CAJ | 학술논문

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。
제출료일충기우RTP/RTCP협의적음시빈동보산법。해산법장도체접수단적음시빈RTP수거포시간착영사지일개공공적절대시간축상，이측산출망락전수과정중적연시오차△Ti。연후，침대접수단음시빈파영불동적속도，측산수거포재완충구등대파영과정중적연시오차△Wi。최후이△Ti화△Wi적화작위총오차급출상응적파영책략。방진실험결과표명，해산법적RSME저우기타동보산법，대우음시빈적실보현상구유교명현적규정효과，용호대우파영질량적만의도대폭제승。
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.