摘要
分析现有音视频传输的丢包恢复技术,结合前向纠错Reed-Solomon冗余编码及交织恢复丢包技术提出二项式概率模型。该模型根据接收端反馈的结果计算需要编码的冗余包个数,使用交织技术将音频冗余包与原始数据混合传输,有效节省了带宽资源。实验结果表明,该模型在网络拥塞的情况下,能根据实际情况产生足够的冗余数据包,使接收端收到数据后还原出原始数据并播放,提高了音视频的传输质量和播放质量。
Through analyzing the current packet loss recovery on voice and video transmission, this paper proposes a Binomial Probability ModeI(BPM) based on the Forward Error Correction(FEC) Reed-Solomon(RS) redundant coding and the interweaving packet loss recovery technique. BPM calculates the number of redundant packets according to the feedback from the receiver. By using interweaving technique to transmit mixed-packet of voice and redundancy, BPM can save bandwidth effectively. Experimental result shows that BPM is able to encode enough redundant packets under bad network condition, which ensures the receiver to decode the original data by using these packets and display it, and the equality of transmission and playing is improved greatly.
出处
《计算机工程》
CAS
CSCD
北大核心
2011年第1期98-100,共3页
Computer Engineering
基金
国家“973”计划基金资助项目(2003CB317003)
香港城市大学应用研发基金资助项目(9668009)
关键词
冗余编码
二项式概率
交织技术
redundant coding
binomial probability
interweaving technique