【作者】 王凡；

【导师】 郭更生；

【作者基本信息】 北京邮电大学 ， 通信与信息系统， 2008， 硕士

【摘要】 自适应多速率(AMR,Adaptive Multi Rate)语音编码是由3GPP(3rd Generation Partnership Proiect)制定的应用于第三代移动通信W—CDMA系统中的语音压缩编码。它以更加智能的方式解决信源和信道编码的速率分配问题,使得网络资源的配置和利用更加灵活和高效。它支持八种速率:12.2kb/s,10.2kb/s,7.95kb/s,7.4kb/s,6.70kb/s,5.90kb/s,5.15kb/s和4.75kb/s,此外,它还包括低速率的(1.80kb/s)背景噪声编码模式。实际的语音编码的速率取决于信道的条件,与采用固定的编码速率的语音编码方式相比,AMR语音编码则可根据信道的传输状况来自适应地选择一种最佳编码模式(以比特率来区分)进行编码传输。传统的VoIP系统都采用固定编码速率的语音编码器,这种固定速率的话音编码器不能够根据信道状况自适应的调整编码速率,因此在网络状况不好的条件下性能就会严重下降。通过分析VoIP系统和自适应多速率语音编码技术的特点,本论文提出了一种将自适应多速率语音编码技术应用于VoIP系统的方法,并通过仿真验证了在不同网络环境下的这种VoIP系统中的AMR技术的有效性。在VoIP系统中实现自适应多速率语音编码后,本文进一步对AMR编码算法中舒适噪声生成、话音激活检测、丢帧隐藏机制三个模块分别进行了优化,系统仿真表明,这种优化后的AMR语音编码算法能够进一步提升VoIP系统的性能,将AMR编码器的编码时延降低20%。最后我们得出结论,本文提出的AMR优化算法可以很好的与VoIP系统相结合,提供较好的话音质量。更多还原

【Abstract】 Traditionally, fixed bit-rate speech coding is used in VoIP system. This technology can not adaptively switch its bit-rate due to network condition. So we select AMR speech coder.The AMR speech coder consists of the multi-rate speech coder, a source controlled rate scheme including a voice activity detector and a comfort noise generation system, and an error concealment mechanism to combat the effects of transmission errors and lost packets. The multi-rate speech coder is a single integrated speech codec with eight source rates from 4.75 kbit/s to 12.2 kbit/s, and a low rate background noise encoding mode. The speech coder is capable of switching its bit-rate every 20 ms speech frame upon command. Compared with fixed bit-rate speech coder AMR can provide more flexibility.This paper focuses on how to efficiently merge VoIP system and Adaptive Multi-Rate (AMR) technology. The general knowledge of VoIP system and AMR were introduced in the first chapter. Based on the analyses of these two technologies, we proposed a method to merge them into one system in order to overcome the drawback of fixed bit-rate voice coding in VoIP system. By simulation we proved that VoIP system based on AMR speech coder can achieve higher performance gain than that based on fixed bit-rate coder.Besides, we do optimization on comfort noise generation system, voice active detector and error concealment mechanism to enhance the performance of the system. Simulation shows that the optimized AMR algorithm in VoIP system reduces the time latency to about 80% compared to standard AMR based VoIP system. Therefore, our proposed optimal AMR speech coder can work well in VoIP system and the performance gain is huge compared with fixed bit-rate speech coder.更多还原