CN1468001A - Media flow self-adapting transmission method based on internet - Google Patents
Media flow self-adapting transmission method based on internet Download PDFInfo
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Abstract
The invention discloses a media stream self-adapting transporting method based on the internet comprising, a. establishing RTP/UDP/IP protocol structure, wherein the data plane of the protocol structure is in charge of using the media video stream and audio stream packaged by RTP, while the control plane of the protocol structure transmits the control instructions simultaneously using the RTCP feed back network information and RTSP, establishing a congestion control structure based on the transmitting end that can be applied both in the real time coding and in the video frequency transfer storage. The invention can realize the best playback quality possible on the user end under the changes of the network operation.
Description
Technical field
The present invention relates to a kind of Media Stream adaptive transmission method based on Internet (internet), more particularly, be about MPEG-4 stream adaptive transmission method based on Internet, make under the situation that the Internet service conditions constantly changes, can dynamically adjust the transmission of MPEG-4 code stream, make client obtain play quality as well as possible.
Background technology
The backbone network basis that stream media technology realizes is the Internet net, and the Internet net provides a kind of " doing one's best " (best effort) service support does not promptly provide QoS (service quality Quality of Service) to guarantee to the transmission data.The Internet net is the heterogeneous network of an opening, and for the transmission of multi-medium datas such as video, audio frequency, its subject matter is to have propagation delay time, delay variation and data-bag lost.The basic reason of these problems is network bandwidth deficiencies, and bandwidth be change, have a unpredictability.Therefore, Streaming Media transmission problem based on the Internet net can be summed up as: how when bandwidth, to become the transmission over networks media data of (limited in other words conj.or perhaps), which kind of adopts dynamically adjust strategy according to the real-time network information and just can make client have media play quality as well as possible?
Obviously, realize best transmission quality, must solve: adaptive coding, congested and ERROR CONTROL problem.From the consideration of ERROR CONTROL angle, proposed several error control methods in the MPEG-4 standard: sync mark again, data are cut apart, error resilience tool such as data are recovered, error concealing.But these instruments are suitable for wireless transmission, and different with the error code problem of wireless transmission be in the Internet net, mainly to be packet loss and delay problem.The main means of the ERROR CONTROL of solution under Internet net environment are forward error corrections (FEC, Forward Error Correction) and retransmit.Therefore, once the Internet image that the diminishes fast transmission (FLIIT that the someone proposed, Fast Loss Internet ImageTransmission) method has been eliminated and has been retransmitted the time delay of bringing, also can reach simultaneously effect with the TCP transmission, in fact it is a kind of information source/channel conjunction with forward error correction coding techniques, but this method has been introduced a large amount of transport overheads, is not suitable for the transmission of MPEG-4.
In addition, can bring white elephant to network based on the error control method of retransmission mechanism, use retransmission mechanism can aggravate network congestion in a large number, can make whole network paralysis when serious, also be can not adopted method therefore.
Summary of the invention
In sum, at being technical problem to be solved by this invention, the object of the present invention is to provide a kind of MPEG-4 stream adaptive transmission method based on Internet based on the code stream Adaptive Transmission in the stream media system of MPEG-4.
Technical scheme of the present invention is:
According to the inventive method, its step comprises:
A. set up RTP/UDP/IP (RTP, Real-time Transport Protocol RTP) protocol architecture, the datum plane of this protocol architecture is responsible for transmitting MPEG-4 video flowing and the audio stream that uses the RTP packing; Its control plane then utilizes RTCP (Real-time Control Protocol real time control protocol) feedback network information and RTSP (Real-time Streaming Protocol real-time streaming protocol) transmission control command simultaneously;
B. set up one and can be applicable to the congested control structure that real-time coding also can be used for the store video transmission based on transmitting terminal.
Further, the said congested control structure based on transmitting terminal of setting up, be meant at receiving terminal and set up the FEEDBACK CONTROL protocol module that connects this QOS monitor module after a state according to the packet that arrives provides the QOS monitor module of network congestion state information and, its form with the RTCP report passes to transmitting terminal with the network congestion state information that is obtained; Set up the Rate Control module and a code check Shaping Module of the network information that an acceptance sends here from transmitting terminal, speed and this code check Shaping Module of control that sends by this code check control module controls then at transmitting terminal.
