J]. Journal of Naval University of Engineering.2008.05.
[43]鲍长春.低比特率数字语音编码基础[M].北京:北京工业大学出版社. 2001.149-178.
[44] Alan V. McCree, Kwan Truong, E. Bryan George and Thomas P. Barnwell. A 2.4K BIT/S MELP coder candidate for the new U.S. Federal[J]. Proceeding of ICASSP.1996.107-112.
[45] McCree AV. BarnwellTP. ANewMixed Excitation LPC Vocoder.IC-ASSP[J]. 199l.593-596.
[46]周建兴.MATLAB从入门到精通(附光盘)[M].北京:北京邮电出版社.2008.11.
[47]张雪英.数字语音处理及MATLAB仿真[M].北京:电子工业出版社.2010.7.
[48]戴明桢,周建江.TMS320C54x DSP结构、原理与应用[M].北京:北京航空航天大学出版社.2007.8.
[49] TMS320C54x DSP Programmer‘s Guide[Z]. Texas Instruments, 2001.
[50] TMS320C54x DSP Optimizing C/C++ Compiler User‘s Guide (Rev.G).Texas Instruments, 2002.
[51] TMS320C54x DSP CPU and Peripherals Reference Set, Volume1 (Rev.G). Texas Instruments, 2001.
[52] TMS320C54x DSP CPU and Peripherals Reference Set, Volume2 (Rev.G). Texas Instruments, 2001.
[53] Chassaing.R,王华.译.DSP原理及其C编程开发技术[M].北京:电子工业出版社.2005.7.
[2]王炳锡.语音编码[M].西安:西安电子科技大学出版社, 2002.6.
[3]杨行峻,迟惠生等.语音信号数字处理[M].北京:电子工业出版社, 1995.
[4]胡光锐.语音处理与识别[M].上海:上海科学技术文献出版社,1994.
[5] Paulo S.R.Diniz, Eduardo A.B.da Silva, SergioL.Netto. Digital Signal Processing-System Analysis and Design[M]. Cambridge University Press, 2002.
[6] Salor.O.,Demirekler,M.An An Analysis on LPC Spectrum Spectrum and LPC Residual Spectrum Relationship for Voice Transrormation[J].ICSLP.2004.
[7] F.C.A.Brooks,Lajos Hanzo. A Multiband Excited Waveform-Interpolated 2.35kb/s Speech Coder for Bandlimited Channels.IEEE Transaction Vehicular Technology,2000,vol.49(3):766～772.
[8] O.Gottesman,A.Gersho. Enhanced waveform interpolation coding at low bit-rate Speech And Audio Processing.IEEE Transaction on Speech and Audio Processing,2001,vol.9: 786～798.
[9] International Telecommunication Union-T Recommendation G.723.1 (03/96).
[10] International Telecommunication Union-T Recommendation G.729 (03/96).
[11] International Telecommunication Union-T Recommendation G.729A (11/96).
[12] International Telecommunication Union-T Recommendation G.729B (11/96).
[13] RabinerL.RandSchafer.R,W.Digital Processing of SpeechSignals[M]. Prentice-Hall,Inc,1978.
[14] Changchun Bao. Harmonic excitation LPC(HE-LPC)speech coding at 2.3 kb/s Acoustics,Speech,and Signal Processing,2003 Proceedings.(ICASSP‘03).2003 IEEE International Conference on,Volume:1,6-10 April 2003.
[15] McCree, A, Kwan Truong,George, E.B. A 2.4 kbit/s MELP coder candidate for the new U.S. Federal Standard[J].IEEE.1996.
[16] Harwell, K.,Scheets, G.,Weber, J. A Multilanguage Study of the Quality of Interleaved MELP Voice Traffic Over a Lossy Network[J].IEEE.2009.
[17] Xia Zou,Xiongwei Zhang. High quality 0.6/1.2/2.4kbps multi-band LPC speech coding algorithm[J].Wireless,Mobile and Multimedia Networks.2006.
[18] Khalili, F,Ardebilipour, M.Sameti, H. Design and implementation of VectorQuantizer for a 600 bps cocoder Based on MELP[J].ICACT 2009.
[19] Tian Wang Koishida, K. Cuperman, V. Gersho, A. Collura, J.S. A 1200/2400 bps coding suite based on MELP[J].IEEE.2002.
[20] Lu Gao,Li Guo,Canxing Lu. Acceleration of MELP Algorithm Using DSP Coprocessor with Extended Registers[J]. Digital System Design, Architectures, Methods and Tools, 2009. DSD '09.
