A kind of ensuring method in Web conference sound intermediate frequency transmission real-timeTechnical field
The invention discloses a kind of ensuring method of Web conference sound intermediate frequency transmission real-time.Under the current network bandwidth condition of limited, at first guarantee the real-time of conference audio transmission, the Bandwidth Dynamic ground according to remainder determines application program and required data collection rate and the transfer rate of video again.
Technical background
Along with computer and development of Communication Technique, the multi-media network meeting has become the focus of research.And in this technology, multiple different media informations, the multi-functional data transmission between a plurality of convention goers has increased the complexity of Web conference technology: at first, and all audio frequency, application program is shared and video capability must realize; The more important thing is that as a conference system, the real-time of audio transmission guarantees to be absolutely necessary, because the delay of voice is unacceptable.
Because the functional diversity of multi-media network conference system, its to the requirement of the network bandwidth also than higher.Under the situation of only utilizing voice conferencing, most current network configuration can satisfy bandwidth requirement, but after the application program sharing functionality started, its requirement meeting to bandwidth increased greatly.How the transmission of balance different pieces of information is the problem that must solve of netmeeting.
The transmission of carrying out audio frequency, application program shared data and video on IP network must guarantee the continuity of transmitting data stream to that is to say real time of data transmission.The requirement of real time communication with those at a high speed but non real-time requirement of communicating by letter is inequality.For traditional internet, applications, as file transmission, Email, client-server application or the like, the performance index that we are concerned about are throughput and time delay normally.We also have requirement to reliability in addition, have therefore taked relevant mechanism to guarantee that data are not lost, damage or out-of-sequence in transmission.Different therewith, use in real time and more be concerned about problem regularly.Under most situations, all require data to send the other side to the constant rate of speed that equals transmission speed.
The QoS relevant with real-time speech communicating (service quality) comprises bandwidth, time-delay and three aspects of Loss Rate.Aspect bandwidth, be unlikely to too poor for making voice quality, need guarantee that under any circumstance voice transfer can both obtain certain bandwidth.The end-to-end time delay of real-time voice transmission requirement can not be too big, and general time delay is in 50ms, and people's ear can't be differentiated substantially; Time delay is between 50～200ms, and people's ear can feel that the interval of speech exists, to the understanding of semanteme with exchanges influence and not quite; In case time delay surpasses 200ms, will have a strong impact on speech quality.Than time delay, delay variation is bigger to the influence of voice quality.The rhythm that fixing time delay may just disturb people to talk, and delay variation will produce interruption at random between conversation.Packet loss will reduce the voice quality of receiving terminal, and serious packet loss will cause voice to understand.Therefore, IP-based real-time speech communicating needs packet loss less than certain value.
In meeting, the information interchange of the overwhelming majority is finished by voice.So guarantee that the real-time of voice is first tasks of conference system.
Existing netmeeting is recognized this problem, and most of netmeeting can check all that before the participant adds meeting its bandwidth that is connected with server is to guarantee the meeting quality.But they have following shortcoming:
1. bandwidth detection was carried out before adding meeting, but the network bandwidth is dynamic change, and at the different time of meeting, the bandwidth that connects with server may be different.
2. audio frequency, the shared data of application program and video acquisition rate are invariable, neatly the dynamic adjustments with the variation of bandwidth.Under the situation that the network bandwidth diminishes, constant data can not in time be transmitted and can produce time-delay, influence the quality of meeting sound intermediate frequency transmission.
The subject matter of current all conference system all is that the assurance of real-time is not enough, to the network bandwidth require too high, carry out the network teleconference and desktop when sharing the time-delay phenomenon very obvious, had a strong impact on the quality of Web conference.
Summary of the invention
The objective of the invention is: at the deficiencies in the prior art, provide a kind of transfer of data stable, real-time, audio frequency, application program are shared, and video playback is effective, the method that the IP network meeting real-time that system cost is low guarantees.