More specifically, in the live broadcast system of real-time coding, what the Rate Control module was directly regulated is encoder, carries out extendible coding.Main coding means is: regulate quantization step and frame-skipping coding.When detecting network congestion, according to seriousness (two threshold value P of packet loss
ThlAnd P
Thh) be divided into two class congested: slight congested and heavy congestion.Work as P
Thl＜P
Loss＜P
ThhThe time be considered as slightly congested; And P
Loss＞P
ThhThe time be considered as heavy congestion.When congested, consider to strengthen quantization step factor u slight, on original basis, increase by 20%, but can not surpass maximum step-length u
MaxWhen heavy congestion, i.e. packet loss P
Loss＞P
Thh, adopt minimizing coding frame per second to alleviate (, considering to abandon the coding of P frame then) till congestion relief such as coding that can not do the B frame earlier.After network is alleviated, carry out improving the output frame rate of encoder exploratoryly, be only the SNR (signal noise ratio) that improves image then, promptly reduce quantization step.Wherein decide the direction of adjustment according to the value of Flag.
And in the store video VOD system, slightly the having any different of self adaptation adjustment strategy of taking and live broadcast system, the Rate Control module controls be the code check Shaping Module of Server (server).This module major function is according to the network information, adjusts the transmission of the encoded video streams with autgmentability, occurs when congested at network, mainly is the network bandwidth that adapts to dynamic change by frame losing optionally.In implementation procedure, still consider seriousness (two threshold value P of packet loss
ThlAnd P
Thh) be divided into two class congested: slight congested and heavy congestion.Work as P
Thl＜P
Loss＜P
ThhThe time be considered as slightly congested; And P
Loss＞P
ThhThe time be considered as heavy congestion.When congested, consider to abandon earlier the B frame slight, the quantity that abandons by few at the most; When heavy congestion, adopt the thought of " property taken advantage of abandon and additivity increase "---at first abandon whole B frames, if just still can not alleviate by far away and near-earth (for the I frame of this subsequence), by few P frame that abandons at the most; Then be to increase frame per second at leisure during congested slowing down, this process is with top to subtract frame opposite, wherein Flag sign indication increase and decrease direction (be Flag=0 subtract frame Flag=1 then add frame).This adaptive frame losing method, the control granularity can be fine to coarse, and therefore is applicable to the compressing video frequency flow of different frame structures.
Aspect ERROR CONTROL, the scheme that the present invention proposes: at first the packet that is later than the 3rd packet (from the RTP sequence number) arrival thereafter or still no show is thought " losing ", this is that broadcast time delay according to maximum defines, and this can be set the requirement of result of broadcast by the user; Concrete controlling schemes is divided into two parts: after decoder detects data-bag lost, then the corresponding therewith view data of front reconstruct is copied into the packet of losing; Coding side can reduce correlation of data, thereby suppress wrong transmission effects in this case like this in the coding or the transmission that reduce B frame even P frame according to the packet loss information of feedback.
At based on the code stream Adaptive Transmission problem in the stream media system of MPEG-4, the present invention has taked the good adaptive coding technology of autgmentability.In live broadcast system, take the adaptive quantization step and the adjustment strategy of coding frame per second to adapt to the Network Transmission condition of variation; And in the program request occasion, streaming media server utilizes the information that RTCP report returns (being the packet loss, propagation delay time etc. of current network) selectively the encoded image frame with autgmentability to be increased and decreased frame per second to come Adaptive Transmission, the advance of this scheme is to consider the congested order of severity clearly and takes corresponding code check to adjust strategy, thereby the control granularity that can realize can be fine to coarse, and therefore is applicable to the compressing video frequency flow of different frame structures.Simultaneously these two kinds of application scenarios all adopt simple and practical error control technique to eliminate the influence of packet loss to reconstructed image in decoding end, thus make the user the time can obtain play quality as well as possible under the network condition that becomes.
