【作者】 花东来；

【导师】 姚萌；

【作者基本信息】 华东师范大学 ， 通信与信息系统， 2005， 硕士

【摘要】 VoIP(Voice-Over-IP)是当前非常流行的一种基于在IP网络上传输的语音媒体流的技术。它以通信成本低和支持的业务丰富等优点,迅速发展起来。然而,由于IP网络的无法保证服务质量(QOS:Quality of Service)的固有特性。在语音包的接收端,由于网络路由,网络延时,网络拥塞或网络发生错误造成的语音包丢失,延迟到达或到达乱序都时有发生。语音包的丢失将带来在接受端语音解码后的话音实时回放质量的损失,劣化。由于语音业务是一种对实时性和交互性要求很高的业务,所以不可能利用丢失数据包重传的机制来改善性能。包丢失修复技术就是一种试图解决语音的实时性要求和改善语音听觉感受的技术。无论是基于发送和接收端或者单纯基于接收端的修复算法,都是产生重构包来替换丢失部分的语音,改善听觉质量。现有的国际标准对包丢失修复技术有很多的研究,如ANSI TI.521(Annex B)及ITU-T Rec.G711(Appendix I),它们都是针对于G711语音编码器的,基于接受端的修复算法,有着较好的性能,但它们都是利用历史数据包,并未考虑应用后继数据包来重构丢失部分的语音。另外,在重构语音和真实语音拼接过程中,仍然有断续现象发生,特别是在从浊音到清音及音节间过渡时刻,这种断续现象比较明显。本论文的目的是对改进的语音丢失修复算法的研究。在后继包有效的前提下,同时应用历史包和后继包的信息。并且采用基于波形相似重叠相加算法来实现时间尺度的调整,重构丢失的语音部分。最后,在VoWIFI(VoIP Over WIFI)的实验系统上实现了这个算法,在模拟随机丢包率从1%到35%的测试环境下,算法表现了优越的性能和健壮性。更多还原

【Abstract】 Voice-Over-IP (VoIP) uses packet transmission of speech over the Internet (IP network). However, at the receiving end, packets are missing due to network delay, network congestion (Jitter) and network errors. This packet loss degrades the quality of speech at the receiving end of a voice transmission system in an IP network . Since the voice transmission is a real-time process, the receiver cannot request for retransmission of the missing packets. Concealment algorithms , either transmitter or receiver based, are used to replace these lost packets . The packet loss concealment (PLC) techniques described in the standards ANSI T1.521 (Annex B) and ITU-T Rec. G. 711 (Appendix I),have good performance, but these algorithms so not use subsequent packets for reconstruction. Furthermore, there are discontinuities between the reconstructed and the subsequent packets, especially at the transitions from voiced to unvoiced and phoneme-to-phoneme.The goal of this work is to develop an improved PLC algorithm, using the subsequent packet information when available. For this, we use the Time-Scale Modification (TSM) technique based on Waveform Similarity Over-Lap Add (WSOLA) to reconstruct the dropped or lost packets. The algorithm looks ahead for subsequent packets., If these packets are not available for reconstruction, algorithm uses information from past packetSo Subjective tests show that the proposed method improves the reconstructed speech quality significantly,,By implementing the algorithm on Hardware Development Kit of Voice-Over-IP over WIFI , under the manually simulate the random packet loss ratio from 1% to 35%, this algorithm shows excellent and robust performance.更多还原