Design Of Synchronous Transmission System Of AVS-Based Video Embedded By Audio

With the rapid development of Internet,the advancement in processing capability of personal computers,and the development of audio-video compress techniques,the requirement for Internet audio-video services,especially for real-time audio and video stream such as video telephony,video conference and distance education,has been increasing continuously.Therefore,the study of Intemet real-time audio and video communication is becoming of great importance,regardless of in scientific research or in application field.Audio and video stream transmission is a complex and special problem,it is not only related with the development and conditions of the Internet,but also restrict with the own characteristic of audio and video coding standard.So aimed at the different conditions of Internet and coding methods,it should take different transmission mechanisms and the method of Internet adaptations.There is a few number of audio and video transmission system based on AVS at present.Achieving AVS transmission system is more actual value of promoting AVS application and industrialization process.Audio and video compression technology is one of the core issues of the audio-video communication technologies.This paper is based on AVS coding standard.AVS which also names' Audio and video coding standard',is the second generation source coding standard.Its intellectual property is owned by China.Compared to existing video compression standard and technology,AVS not only has more efficient compression performance,but also has good network adaptability.AVS divides codec into two layers:Video Coding Layer(VCL) and Network Abstraction Layer(NAL) in order to fit different network.VCL provide core high qualified video compression.NAL encapsulate compressed video and audio data based on specific network environment.For another,AVS has low implementation complexity and clear intellectual property.AVS can be applied widely in multimedia field.The choice of multimedia network transport protocol is one of the important technologies of transmission system.Compared to traditional transport protocol,RTP/RTCP protocol is a real-time stream transport standard put forward by IETF,it introduce timestamp and increasing sequence number for real-time data stream transport,and use RTCP to monitor QoS information,it can transport real-time audio and video data stream very well.The continuity of audio and video stream on the timeline requires network is real-time and high-bandwidth,at the same time,it allows the existence of a certain data error rate and data loss rate on the transmission.RTP does not have an independent transmission capability,It must be combined with the low-level network orotocol to complete the data transmission services.Audio and video don't have a strong correlation on the timeline,but the real-time characteristic of data is higher than the reliability of the data.So it can make use of RTP/RTCP protocol to packaging,packing,synchronizing audio and video stream above the UDP,and RTP/RTCP protocol can minimize the network transmission delay of the audio and video data.UDP is a connectionless datagram delivery service,it is less reliable than TCP,and it can not guarantee real-time video transmission quality of service.It requires RTCP to monitoring data transmission and service quality.But,UDP is less transmission delay than the TCP,it can make a good match to the audio and video steam.For the reason above,the use of RTP/RTCP/UDP as an audio and video data transfer protocol is a relatively good option.The mainly research work and results of this paper is listed as follows.1.This paper addresses the current situation on audio and video transmission and then gives a detailed analysis of coding technology and real-time transmission of audio and video. As for audio and video coding,this paper first studies the architecture of AVS,and then turns the focus on hierarchical structure,which is designed for transmission,finally,on the side of technology and complexity,by comparing the AVS and MPEG standards,it reaches the advanced nature of AVS.As for transmission technology,this paper first discusses that traditional network transport protocol is not suitable for real-time audio and video data transmission,and come to a conclusion that RTP/RTCP protocol is the best transport protocol, and then do some research on the RTP/RTCP protocol.RTP protocol is responsible for packing multimedia data and achieves real-time media streaming transmission.Its core content is the packet format.Each RTP packet includes the header and the payload.It only defines data structure of the header,and not restricts the size of the payload.RTP protocol does not guarantee the transmission reliability itself,and don't provide traffic congestion control mechanism.It uses RTCP to achieve traffic congestion control.Finally,this paper discusses the working mechanism of RTP protocol and the QoS control mechanism of audio and video transmission.2.It designs a RTP packet format for AVS,based on studying AVS coding standard and RTP/RTCP protocol in this paper.This part is one of the key elements in real-time transmission system.This part first studies the principle of RTP package,which makes the relationship between Payload data and the cost of the head optimization,reduces the packet loss rate as far as possible.And then studies the three RTP payload formats of AVS.Finally,it achieves a RTP packing strategy for AVS real-time transmission.3.Audio and video synchronization is a key problem in real-time transmission system.It uses the method of video embedded by audio in AVS to solve the problem of synchronization in this paper,and rises a new algorithm for video corresponding embedded by audio. Compared to traditional methods of synchronization,the method of synchronization in this paper is simple,and reducing the synchronization overhead.It has great advantages.4.According to the research result,the thesis designs an audio-video real-time transmission system based on RTP/RTCE Transmission system can be divided into the sending end and receiving end logically.From the realization of the function,it can be divided into audio-video capture module,audio-video encoding/decoding module and network transmission module.At the sending end,the collections of the original audio and video are compressed first,and then packing the audio and video data into packages,finally,writing the characteristics of the audio and video information into packages(timestamp,sequence number, type,etc),and then processed by UDP.RTP and UDP complete the task of transport layer protocol together.At the receiving end,after analyzing the packets of successful transmission through IP/UDP/RTP layers,and then put them into decoder,and playback synchronized.Audio-video capture module:video capture uses the VFW technology of Microsoft.It is a software group which has functions of capturing,editing,controlling and processing the video,image and audio data.Audio capture uses the technology of API,because the data of capture should be controlled.Audio-video encoding/decoding module:the thesis uses AVS codec-rm52c to realize concrete encoding/decoding.Rm52c is an official standard codec,and it is the most widely used open source codec.Network transmission module:RTP/RTCP is used to transport audio and video stream in system.The thesis uses the open-source library JPTPLIB to implement the RTP,design and realize the RTP packets of audio-video stream.Generally,we do some research and deduction in the thesis,which mainly talk about the AVS coding technology,RTP/RTCP transport mechanism,especially the audio and video transmission system.Based on the study,we design an audio-video real-time synchronized transmission system,and do some tests on the PC platform.Based on the test results,we can conclude that the scheme has achieved fine effect.However,there are still some problems need to be perfected.