8kbit/s CS-ACELP SPELP语音编码算法的研究与实现
摘要
本文先介绍了语音编码的发展情况，之后详细地论述了CS-ACELP算法，即8kbit/s的共轭结构代数码激励线性预测编码的完整结构和算法，对包括预处理、线性预测分析和量化(加窗和自相关计算、Levinson-Durbin算法、LP到LSP的转换、LSP系数的量化、LSP系数的内插、LSP到LP的转换)、感觉加权、开环基音分析、冲激响应计算、目标信号的计算、自适应码书的搜索(自适应码书矢量的产生、延迟码字的计算)、固定码书的构造和搜索、增益的量化(增益预测、用于增益量化的码书搜索、用于增益量化的码字计算)、存储器更新，语音合成后的处理等各模块的功能和理论基础作了细致分析。并对涉及到的语音处理的关键技术，如线性预测、LPC与LSP的转换、矢量量化、基音分析等技术作了深入研究。用标准C语言仿真实现了该算法，计算了MOS分值，女声：4.180497，男声：4.199782，并在相同的测试语句中加入噪声进行测试，含噪语句通过该编解码器，输出的合成语音用主、客观评价标准评价，与原始不含噪语音效果差别不大，平均MOS分值为：女声4.1375，男声4.1668，说明该算法是优秀的编解码算法。
此外，特别就CS-ACELP算法中的LSP量化方面作了深入的研究，尝试了几种不同的量化方法：(1)改变分裂式矢量量化的维数组合，原算法中第二级残差量化时用了两段式分裂量化法，将10维矢量分裂为两个5维矢量。本研究中，通过实验发现3维-7维的分法效果最好。(2)进行了码书优化。对LSP参数量化中的第一级码书的128个码字的使用频率进行了统计试验，选用了128个码字中使用频率高的112个码字作为新码书，语音质量基本不变但降低了码书搜索的复杂度。
This paper describes the 8kbit/s speech coding algorithm which has been standardized by ITU-T in 1996. The algorithm is based on a Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding technique and uses 1 Oms (80 samples at an 8 kHz sample rate) speech frames. This coder will be used for the Future Public Land Mobile Telecommunication System and will be suitable for Personal Communication Service. The coder delivers toll-quality speech (equivalent to 32kbit/s ADPCM) for most operating conditions. The coder operates on speech frames of 10ms, computes the long-term predictor coefficients, and operates in an analysis-by-synthesis loop to find the excitation vector that minimizes the perceptually weighted error signal.
In this paper, the coder structure is described, the algorithm about CS-ACELP is discussed, and its central aspects are analyzed in detail. To achieve high-quality speech and real-time implementation, CS-ACELP has been revised by novel schemes. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech. LSP parameters are quantized by multi-stage VQ with fourth-order interframe MA prediction. This scheme has little spectrum distortion, even if the two types of speech have many variations of LSP parameters. Moreover, computational complexity for implementation is reduced in adaptive and fixed-shape codebooks without degrading the quality. Multi-stage selection is adopted in the adaptive codebook; this selection uses a
truncated impulse response. Improved pre-selection is proposed in the fixed-shape codebook. Subjective testing indicates that the quality of CS-ACELP is equivalent to that of the 32kbit/s Adaptive Differential Pulse Code Modulation (ADPCM) under error-free conditions and it outperforms G.726 under error condition.
In this paper, Standard C is adopted in realization of the algorithm, presents program strategies and steps of algorithm of each module. The coder and decoder is tested by utterances with noise. The results are satisfying. Moreover, the paper studies the quantization of LSP and tries some other means to quantize the LSP parameter. We adopt the means as follows: first, we change the splitting dimension of the second grade codebook which was splirted into double five dimension. By testing, we find a better splitting way. Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. The result is not degraded too much while the complexity is reduced. At the end of the paper the development prospect of CS-ACELP and speech coding are described.
