【作者】 贾龙涛；

【导师】 鲍长春；

【作者基本信息】 北京工业大学 ， 电路与系统， 2004， 硕士

【摘要】 摘要 近年来，随着信息产业的不断发展，网络建设无论是有线网络还是无线网络都发展迅猛。IPv6 的呼之欲出，“三网合一”（数据网，电话网，有线电视网）地即将出现，都说明网络语音传输系统（如 VoIP）会成为人们生活的一部分，而与此相关的语音编解码技术﹑网络传输技术，以及有线网络和无线网络融合技术也成为当前研究的热点。 目前，所有的网络语音电话系统（VoIP）都采用固定速率进行传输，这使得网络分组丢失，特别是连续分组丢失无法避免，导致了严重的语音质量下降，这也是阻碍网络语音电话广泛应用的一个瓶颈。 本文针对上述难以解决的问题，对各种方案进行了研究分析，提出了一种有效的自适应网络语音传输方案。首先，本文对于自适应多速率语音编解码算法（AMR）进行了深入的研究,针对 AMR 的编码特点对其进行了改进，给出了一种抗分组丢失的主次型编解码器。然后，提出一种基于 AMR 的抗分组丢失的网络语音传输系统。最后，应用 NS(network simulation)和 C 语言对整个系统进行了仿真实现。 通过计算机仿真实验证明，本文研究的网络语音传输系统在网络分组丢失、平均延迟和主观听觉方面明显优于传统的网络语音电话系统。更多还原

【Abstract】 In recent years, with the development of information technology, the networkconstruction including the wireless and the fixed phone system is developed rapidly.The IP version 6 will be applied in IP network and all the traffic will be integratedtogether including the data network, the CATV network and the fixed phone network.It can be said that voice over IP will be used widely, and related technologies likespeech coding, network transmission and integration between wireless and fixedphone network have become the focus. At present, most of the VoIP systems are based on the speech codec with a fixedrate, which makes packet-loss serious, especially for continuous packet-loss.Packet-loss will cause degradation of the speech quality. This is a bottleneck, whichmakes the application of VoIP limited. In this thesis, an efficient and adaptive network speech communication systemwas proposed based on above problems through analyzing all kinds of existed VoIPspeech transmission system. First, the AMR codec was investigated carefully and anew main-sub codec was proposed according to the character of AMR codec. Second,a network speech communication system for preventing packet-loss was given basedon AMR. Finally, the system is simulated by NS and C language. The computer simulation and subject listening test show that the proposedsystem outperforms traditional IP system over the average packet-loss, average delayand subjective listening quality.更多还原