Description of drawings
What Fig. 1 described is host-host protocol structural representation of the present invention.
Fig. 2 is the congested control structure schematic diagram that uses among the present invention.
Fig. 3 is the adaptive coding control procedure schematic diagram in the live process.
Fig. 4 is the self adaptation frame per second control flow chart in the on-demand process.
Embodiment
In general video transmission scheme, adopt UDP as transport layer protocol based on Internet.This is to have retransmission mechanism to guarantee transmission reliably because the UDP efficiency of transmission, be the more important thing is Transmission Control Protocol than TCP height, thereby can introduce time delay, and this is totally unfavorable for real-time multimedia transmission.In real-time Streaming Media transmission course, decoder is considered as the multi-medium data bag that arrives after the decoding generated time of regulation invalid.RTP is used for the last a kind of host-host protocol at multimedia data stream of Internet.RTP be defined in one to one or the transmission situation of one-to-many under work, its objective is provides temporal information and realizes that stream is synchronously.RTP relies on RTCP that these services are provided.RTCP Real-time Transport Control Protocol RTCP and RTP provide flow control and congested control service together.During the RTP session, each participant periodically transmits the RTCP bag.Contain the quantity of data packets that has sent, the statistics of losing such as quantity of data packets in the RTCP bag, therefore, server can utilize these information dynamically to change transmission rate, even changes PT Payload Type.RTP and RTCP are used, and they can make the efficiency of transmission optimization with effective feedback and minimum expense, thereby are particularly suitable for transmitting online real time data.
Real-time streaming protocol RTSP has defined how application program transmits multi-medium data by IP network effectively under the situation of one-to-many.RTSP is being positioned on the architecture on RTP and the RTCP, and it uses TCP or RTP to finish transfer of data.HTTP compares with RTSP, and HTTP transmits HTML, and the RTSP transmission is multi-medium data.The HTTP request is sent by client computer, and server responds; When using RTSP, client-server can be sent request, and promptly RTSP can be two-way.RTSP is an application level protocol, the transmission of control real time data.It provides an extensible framework, makes real time data, becomes possibility as controlled, the program request of audio ﹠ video.Data source comprises field data and the data that are stored in the montage.This agreement purpose is to control a plurality of data and sends connection, for selecting sendaisle, as UDP, multicast UDP and TCP, provides approach, and provides method for selecting based on the transmit mechanism on the RTP.
Fig. 1 is that the present invention adopts the RTP/UDP/IP protocol architecture, and datum plane 10 is responsible for transmission actual MPEG-4 video flowing and audio stream with the RTP packing; Control plane 20 utilizes RTSP transmission control command when utilizing RTCP feedback network information of same.
Consider from congested control technology angle, congested control be guarantee multimedia messages normally, the important means of transmission continuously, wherein merged sport technique segments such as autgmentability coding, Adaptive Transmission.Two kinds of thinkings are arranged basically: based on terminal and based on network jamming control method.Because the existing complicated network structure and the unified management organization of neither one, solution that the present invention adopts is based on terminal.The basic ideas of this kind control method are to utilize current network information, adjust the size of transport stream by terminal, make it to adapt with current network condition.This method is divided into again based on transmitting terminal and receiving terminal scheme, the transmitting terminal scheme is meant with streaming media server adjusts code check, the receiving terminal scheme mainly is under hierarchical coding, the multicast transmission environment, client according to the network channel situation adjustment ask to control congested.In order to alleviate the burden of client, the present invention adopts is based on transmitting terminal, the i.e. congested control of streaming media server.
In various control methods, obtaining of current network conditions (mainly being information such as packet loss, time delay, delay variation) is an important techniques link.The discharge model method that the means that realize mainly contain based on the congested control of TCP is a FEEDBACK CONTROL agreement method with utilizing rtcp protocol, considers the host-host protocol structure of reality of the present invention, therefore takes a kind of method in back.
