摘要
当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。
Basic jitter buffering algorithms can work well only when no spike delay exists in the IP networks. Here, an adaptive jitter buffering algorithm is presented to promote the quality of voice communication with spike delay. The algorithm controls the playout time by controlling the position of voice packet in the buffer queue, which is quite different from the mode of predicting the playout time of the basic algorithms that much easily generates voice gap. The control target of packet loss rate can be extended to 20%. Perceptual evaluation of speech quality（PESQ） is applied to assess the speech quality in the simulation. The practical application can also indicate the effects of voice quality promotion.
出处
《信号处理》
CSCD
北大核心
2006年第3期417-421,共5页
Journal of Signal Processing
基金
本课题为国家自然科学基金资助项目(69574026)部分资助。
关键词
语音质量
PESQ
突发大时延
丢包率
voice quality
jitter buffering
spike of delay
rate of packet loss rate
adaptive algorithm
PESQ