此外，特别就CS-ACELP算法中的LSP量化方面作了深入的研究，尝试了几种不同的量化方法：(1)改变分裂式矢量量化的维数组合，原算法中第二级残差量化时用了两段式分裂量化法，将10维矢量分裂为两个5维矢量。本研究中，通过实验发现3维-7维的分法效果最好。(2)进行了码书优化。对LSP参数量化中的第一级码书的128个码字的使用频率进行了统计试验，选用了128个码字中使用频率高的112个码字作为新码书，语音质量基本不变但降低了码书搜索的复杂度。
This paper describes the 8kbit/s speech coding algorithm which has been standardized by ITU-T in 1996. The algorithm is based on a Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding technique and uses 1 Oms (80 samples at an 8 kHz sample rate) speech frames. This coder will be used for the Future Public Land Mobile Telecommunication System and will be suitable for Personal Communication Service. The coder delivers toll-quality speech (equivalent to 32kbit/s ADPCM) for most operating conditions. The coder operates on speech frames of 10ms, computes the long-term predictor coefficients, and operates in an analysis-by-synthesis loop to find the excitation vector that minimizes the perceptually weighted error signal.
In this paper, the coder structure is described, the algorithm about CS-ACELP is discussed, and its central aspects are analyzed in detail. To achieve high-quality speech and real-time implementation, CS-ACELP has been revised by novel schemes. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech. LSP parameters are quantized by multi-stage VQ with fourth-order interframe MA prediction. This scheme has little spectrum distortion, even if the two types of speech have many variations of LSP parameters. Moreover, computational complexity for implementation is reduced in adaptive and fixed-shape codebooks without degrading the quality. Multi-stage selection is adopted in the adaptive codebook; this selection uses a
truncated impulse response. Improved pre-selection is proposed in the fixed-shape codebook. Subjective testing indicates that the quality of CS-ACELP is equivalent to that of the 32kbit/s Adaptive Differential Pulse Code Modulation (ADPCM) under error-free conditions and it outperforms G.726 under error condition.
In this paper, Standard C is adopted in realization of the algorithm, presents program strategies and steps of algorithm of each module. The coder and decoder is tested by utterances with noise. The results are satisfying. Moreover, the paper studies the quantization of LSP and tries some other means to quantize the LSP parameter. We adopt the means as follows: first, we change the splitting dimension of the second grade codebook which was splirted into double five dimension. By testing, we find a better splitting way. Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. The result is not degraded too much while the complexity is reduced. At the end of the paper the development prospect of CS-ACELP and speech coding are described.
引文
[1] 易克初、田斌、付强，语音信号处理，国防工业出版社，2000
[2] 张刚、张雪英等，语音处理与编码，兵器工业出版社2000年
[3] 姚天任，数字语音处理，武汉：华中理工大学出版社，1992
[4] CCITT, Recommendation G. 721, 32kbit/s自适应差分脉冲编码调制(ADPCM) 1984
[5] CCITT, Recommendation G.729, Coding of speech at 16kbit/s using low-delay code excited linear prediction, Geneva, September 1992
[6] 鲍长春，低比特率数字语音编码基础，北京工业大学出版社，2001
[7] 杨行俊、迟惠生等，语音信号数字处理，电子工业出版社，1995
[8] Shinji Hayashi, Akitoshi Kataoka, Takehiro Moriya. '8 kbit/s short and medium delay speech codecs based on CELP coding'. Vol.5, No.5, September-October 1994.
