中速率语音编解码算法在VoIP系统中的定点DSP实现
|
摘要
VoIP业务是当前计算机网络技术和通信技术研究的热点之一,也是因特网增长最快的业务之一,指的是以数据封包的形式在IP分组网络的环境下进行语音信号的传输。与传统的电路交换网络相比,IP分组网络存在带宽资源有限,丢包和延时抖动的问题,因此需要研究和实现适合于分组网络传输环境的语音编解码算法,来完成VoIP中的终端编解码功能。通过对各种语音算法的分析和研究发现,ILBC、Speex等语音编解码算法不仅编码速率低,而且有多种模式可以根据网络状况灵活选择,同时增加了丢包隐藏,去延时抖动等模块,非常适用于因特网上的语音传输。另外ILBC、Speex算法不需要交专利费,因此有很大的商业应用价值。
根据对语音编码器的分类标准,编码速率介于4.6kb/s～24kb/s的语音编码器称为中速率语音编码器,因此ILBC,G729以及Speex大部分模式下的编码算法均为中速率语音编码算法。课题以研究和实现以ILBC为主的适合于分组网络的几种中速率语音编解码算法为目标,借助PalmADSP、Visual C++等仿真和开发软件,经过了由浮点C语言代码到定点C语言代码,再到定点DSP代码的转换过程,并对代码进行了系统的测试和优化,最后将代码嵌入到DSP芯片中,完成了算法向DSP芯片的搬移。工程实践中主要解决了以下两个问题:一、定点化过程中,如何选择合适的定标值以保证数据的动态范围和精度,二、在芯片的数据存储空间和程序存储空间有限的情况下,如何对代码进行系统的优化以提高程序执行效率,压缩数据和代码占用的空间。最终,课题通过ILBC等算法的定点化工作总结出了一套适用于各种语音算法的定点化方法,并通过具体的工程实践提出了针对DSP开发和应用的代码转换和优化方法。在AR168G话机上的实际通话测试结果表明,课题中实现的几种语音算法能很好地运用于VoIP系统,对各种网络状况具有很好的适应性,获得了良好的通话质量。
VoIP is one of the hottest topics of computer network and communication technologies and one of the fastest growing Internet businesses at present. It is to transport speech signals in the form of packets through the IP packet switched network. Different from traditional circuit switched network, there are several problems exist in the packet switched network, such as limited bandwidth, packet loss and delay jittering, so we need to analyze and realize audio coding/decoding algorithms which are more suitable for the packet switched network, add some extra modules to deal with different network situations and implement these algorithms on IP phones in VoIP system. By analyzing different kinds of audio algorithms, we found that audio codecs such as ILBC and Speex had a low bit-rate and several modes to be selected according to different network situations; they also add some extra modules such as packet loss concealment and de-jittering. So these codecs are very suitable to transmit speech signals through internet. In addition, ILBC and Speex algorithms are open source and free, so they have great business value.
According to the classifying standard of audio codecs, codec with bit-rate between 4.6kb/s～24kb/s is called middle-bitrate codec. So ILBC, G729 and most modes of Speex are middle-bitrate codecs. In this paper we aimed at analyzing and realizing middle-rate audio codecs such as ILBC, G729 and Speex which were better for the packet switched network with the simulating and developing tools such as PalmADSP, and Visual C++. Through the process of converting float point C codes to fixed point C codes, translating C codes to DSP codes, embedding the codes into DSP chips and executing system testing and optimization, we finally implemented these algorithms on DSP chips. During practical work, we solved two problems: how to choose a proper fixed point transforming scheme to assure data range and precision and how to enhance the efficiency of code execution while compressing the data and code space as much as possible for the limited data and program memory of DSP chip. Finally in this paper, we summarized some general float-point codes to fixed-point codes transforming methods through the fixed-point code transforming work of ILBC algorithm and put forward some useful C codes to DSP codes translating and optimizing methods for the development of DSP applications. Results of implementation on AR168G phones indicate that audio algorithms realized in the VoIP system could adapt to different network situations and gain good communication quality.
