ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and o...ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and optimizing methods of program code to reduce the speech processing delay time of G. 729 codec. Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway. Finally, the paper analyzes the performance results of ITU-T G. 729 codec based on TMS320C6201.展开更多
At medium or long distance （〉 10 kin） underwater acoustic speech communication, information transfer rate is constrained by the complicated, time varying channel and limited bandwidth. The bit rate of speech coding...At medium or long distance （〉 10 kin） underwater acoustic speech communication, information transfer rate is constrained by the complicated, time varying channel and limited bandwidth. The bit rate of speech coding is required to be as low as possible. The time delay of underwater acoustic wave propagation can be used for low bit rate speech coding. After investigating the Mixed Excitation Linear Prediction （MELP） standard and taking account of the auditory perceptual features, a variable and adjustable bit rate speech codec algorithm has been proposed, whose average bit rate is about 600 bps. The average Perceptual Evaluation of Speech Quality Mean Opinion Score （PESQ MOS） of synthesized speeches is about 2.8. It has been proved by the computer simulation and sea trial that the performance of the proposed algorithm is well and robust when bit error rate is no more than 10-3. The synthesized speech is vivid and intelligible, and keeps main individual characteristics of speaker.展开更多
QIM(Quantization Index Modulation,量化索引调制)隐写在标量或矢量量化时嵌入机密信息,可在语音压缩编码过程中进行高隐蔽性的信息隐藏,文中试图对该种隐写进行检测.文中发现该种隐写将导致压缩语音流中的音素分布特性发生改变,提出...QIM(Quantization Index Modulation,量化索引调制)隐写在标量或矢量量化时嵌入机密信息,可在语音压缩编码过程中进行高隐蔽性的信息隐藏,文中试图对该种隐写进行检测.文中发现该种隐写将导致压缩语音流中的音素分布特性发生改变,提出了音素向量空间模型和音素状态转移模型对音素分布特性进行了量化表示.基于所得量化特征并结合SVM(Support Vector Machine,支持向量机)构建了隐写检测器.针对典型的低速率语音编码标准G.729以及G.723.1的实验表明,文中方法性能远优于现有检测方法,实现了对QIM隐写的快速准确检测.展开更多
基金Supported by the National Natural Science Foundation of China under grant!69773046
文摘ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and optimizing methods of program code to reduce the speech processing delay time of G. 729 codec. Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway. Finally, the paper analyzes the performance results of ITU-T G. 729 codec based on TMS320C6201.
基金supported by the National Natural Science Foundation of China(61102152)
文摘At medium or long distance （〉 10 kin） underwater acoustic speech communication, information transfer rate is constrained by the complicated, time varying channel and limited bandwidth. The bit rate of speech coding is required to be as low as possible. The time delay of underwater acoustic wave propagation can be used for low bit rate speech coding. After investigating the Mixed Excitation Linear Prediction （MELP） standard and taking account of the auditory perceptual features, a variable and adjustable bit rate speech codec algorithm has been proposed, whose average bit rate is about 600 bps. The average Perceptual Evaluation of Speech Quality Mean Opinion Score （PESQ MOS） of synthesized speeches is about 2.8. It has been proved by the computer simulation and sea trial that the performance of the proposed algorithm is well and robust when bit error rate is no more than 10-3. The synthesized speech is vivid and intelligible, and keeps main individual characteristics of speaker.
文摘QIM(Quantization Index Modulation,量化索引调制)隐写在标量或矢量量化时嵌入机密信息,可在语音压缩编码过程中进行高隐蔽性的信息隐藏,文中试图对该种隐写进行检测.文中发现该种隐写将导致压缩语音流中的音素分布特性发生改变,提出了音素向量空间模型和音素状态转移模型对音素分布特性进行了量化表示.基于所得量化特征并结合SVM(Support Vector Machine,支持向量机)构建了隐写检测器.针对典型的低速率语音编码标准G.729以及G.723.1的实验表明,文中方法性能远优于现有检测方法,实现了对QIM隐写的快速准确检测.