[9] G.Schroeder. 'The standardization process for the proposed ITU-T 8kb/s speech coder standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[10] P.Usai, P.Coverdale, D.Pascal, G.Schroeder, J.Sotscheck, and A.Takahashi. 'Subjective performance of the proposed ITU-T 8kb/s speech coding standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[11] R. Salami et al. 'Description of the proposed ITU-T 8kb/s speech coding standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[12] L．R．拉宾纳[美]，R．W．谢佛著，语音信号数字处理，科学出版社，1983
[13] Salami R., at al. 'Design and description of CS-ACELP: A toll quality 8kb/s speech coder', IEEE Trans. Speech Audio Process., 1996, 6, (2)
[14] Mark E. Perkins, Keith Evans, Dominique Pascal, Leigh A. Thorpe. Characterizing the Subjective Performance of the ITU-T 8kb/s speech coding algorithm-ITU-T G.729'. IEEE Communications Magazine, September 1997．
[15] 曹志刚、钱亚生，现代通信原理，清华大学出版社, 1992
[16] R.Salami, C.Laflamme and J-P.Adoul. 'ACELP Speech Coder at 8kbit/s with a 10ms frame: A Candidate for CCITT Standardization'. Proc. IEEE workshop on Speech Coding, pp.23-24, 1993．
[17] A.Kataoka, T.Moriya and S.Hayashi. 'Conjugate Structure CELP for the CCITT 8-kbit/s Standardization Candidate'. Proc. IEEE Workshop on Speech Coding, pp.25-26, 1993．
[18] Richard V. Cox and Peter Kroon. 'Low bit-rate speech coders for multimedia communication'. IEEE Communications Magazine, December 1996,pp.34-40．
[19] Akitoshi Kataoka, Jean-Pierre Adoul, Pierre Combescure and Peter Kroon. 'ITU-T 8-kbit/s Standard Speech Coder for Personal Communication Services'. Proc. Int. Conf. Universal Personal Communications 1995, pp.818-822．
[20] R.Salami, C.Laflamme and J-P.Adoul. '8hbit/s ACELP Coding of Speech with 10ms Speech-frame: A Candidate for CCITT Standardization'. IEEE Proceedings ICASSP'94 Vol.2 PII-97-P II-100．
[21] Lan Juan, Lin Biqin, Fu Qiuliang. 'An 8-kb/s Conjugate-structure Algebraic CELP (CS-ACELP) Speech Coding. Proceedings of ICSP'98． pp. 1729-1732．
[22] J.-H. Chen and R.V.Cox, The creation and evolution of 16kbit/s
LD-CELP: From concept to standard, Speech Communication, Vol. 12, No.2, June 1993, pp.103～P111
[23] Jey-Hsin Yao, John J.S. and Allen Gersho, Low-Delay Vector Excitation Coding of Speech at 8 kbit/s, Proc. IEEE ICASSP, 1991, pp.0695-0698
[24] 鲍长春等，码激励线性预测语音编码快速算法和评价，电子科学学刊，1995年第1期
[25] S.Dimolitsas and F.L.Corcoran, Use of Low-Delay Code-Excited Linear Prediction Technology in Circuit Multiplexed Networks, Proc. IEEE ICASSP, 1993, pp.Ⅱ-608-Ⅱ-611
[26] 王磊、刘维献、姜威，语音信号DPCM压缩编码方法的研究，山东电子，1996年，第4期，pp．25-27
[27] D.Ritra, An almost linear relation between the step size behavior and the input signal in density in robust adaptive quantization, IEEE. Trans. Commun., Vol. COM-27, NO.3.March 1957
[28] L.H.Goldstein, Quantization noise in ADPCM systems, IEEE Transactions on Communications, Vol. COM-25, No.2, pp.227-237, February 1977
[29] L. H.Goldstein and B.Liu, Quantization error and step-size distributions in ADPCM, IEEE Transactions on Information Theory, VOL. IT-23, NO.2, March 1977, pp.216-223
[30] 欧阳缮、陈云宇、方惠均，一种改进的语音信号非线性自适应预测编码方案，电路与系统学报，1999年，第4卷，第2期，pp．7-9
[31] 林嘉宇、易波、王跃科，低速率语音编码体制研究，系统工程与电子技术，1998年，第11期，pp．18-21
[32] 胡毅、胡咏梅、柴乔林，语音通信中语音帧丢失补偿算法的设计与实现，计算机工程与科学，2001年，第23卷，第3期，pp．32-34
[33] A.Kataoka, T.Moriya and S.Hayashi. 'An 8-kbit/s Speech Coder on Conjugate Structure CELP'. Proc. ICASSP'93, pp.592-595, 1993.