根据对语音编码器的分类标准,编码速率介于4.6kb/s～24kb/s的语音编码器称为中速率语音编码器,因此ILBC,G729以及Speex大部分模式下的编码算法均为中速率语音编码算法。课题以研究和实现以ILBC为主的适合于分组网络的几种中速率语音编解码算法为目标,借助PalmADSP、Visual C++等仿真和开发软件,经过了由浮点C语言代码到定点C语言代码,再到定点DSP代码的转换过程,并对代码进行了系统的测试和优化,最后将代码嵌入到DSP芯片中,完成了算法向DSP芯片的搬移。工程实践中主要解决了以下两个问题:一、定点化过程中,如何选择合适的定标值以保证数据的动态范围和精度,二、在芯片的数据存储空间和程序存储空间有限的情况下,如何对代码进行系统的优化以提高程序执行效率,压缩数据和代码占用的空间。最终,课题通过ILBC等算法的定点化工作总结出了一套适用于各种语音算法的定点化方法,并通过具体的工程实践提出了针对DSP开发和应用的代码转换和优化方法。在AR168G话机上的实际通话测试结果表明,课题中实现的几种语音算法能很好地运用于VoIP系统,对各种网络状况具有很好的适应性,获得了良好的通话质量。
VoIP is one of the hottest topics of computer network and communication technologies and one of the fastest growing Internet businesses at present. It is to transport speech signals in the form of packets through the IP packet switched network. Different from traditional circuit switched network, there are several problems exist in the packet switched network, such as limited bandwidth, packet loss and delay jittering, so we need to analyze and realize audio coding/decoding algorithms which are more suitable for the packet switched network, add some extra modules to deal with different network situations and implement these algorithms on IP phones in VoIP system. By analyzing different kinds of audio algorithms, we found that audio codecs such as ILBC and Speex had a low bit-rate and several modes to be selected according to different network situations; they also add some extra modules such as packet loss concealment and de-jittering. So these codecs are very suitable to transmit speech signals through internet. In addition, ILBC and Speex algorithms are open source and free, so they have great business value.
According to the classifying standard of audio codecs, codec with bit-rate between 4.6kb/s～24kb/s is called middle-bitrate codec. So ILBC, G729 and most modes of Speex are middle-bitrate codecs. In this paper we aimed at analyzing and realizing middle-rate audio codecs such as ILBC, G729 and Speex which were better for the packet switched network with the simulating and developing tools such as PalmADSP, and Visual C++. Through the process of converting float point C codes to fixed point C codes, translating C codes to DSP codes, embedding the codes into DSP chips and executing system testing and optimization, we finally implemented these algorithms on DSP chips. During practical work, we solved two problems: how to choose a proper fixed point transforming scheme to assure data range and precision and how to enhance the efficiency of code execution while compressing the data and code space as much as possible for the limited data and program memory of DSP chip. Finally in this paper, we summarized some general float-point codes to fixed-point codes transforming methods through the fixed-point code transforming work of ILBC algorithm and put forward some useful C codes to DSP codes translating and optimizing methods for the development of DSP applications. Results of implementation on AR168G phones indicate that audio algorithms realized in the VoIP system could adapt to different network situations and gain good communication quality.
引文
[1]肖丁,徐六通,何拥军,VoIP:技术、应用与主要问题,《现代电信科技》,第10期,1999.10
[2]Uyless Black(温斌译),VoIP:IP语音技术,北京:机械工业出版社,2000.05,7-19页
[3]秦建飞,唐晓燕,贾国锋,SIP协议在VoIP中的实现,电力系统通信,Vol.27,No.163.2005.05
[4]ITU-T Recommendation G723.1,Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3kbit/s,Mar,1996.
[5]ITU-T Recommendation G.728,Coding of speech at 16kbit/s using low-delay code excited linear prediction,Sep 1992.
[6]ITU-T Recommendation G.729-Annex A,Reduced complexity 8kbit/s CS-ACELP speech codec,Nov,1996.