[21] Takahiro Unno, Thomas P.Barnwell, KwanTruong .An improved Mixed Excitation Linear Prediction(MELP) coder[J].IEEE IC ASSP,2003.
[22] Ronald P.Cohn and John S.Collura.Incorporating perception into LSF quantization–some experiments[J].IEEE.1997.
[23] Wei Ran Lin,Soo Ngee Koh,Xiao Lin.Widerband speech coding using MELP model[J].School of IEEE.1999.
[24]Chaogang Wu,Hongxu Jiang,Bo Li.An improved MELP speech coder.International Conference on Information Technology and Computer Science.2009.
[25] Qzgiil Salor,Miibeccel Demirekler.Spectral modification for context-free voice conversion using MELP speech coding framework[J].International Symposium on Intelligent Multimedia.2004.
[26] M.Djamah and D.O‘Shaughnessy.Low-complexity encoding of speech LSF parameters using multistage tree-structured vector quantization:application to the MELP coder[J].IEEE.2009.
[27] Selma Ozaydin,Buyurman Baykal.A 1200 bps speech coder with LSF matrix quzatization[J].IEEE.2001.
[28]顾堃.单芯片语音密码机中的语音压缩算法研究[D].西安电子科技大学.2007.
[29]贾亮,赵鹏飞,危国腾.混合激励线性预测低速率语音编码研究[J].电子工程设计.2010.3.25-28.
[30]戚银城,张巍,苑津莎.超低速率MELP语音编码算法研究[J].声学技术.2007.12.1196-1200.
[31]郭莉,郭丽红,王炳锡.极低速率语音编码的新发展与应用[J].电声技术.2005.6.38-41.
[32] S.Haykin.Nonlinear Methods of Spectral Analysis[A]. Springer Verlag, Berlin,Heidelberg, 1979.
[33] A.V.Oppenheim, R.w.Schafer. Digital Signal Processing[M]. Prentice Hall, 1975.
[34] J. Makhoul. Linear Prediction: A Tutorial Review[A]. Proc. IEEE, 1975, 63:561-580.
[35] Thomas E. Tremain. The Government Standard Linear Predictive Coding Algorithm: LPC-10[J]. Speech Technology, 1982, April, 40-49.
[36] Chu, W.C. A scalable MELP coder based on embedded quantization of line spectral frequencies[J].ISPACS.2005.
[37] Guo Li,Zhu Wenyu,Wang Bingxi. An improved 1.2 kb/s speech coder based on MELP[J].ICSP.2004.
[38] McCree.A,De Martin, J.C. A 1.7 kb/s MELP coder with improved analysis and quantization[J].IEEE.1998.
[39] J. makhou, IEEE Trans. Acousties, speech and signal proc[A]. 1997.
[40] GershoA.Advances in speech and audio compression IEEE Proc . Data Compression, 1994.
[41] Texas Instruments. TLV320 AIC23 Stereo Audio Coded, 8 to 96KHz[J], With Integrated Headphone Amplifier Data Manual.July, 2001.
[42] WANG Ping,GAO Jun,TIAN Pei-gen. Parameter compensation of UV communication system of speech signal and simulation by Matlab[J]. Journal of Naval University of Engineering.2008.05.
[43]鲍长春.低比特率数字语音编码基础[M].北京:北京工业大学出版社. 2001.149-178.
[44] Alan V. McCree, Kwan Truong, E. Bryan George and Thomas P. Barnwell. A 2.4K BIT/S MELP coder candidate for the new U.S. Federal[J]. Proceeding of ICASSP.1996.107-112.
[45] McCree AV. BarnwellTP. ANewMixed Excitation LPC Vocoder.IC-ASSP[J]. 199l.593-596.
[46]周建兴.MATLAB从入门到精通(附光盘)[M].北京:北京邮电出版社.2008.11.
[47]张雪英.数字语音处理及MATLAB仿真[M].北京:电子工业出版社.2010.7.
[48]戴明桢,周建江.TMS320C54x DSP结构、原理与应用[M].北京:北京航空航天大学出版社.2007.8.
[49] TMS320C54x DSP Programmer‘s Guide[Z]. Texas Instruments, 2001.
[50] TMS320C54x DSP Optimizing C/C++ Compiler User‘s Guide (Rev.G).Texas Instruments, 2002.
[51] TMS320C54x DSP CPU and Peripherals Reference Set, Volume1 (Rev.G). Texas Instruments, 2001.
[52] TMS320C54x DSP CPU and Peripherals Reference Set, Volume2 (Rev.G). Texas Instruments, 2001.
[53] Chassaing.R,王华.译.DSP原理及其C编程开发技术[M].北京:电子工业出版社.2005.7.