一种基于RTP/RTCP协议的音视频同步算法研究

RESEARCH OF AUDIO AND VIDEO SYNCHRONIZATION ALGORITHM BASED ON RTP/RTCP PROTOCOL

提出了一种基于RTP/RTCP协议的音视频同步算法。该算法将到达接收端的音视频RTP数据包时间戳映射至一个公共的绝对时间轴上,以测算出网络传输过程中的延时误差△Ti。然后,针对接收端音视频播映不同的速度,测算数据包在缓冲区等待播映过程中的延时误差△Wi。最后以△Ti和△Wi的和作为总误差给出相应的播映策略。仿真实验结果表明,该算法的RSME低于其他同步算法,对于音视频的失步现象具有较明显的纠正效果,用户对于播映质量的满意度大幅提升。

An algorithm of audio and video synchronization control based on RTP/RTCP protocol is proposed. The core idea of this algorithm was to map the timestamp of the audio frame and the video frame to a public absolute time axis at the receiver so as to compute the delay error△Ti during the transfer process across the internet. Furthermore, the delay error △Wi of the packet which is waiting in the buffer was obtained according to different speed of audio and video broadcast at the receiving end. Finally, we use△Ti and△Wi as the total error and give the corresponding broadcast strategy. Simulations results show that RSME of this algorithm is less than other synchronization algorithm. The method can well correct audio and video out of synchronization. The users’ feeling of quality greatly enhanced.