Fig. 2 is congested control structure based on transmitting terminal 100 of the present invention, can be applied to the transmission that real-time coding also can be used for store video.In a kind of application in back, just there is not self-adapting compressing this part function of encoding.At receiving terminal 200 a QoS monitor module 204 is arranged, its major function is according to the state (such as packet loss, time delay) that is delivered to packet, and the network congestion state information is provided.These information are imported a FEEDBACK CONTROL protocol module 205, and the form with the RTCP report passes to transmitting terminal 100 then.In transmitting terminal 100, the network information is sent into Rate Control module 104, control the transmission rate of transmission then by this module 104, as shown in the figure, its control code check adaptation module 107 send compression layer 106 again, and the transmission rate of being regulated is warp let-off code check Shaping Module 105 again, enter RTP layer 103, UDP layer 102 and IP layer 101, and enter the internet until receiving terminal 200.
In the live broadcast system of real-time coding, what Rate Control module 104 was directly regulated is encoder, carries out extendible coding.Main coding means is: regulate quantization step and frame-skipping coding.When detecting network congestion, according to seriousness (two threshold value P of packet loss
ThlAnd P
Thh) be divided into two class congested: slight congested and heavy congestion.Work as P
Thl＜P
Loss＜P
ThhThe time be considered as slightly congested; And P
Losss＞P
ThhThe time be considered as heavy congestion.When congested, consider to strengthen quantization step factor u slight, on original basis, increase by 20%, but can not surpass maximum step-length u
MaxWhen heavy congestion, i.e. packet loss P
Loss＞P
Thh, adopt minimizing coding frame per second to alleviate (, considering to abandon the coding of P frame then) till congestion relief such as coding that can not do the B frame earlier.After network is alleviated, carry out improving the output frame rate of encoder exploratoryly, be only the SNR that improves image then, promptly reduce quantization step.Wherein decide the direction of adjustment according to the value of Flag.According to the flow chart of Fig. 3, adaptive coding control procedure 300 comprises following detailed process in the live process:
S310 initial time: a lower frame per second f0 transmission, moderate quantization step u0.
S320 utilizes RTCP (RTCP Real-time Transport Control Protocol) report to calculate the packet loss P of estimation network
Loss
S330 judges P
Loss＞P
ThlIf, enter S340 then for very then Flag=0 is set; Otherwise enter S390.
S340 judges P
Loss＞P
ThhIf be vacation then enter S360, otherwise S350.
Does S350 continue to judge frame per second=minimum value? it is too poor that this condition establishment then illustrates because of transmission environment, and frame per second has reached minimum value, still has heavy congestion to occur now, therefore, enters S351 and stop transmission course; Otherwise enter frame per second adjustment process S380.
If S360 quantization step u=maximum just changes S350 over to, otherwise enter quantization step u adjustment process S370.
If S370 Flag=0 then quantization step u is increased at the most by few is until maximum; If Flag=1 then lower quantization step-length progressively.Return S320 after this adjustment process is finished, continue to detect network state.
If S380 Flag=0 then reduce the frame per second of encoding at first considers not do part B frame coding, then is the coding of abandoning the B frame fully, and then do not do P frame coding, until abandoning whole P frames by (image sequence that begins with respect to this I frame) far away and near; If Flag=1 then progressively increase the P frame is progressively to increase the B frame then.Return S320 after this adjustment process is finished, continue to detect network state.
S390 is provided with Flag=1; Do you judge frame per second=maximum? condition is that vacation then enters S380, otherwise enters S391, judge quantization step=minimum value? condition is that vacation then enters S370 otherwise changes S320 over to, continues to detect network state.
