通信系统中语音编码判别技术研究
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摘要
语言是人类进行交流的重要手段，通信系统中最常见的数据形式就是语音。随着移动通信与网络技术的飞速发展，语音通信技术也在不断地进行更新并与之相融合。由于通信网络用户数量的增加、网络业务综合化、多样化，网络带宽与系统容量、服务质量的矛盾日益突出，传统的64kbit／s的PCM语音压缩编码己不能满足不断变得拥挤的传输信道要求。因此，在不牺牲语音通话质量的前提下大幅度降低传输比特速率的混合编码得到了广泛运用，在第三代移动通信系统3G标准中以自适应多速率语音编码AMR为代表的语音编码方式也日益成熟。
本文针对语音编码，主要是参量编码和混合编码的结构进行分析。前三章从语音产生的离散数学模型出发，简要叙述了语音压缩编码中采用到的一些相关基本原理和主要技术，以及在WCDMA系统中应用的自适应多速率语音编码器(AMR)。
在论文第四章中，我根据语音编码的结构特点，提出一种语音编码的判别方法。即在未知发送端使用了何种语音编码方式的情形下，在接收端通过对接收到的编码后的数据流进行分析处理，从分析结果中判断在发送端到底使用了哪一种编码方式，从而便于进行解码。通过计算机试验，对整个流程进行仿真，可以得到一个比较有信服力的结果以证明此方法的正确性。
There is no doubt that speech communication is an important intercourse tool for human. It is also the most common form in data communication systems. With the development of mobile communication and network, the speech communication technology has innovated and developed rapidly. With the increasing of network users, the integrating and diversifying of network service, the contradictions among bandwidth、system capacity and service quality are more and more obvious. Traditional 64kbps PCM speech coding can not satisfy the requirement of transmit bandwidth which is becoming more and more congested. So, hybrid coding who can reduce the bit-rate of speech without notably degrading its perceptual quality is used widely, and the speech coding represented by AMR are more mature in 3G standard.
My thesis analyze the structure of speech coding, especially the parameter coding and hybrid coding. The first three chapters describe the basic principle and main technology of speech coding, and then introduce AMR which used in WCDMA.
The fourth chapter proposed a method of speech coding identification based on the characteristics of coding. According to this method, if we do not know which speech coding is used at the transmitter, we can analyze the data stream received at the receiver and find out the exact coding style from the deal result, so that it is convenient to next step decoding. The whole course is simulated by computer test, and it shows a believed result.
本文针对语音编码，主要是参量编码和混合编码的结构进行分析。前三章从语音产生的离散数学模型出发，简要叙述了语音压缩编码中采用到的一些相关基本原理和主要技术，以及在WCDMA系统中应用的自适应多速率语音编码器(AMR)。
在论文第四章中，我根据语音编码的结构特点，提出一种语音编码的判别方法。即在未知发送端使用了何种语音编码方式的情形下，在接收端通过对接收到的编码后的数据流进行分析处理，从分析结果中判断在发送端到底使用了哪一种编码方式，从而便于进行解码。通过计算机试验，对整个流程进行仿真，可以得到一个比较有信服力的结果以证明此方法的正确性。
There is no doubt that speech communication is an important intercourse tool for human. It is also the most common form in data communication systems. With the development of mobile communication and network, the speech communication technology has innovated and developed rapidly. With the increasing of network users, the integrating and diversifying of network service, the contradictions among bandwidth、system capacity and service quality are more and more obvious. Traditional 64kbps PCM speech coding can not satisfy the requirement of transmit bandwidth which is becoming more and more congested. So, hybrid coding who can reduce the bit-rate of speech without notably degrading its perceptual quality is used widely, and the speech coding represented by AMR are more mature in 3G standard.
My thesis analyze the structure of speech coding, especially the parameter coding and hybrid coding. The first three chapters describe the basic principle and main technology of speech coding, and then introduce AMR which used in WCDMA.
The fourth chapter proposed a method of speech coding identification based on the characteristics of coding. According to this method, if we do not know which speech coding is used at the transmitter, we can analyze the data stream received at the receiver and find out the exact coding style from the deal result, so that it is convenient to next step decoding. The whole course is simulated by computer test, and it shows a believed result.
引文
[1] WAI C.CHU Speech Coding Algorithms Foundation and Evolution of Standardized Coders A John Wiley & Sons,Inc.
[2] 李昌立 吴善培 数字语音—语音编码实用教程 人民邮电出版社2004年
[3] 曹志刚 钱亚生 现代通信原理 清华大学出版社 1992年
[4] Announcing the Standard for Analog to Digital Conversion of Voice by 2,400 Bit/second Mixed Excitation Linear Prediction (MELP) Federal Information Processing Standards Publication
[5] ITU-T G.711 Pulse Code Modulation(PCM) of Voice Frequencies 1993
[6] ITU-T G.726 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM) 1997
[7] ITU-T G.723 Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s 1997
[8] ITU-T G.729 Coding of Speech At 8kbps Using Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-CELP) 1996.3
[9] ITU-T G.728 Coding of Speech at 16kbit/s Using Low-delay Code Excited Linear Prediction 1992
[10] Richard V. Cox. Speech Coding 1999.5.8.
[11] Jeremy Bradbury Linear Predictive Coding 2000.12.5
[12] 胡航 语音信号处理 哈尔滨工业大学出版社，2000年
[13] 戴沁云 胡捍英 第三代移动通信系统中的语音编码—自适应多码率AMR语音编码无线通信技术 2001年第3期
[2] 李昌立 吴善培 数字语音—语音编码实用教程 人民邮电出版社2004年
[3] 曹志刚 钱亚生 现代通信原理 清华大学出版社 1992年
[4] Announcing the Standard for Analog to Digital Conversion of Voice by 2,400 Bit/second Mixed Excitation Linear Prediction (MELP) Federal Information Processing Standards Publication
[5] ITU-T G.711 Pulse Code Modulation(PCM) of Voice Frequencies 1993
[6] ITU-T G.726 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM) 1997
[7] ITU-T G.723 Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s 1997
[8] ITU-T G.729 Coding of Speech At 8kbps Using Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-CELP) 1996.3
[9] ITU-T G.728 Coding of Speech at 16kbit/s Using Low-delay Code Excited Linear Prediction 1992
[10] Richard V. Cox. Speech Coding 1999.5.8.
[11] Jeremy Bradbury Linear Predictive Coding 2000.12.5
[12] 胡航 语音信号处理 哈尔滨工业大学出版社，2000年
[13] 戴沁云 胡捍英 第三代移动通信系统中的语音编码—自适应多码率AMR语音编码无线通信技术 2001年第3期