[34] Redwan Salami,Claude L,Jean-P.'Design and Description of CS-ACELP:A Toll Quality 8kb/s Speech Coder',IEEE transactions on speech and audio processing, Vol.6, No.2, Match, 1998
[35] Akitoshi Kataoka, Takehiro Moriya and Shinji Hayashi. 'Implementation and preformance of an 8-kbit/s conjugate structure CELP speech coder'. IEEE Proceedings ICASSP'94 Vol.2 PⅡ-93～PⅡ-96.
[36] 温斌等，中低速率语音编码技术的发展及应用，电信科学，1996年，第10期，pp．35-38
[37] 张献民，中、低速率话音编码技术在卫星通信中的应用，通信技术与发展，1994年，第5期，pp．36-41
[38] [美]J．D．马卡尔等编著，娄乃英翻译，语音信号线性预测，中国铁道出版社，1987
[39] 于增贵，低速语音编码的最新成果，通信技术，1996年，第4期，pp55-64
[40] S.Singhal and B.S Atal. Improving Performance of Mutipulse LPC Coders at Low Rates. Proc.IEEE ICASSP, 1984,1.3.1-1.3.4
[41] G.Davidson and A.Gersho. Complexity Reduction Methods for Vector Excitation Coding. Proc.IEEE ICASSP, 1986,3055-3058
[2] 张刚、张雪英等，语音处理与编码，兵器工业出版社2000年
[3] 姚天任，数字语音处理，武汉：华中理工大学出版社，1992
[4] CCITT, Recommendation G. 721, 32kbit/s自适应差分脉冲编码调制(ADPCM) 1984
[5] CCITT, Recommendation G.729, Coding of speech at 16kbit/s using low-delay code excited linear prediction, Geneva, September 1992
[6] 鲍长春，低比特率数字语音编码基础，北京工业大学出版社，2001
[7] 杨行俊、迟惠生等，语音信号数字处理，电子工业出版社，1995
[8] Shinji Hayashi, Akitoshi Kataoka, Takehiro Moriya. '8 kbit/s short and medium delay speech codecs based on CELP coding'. Vol.5, No.5, September-October 1994.
[9] G.Schroeder. 'The standardization process for the proposed ITU-T 8kb/s speech coder standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[10] P.Usai, P.Coverdale, D.Pascal, G.Schroeder, J.Sotscheck, and A.Takahashi. 'Subjective performance of the proposed ITU-T 8kb/s speech coding standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[11] R. Salami et al. 'Description of the proposed ITU-T 8kb/s speech coding standard'. IEEE Speech Coding Workshop, Annapolis, 1995.
[12] L．R．拉宾纳[美]，R．W．谢佛著，语音信号数字处理，科学出版社，1983
[13] Salami R., at al. 'Design and description of CS-ACELP: A toll quality 8kb/s speech coder', IEEE Trans. Speech Audio Process., 1996, 6, (2)
[14] Mark E. Perkins, Keith Evans, Dominique Pascal, Leigh A. Thorpe. Characterizing the Subjective Performance of the ITU-T 8kb/s speech coding algorithm-ITU-T G.729'. IEEE Communications Magazine, September 1997．
[15] 曹志刚、钱亚生，现代通信原理，清华大学出版社, 1992
[16] R.Salami, C.Laflamme and J-P.Adoul. 'ACELP Speech Coder at 8kbit/s with a 10ms frame: A Candidate for CCITT Standardization'. Proc. IEEE workshop on Speech Coding, pp.23-24, 1993．
[17] A.Kataoka, T.Moriya and S.Hayashi. 'Conjugate Structure CELP for the CCITT 8-kbit/s Standardization Candidate'. Proc. IEEE Workshop on Speech Coding, pp.25-26, 1993．
[18] Richard V. Cox and Peter Kroon. 'Low bit-rate speech coders for multimedia communication'. IEEE Communications Magazine, December 1996,pp.34-40．