[7]L.R.Rabiner and R.Schafer,Digital processing of speech signals,Englewood Cliffs,NJ:Prentice Hall,1978
[8]A.Kataoka,T.Moriya,S.Hayashi.A 8bit/s Conjugate Structure CELP(CS-CELP)Speech Coder,IEEE Transactions On Speech and Audio Processing,1999,4(6):Page401-411.
[9]IETF:RFC 3952- Real-time Transport Protocol(RTP)Payload Format for internet Low Bit Rate Codec(ILBC)Speech,2004
[10]杨行峻,迟惠生,唐昆,等,语音信号数字处理,北京:电子工业出版社,1995
[11]Global IP Sound,ILBC White Paper,Oct,2004
[12]IETF:RFC 3951-Internet Low Bit Rate Codec.Dec,2004
[13]ADI Corporation,ADSP2181 User Manual,2001
[14]ADI Corporation,ADSP-218x DSP Hardware Reference(Rev 1.0),2001
[15]胡航,语音信号处理,哈尔滨:哈尔滨工业大学出版社,2000
[16]J.D.Marker,Jr A.H.Gray,Linear Prediction of Speech,Sprintger-Verlag,1976
[17]鲍长春,低比特率数学语音编码基础,北京:北京工业大学出版社,2001
[18]P.KaBal and R.P.Ramachandran,The Computation of Line Spectral Frequencies Using Chebyshev Polynominals,IEEE,Vol.ASSP-34,NO.6,1986,Page 1419-1426
[19]J.Rothweiler,A root finding algorithm on line spectral frequencies,ICASSP-99,vol.2,1999,Page661-664
[20]S.V.Andersen et al,ILBC-A Linear Predictive Coder with Robustness to Packet Losses,IEEE Workshop on Speech Coding 2002,Tsukuba,Ibaraki,Japan,October 2002
[21]J.chen,A.Gersho,Adaptive postfiltering for quality enhancement of coded speech,IEEE Trans.Speech Audio Process.Vol.3,No.1,1995,Page59-71,
[22]V.Ramamoorthy,N.s.Jayant,Enhancement of ADPCM speech by adaptive postfiltering,AT&T Bell Labs.Tech.J,1984,Page 1465-1475,
[23]郭廷廷,李敬,ILBC编解码算法及其在VoIP中的应用,电子技术应用,2006.07
[24]IETF:RFC 3952- Real-time Transport Protocol(RTP)Payload Format for internet Low Bit Rate Codec(ILBC)Speech,2004
[25]李亚民,计算机组成与系统结构,北京:清华大学出版社,2004
[26]Erick L.Oberstar,"Fixed Point Representation and Fractional Math," Oberstar Consulting,2005
[27]ADI Corporation.Using the ADSP-2100 Family Volume 1(Rev 1.0),1990
[28]张雄伟,陈亮,徐光辉,DSP集成开发与应用实例,北京:电子工业出版社,2002
[29]徐科军,黄云志,定点DSP的原理、开发与应用,北京:清华大学出版社,2002
[30]张雄伟,曹铁勇,DSP芯片的原理与开发应用(第二版),北京:电子工业出版社,2000
[31]A.G.M Cilio and H.Corporaal,Floating Point to Fixed Point Conversion of C Code,Delft University of Technology:Computer Architecture and Digital Techniques Dept,2004
[32]C.H.Wu and J.H.Chen,A novel two-level method for the computation of LSP frequencies using a decimation-in-degree algorithm,IEEE Trans.Speech Audio Processing,vol.5,1997,Page106-115,
[33]宗孔德,多采样率数字信号处理,北京:清华大学出版社,1996
[34]ADI Corporation,ADSP-218x DSP instruction Set Reference,2004
[35]ADI Corporation,VisualDSP++2.0 User's Guide for ADSP21xx DSPs,2001
[2]Uyless Black(温斌译),VoIP:IP语音技术,北京:机械工业出版社,2000.05,7-19页
[3]秦建飞,唐晓燕,贾国锋,SIP协议在VoIP中的实现,电力系统通信,Vol.27,No.163.2005.05
[4]ITU-T Recommendation G723.1,Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3kbit/s,Mar,1996.