In order to solve the problems of the technologies described above, the technical solution used in the present invention is: voice data, the application program shared data, also have video data to make up data channel respectively, in data channel separately, adopt independent thread carry out data collection, compression, packing, transmission, receive, unpack, processing such as decompress(ion); Provide enough bandwidth to guarantee the real-time of conference audio transmission to voice-grade channel; Other has a bandwidth detection thread to monitor acquisition rate and the transfer rate of current bandwidth with decision application program shared data and video data in real time.
Client is when adding meeting, and the bandwidth detection thread obtains its available bandwidth that is connected with server, and this available bandwidth is by audio frequency, and the application program shared data is shared with video channel.The real-time of voice is assurances that conferencing over ip normally carries out, and a standing part is taken by voice-grade channel in the available bandwidth; Remaining bandwidth is shared with video by the application program shared data, and this remaining bandwidth has determined the acquisition rate of application program shared data and video, can be not excessive and cause network congestion to guarantee the image data amount.
Carry out in the process in meeting, the bandwidth detection thread also can periodically detect the available bandwidth that is connected with server, under the situation that a standing part is taken by voice-grade channel in guaranteeing available bandwidth, remaining bandwidth has determined the acquisition rate and the transfer rate of application program shared data and video.
In technique scheme, the present invention uses 3 different threads transmitting audio data bag respectively, application program shared data bag and video packets of data by adopting 3 separate passages.The data channel of each thread can be occupied different bandwidth respectively, and its allocated bandwidth is by a real-time bandwidth detection line thread management.The audio frequency thread is assigned enough bandwidth to guarantee the real-time of voice transfer; Application program threads and video thread determine its acquisition rate according to the bandwidth of its distribution, thereby guarantee the real-time and the synchronism of transmission, can reduce some its transmission qualities under the not enough situation of bandwidth.After all, under band-limited situation, the assurance of voice quality is primary.
The present invention for System Hardware Requirement lower, can save system cost, the scope of application is wider; Current network configuration there is not specific demand.Therefore, relative prior art, characteristics such as the present invention has practical, and cost is low, and sound, video Data Transmission stability and real-time are good, and synchronous effect is good, and the effect of meeting is better.
Embodiment
Below in conjunction with Figure of description and specific embodiment the present invention is described in further detail.
The invention provides is a kind of ensuring method of Web conference sound intermediate frequency transmission real-time, in order to overcome transmission data imbalance, the network condition instability, voice time-delay the and influence quality of meeting avoids that system cost is too high to cause problems such as domestic consumer is inapplicable the while again.
Present embodiment is specially: set up separate voice-grade channel and share application data channel and video channel, and with a thread dynamic monitoring and distribution network bandwidth, share according to Bandwidth Dynamic ground decision application program, the data collection rate of video is to guarantee the real-time of data.
Wherein the bandwidth of voice-grade channel will preferentially guarantee, 8kbps at least, and the bandwidth of voice-grade channel can be issued to 48kbps in situation about allowing.The most important factor of the quality of meeting is the real-time of voice transfer.
Carry out in the process in meeting, the cycle that the bandwidth detection thread detects the available bandwidth that is connected with server is 10 seconds.The bandwidth detection thread dynamically distributes bandwidth to give different data channel according to current situation.Because of the bandwidth of network can not be unalterable in conference process, periodic bandwidth detection and suitable distribution are necessary fully.
Application program is shared the bandwidth decision of the data collection rate of thread by its distribution, according to bandwidth reduce be respectively:
| Resolution: 1024x768 frame number/second: 2+FPS color: 24bits | Resolution: 1024X768 frame number/second: 1FPS color: 8bits | Resolution: 800X600 frame number/second: 0.25FPS color: 4bits |
The data collection rate of video thread is determined by the surplus bandwidth of institute after guaranteeing audio frequency and application program shared data, is respectively: 194kbps, 38kbps, 0kbps.Transmitting video data not under the situation of bandwidth wretched insufficiency.Conference system different with audio-video Play System, under the network configuration condition of limited, sacrifice some picture qualities and guarantee conferencing data synchronously and real-time be very important.