[19] Akitoshi Kataoka, Jean-Pierre Adoul, Pierre Combescure and Peter Kroon. 'ITU-T 8-kbit/s Standard Speech Coder for Personal Communication Services'. Proc. Int. Conf. Universal Personal Communications 1995, pp.818-822．
[20] R.Salami, C.Laflamme and J-P.Adoul. '8hbit/s ACELP Coding of Speech with 10ms Speech-frame: A Candidate for CCITT Standardization'. IEEE Proceedings ICASSP'94 Vol.2 PII-97-P II-100．
[21] Lan Juan, Lin Biqin, Fu Qiuliang. 'An 8-kb/s Conjugate-structure Algebraic CELP (CS-ACELP) Speech Coding. Proceedings of ICSP'98． pp. 1729-1732．
[22] J.-H. Chen and R.V.Cox, The creation and evolution of 16kbit/s
LD-CELP: From concept to standard, Speech Communication, Vol. 12, No.2, June 1993, pp.103～P111
[23] Jey-Hsin Yao, John J.S. and Allen Gersho, Low-Delay Vector Excitation Coding of Speech at 8 kbit/s, Proc. IEEE ICASSP, 1991, pp.0695-0698
[24] 鲍长春等，码激励线性预测语音编码快速算法和评价，电子科学学刊，1995年第1期
[25] S.Dimolitsas and F.L.Corcoran, Use of Low-Delay Code-Excited Linear Prediction Technology in Circuit Multiplexed Networks, Proc. IEEE ICASSP, 1993, pp.Ⅱ-608-Ⅱ-611
[26] 王磊、刘维献、姜威，语音信号DPCM压缩编码方法的研究，山东电子，1996年，第4期，pp．25-27
[27] D.Ritra, An almost linear relation between the step size behavior and the input signal in density in robust adaptive quantization, IEEE. Trans. Commun., Vol. COM-27, NO.3.March 1957
[28] L.H.Goldstein, Quantization noise in ADPCM systems, IEEE Transactions on Communications, Vol. COM-25, No.2, pp.227-237, February 1977
[29] L. H.Goldstein and B.Liu, Quantization error and step-size distributions in ADPCM, IEEE Transactions on Information Theory, VOL. IT-23, NO.2, March 1977, pp.216-223
[30] 欧阳缮、陈云宇、方惠均，一种改进的语音信号非线性自适应预测编码方案，电路与系统学报，1999年，第4卷，第2期，pp．7-9
[31] 林嘉宇、易波、王跃科，低速率语音编码体制研究，系统工程与电子技术，1998年，第11期，pp．18-21
[32] 胡毅、胡咏梅、柴乔林，语音通信中语音帧丢失补偿算法的设计与实现，计算机工程与科学，2001年，第23卷，第3期，pp．32-34
[33] A.Kataoka, T.Moriya and S.Hayashi. 'An 8-kbit/s Speech Coder on Conjugate Structure CELP'. Proc. ICASSP'93, pp.592-595, 1993.
[34] Redwan Salami,Claude L,Jean-P.'Design and Description of CS-ACELP:A Toll Quality 8kb/s Speech Coder',IEEE transactions on speech and audio processing, Vol.6, No.2, Match, 1998
[35] Akitoshi Kataoka, Takehiro Moriya and Shinji Hayashi. 'Implementation and preformance of an 8-kbit/s conjugate structure CELP speech coder'. IEEE Proceedings ICASSP'94 Vol.2 PⅡ-93～PⅡ-96.
[36] 温斌等，中低速率语音编码技术的发展及应用，电信科学，1996年，第10期，pp．35-38
[37] 张献民，中、低速率话音编码技术在卫星通信中的应用，通信技术与发展，1994年，第5期，pp．36-41
[38] [美]J．D．马卡尔等编著，娄乃英翻译，语音信号线性预测，中国铁道出版社，1987
[39] 于增贵，低速语音编码的最新成果，通信技术，1996年，第4期，pp55-64
[40] S.Singhal and B.S Atal. Improving Performance of Mutipulse LPC Coders at Low Rates. Proc.IEEE ICASSP, 1984,1.3.1-1.3.4
[41] G.Davidson and A.Gersho. Complexity Reduction Methods for Vector Excitation Coding. Proc.IEEE ICASSP, 1986,3055-3058
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