[5]ITU-T Recommendation G.728,Coding of speech at 16kbit/s using low-delay code excited linear prediction,Sep 1992.
[6]ITU-T Recommendation G.729-Annex A,Reduced complexity 8kbit/s CS-ACELP speech codec,Nov,1996.
[7]L.R.Rabiner and R.Schafer,Digital processing of speech signals,Englewood Cliffs,NJ:Prentice Hall,1978
[8]A.Kataoka,T.Moriya,S.Hayashi.A 8bit/s Conjugate Structure CELP(CS-CELP)Speech Coder,IEEE Transactions On Speech and Audio Processing,1999,4(6):Page401-411.
[9]IETF:RFC 3952- Real-time Transport Protocol(RTP)Payload Format for internet Low Bit Rate Codec(ILBC)Speech,2004
[10]杨行峻,迟惠生,唐昆,等,语音信号数字处理,北京:电子工业出版社,1995
[11]Global IP Sound,ILBC White Paper,Oct,2004
[12]IETF:RFC 3951-Internet Low Bit Rate Codec.Dec,2004
[13]ADI Corporation,ADSP2181 User Manual,2001
[14]ADI Corporation,ADSP-218x DSP Hardware Reference(Rev 1.0),2001
[15]胡航,语音信号处理,哈尔滨:哈尔滨工业大学出版社,2000
[16]J.D.Marker,Jr A.H.Gray,Linear Prediction of Speech,Sprintger-Verlag,1976
[17]鲍长春,低比特率数学语音编码基础,北京:北京工业大学出版社,2001
[18]P.KaBal and R.P.Ramachandran,The Computation of Line Spectral Frequencies Using Chebyshev Polynominals,IEEE,Vol.ASSP-34,NO.6,1986,Page 1419-1426
[19]J.Rothweiler,A root finding algorithm on line spectral frequencies,ICASSP-99,vol.2,1999,Page661-664
[20]S.V.Andersen et al,ILBC-A Linear Predictive Coder with Robustness to Packet Losses,IEEE Workshop on Speech Coding 2002,Tsukuba,Ibaraki,Japan,October 2002
[21]J.chen,A.Gersho,Adaptive postfiltering for quality enhancement of coded speech,IEEE Trans.Speech Audio Process.Vol.3,No.1,1995,Page59-71,
[22]V.Ramamoorthy,N.s.Jayant,Enhancement of ADPCM speech by adaptive postfiltering,AT&T Bell Labs.Tech.J,1984,Page 1465-1475,
[23]郭廷廷,李敬,ILBC编解码算法及其在VoIP中的应用,电子技术应用,2006.07
[24]IETF:RFC 3952- Real-time Transport Protocol(RTP)Payload Format for internet Low Bit Rate Codec(ILBC)Speech,2004
[25]李亚民,计算机组成与系统结构,北京:清华大学出版社,2004
[26]Erick L.Oberstar,"Fixed Point Representation and Fractional Math," Oberstar Consulting,2005
[27]ADI Corporation.Using the ADSP-2100 Family Volume 1(Rev 1.0),1990
[28]张雄伟,陈亮,徐光辉,DSP集成开发与应用实例,北京:电子工业出版社,2002
[29]徐科军,黄云志,定点DSP的原理、开发与应用,北京:清华大学出版社,2002
[30]张雄伟,曹铁勇,DSP芯片的原理与开发应用(第二版),北京:电子工业出版社,2000
[31]A.G.M Cilio and H.Corporaal,Floating Point to Fixed Point Conversion of C Code,Delft University of Technology:Computer Architecture and Digital Techniques Dept,2004
[32]C.H.Wu and J.H.Chen,A novel two-level method for the computation of LSP frequencies using a decimation-in-degree algorithm,IEEE Trans.Speech Audio Processing,vol.5,1997,Page106-115,
[33]宗孔德,多采样率数字信号处理,北京:清华大学出版社,1996
[34]ADI Corporation,ADSP-218x DSP instruction Set Reference,2004
[35]ADI Corporation,VisualDSP++2.0 User's Guide for ADSP21xx DSPs,2001