In the store video VOD system, slightly the having any different of self adaptation adjustment strategy of taking and live broadcast system, referring to Fig. 2,104 controls of Rate Control module be the code check Shaping Module 105 of Server (server).This module major function is according to the network information, adjusts the transmission of the encoded video streams with autgmentability, occurs when congested at network, mainly is the network bandwidth that adapts to dynamic change by frame losing optionally.In implementation procedure, still consider seriousness (two threshold value P of packet loss
ThlAnd P
Thh) be divided into two class congested: slight congested and heavy congestion.Work as P
Thl＜P
Loss＜P
ThhThe time be considered as slightly congested; And P
Loss＞P
ThhThe time be considered as heavy congestion.When congested, consider to abandon earlier the B frame slight, the quantity that abandons by few at the most; When heavy congestion, adopt the thought of " property taken advantage of abandon and additivity increase "---at first abandon whole B frames, if just still can not alleviate by far away and near-earth (for the I frame of this subsequence), by few P frame that abandons at the most; Then be to increase frame per second at leisure during congested slowing down, this process is with top to subtract frame opposite, wherein Flag sign indication increase and decrease direction (be Flag=0 subtract frame Flag=1 then add frame).This adaptive frame losing method, the control granularity can be fine to coarse, and therefore is applicable to the compressing video frequency flow of different frame structures.According to Fig. 4, the self adaptation frame per second control flow 400 in the on-demand process comprises the steps:
S410 initial time: a lower frame per second f0 transmission.
S420 utilizes the RTCP report to calculate the packet loss P of estimation network
Loss
S430 judges P
Loss＞P
ThlIf for very then enter S431, Flag=0 is set, enters S440 then; Otherwise enter S490.
S440 judges P
Loss＞P
ThhIf be vacation then enter S460, otherwise S450.
Does S450 continue to judge P frame counter=0? this condition establishment illustrates then because of transmission environment is too poor and has abandoned whole P frames that still have heavy congestion to occur now, therefore entering S451 stops transmission course; Otherwise enter P framing control process S480.
If S460 B frame counter=0 just changes S450 over to, otherwise enter B framing control process S470.
If S470 Flag=0 then the B frame is carried out by few abandoning at the most (the B frame such as 1/3,2/3 is decided on concrete coding), until abandoning whole B frames; If Flag=1 then progressively increase the B frame.Return S420 after this adjustment process is finished, continue to detect network state.
If S480 Flag=0 then the P frame is carried out by (image sequence that begins with respect to this I frame) far away and near, by few abandoning of (the P frame such as 1/3,2/3 being decided on concrete coding) at the most, until abandoning whole P frames; If Flag=1 then ground near and far progressively increases the P frame.Return S420 after this adjustment process is finished, continue to detect network state.
S490 is provided with Flag=1; Do you judge whether the P frame is sent out entirely (P frame counter=maximum)? is condition that vacation then enters S480, otherwise judges whether the B frame is sent out entirely (B frame counter=maximum)? condition is that vacation then enters S470 otherwise changes S420 over to, continues to detect network state.
Claims (14)
1. the MPEG-4 based on the internet flows adaptive transmission method, and its step comprises:
A. set up the RTP/UDP/IP protocol architecture, the datum plane of this protocol architecture is responsible for transmitting MPEG-4 video flowing and the audio stream that uses the RTP packing; Its control plane then utilizes RTCP feedback network information and RTSP transmission control command simultaneously;
B. set up one and can be applicable to the congested control structure that real-time coding also can be used for the store video transmission based on transmitting terminal.
2. the MPEG-4 stream adaptive transmission method based on the internet according to claim 1, it is characterized in that, the said congested control structure based on transmitting terminal of setting up, be meant at receiving terminal and set up the FEEDBACK CONTROL protocol module that connects this QoS monitor module after a state according to the packet that arrives provides the QoS monitor module of network congestion state information and, its form with the RTCP report passes to transmitting terminal with the network congestion state information that is obtained; Set up Rate Control module and this code check Shaping Module of the network information that a reception sends here from transmitting terminal, the transmission speed that sends by this Rate Control module controls then at transmitting terminal.
3. the MPEG-4 stream adaptive transmission method based on the internet according to claim 1 and 2 is characterized in that, the said Rate Control module that is positioned at transmitting terminal, and it directly regulates encoder in the live broadcast system of real-time coding, carry out extendible coding.
4. the MPEG-4 stream adaptive transmission method based on the internet according to claim 3 is characterized in that, the coding method that the Rate Control module is regulated encoder is to regulate quantization step and frame-skipping coding.
5. the MPEG-4 stream adaptive transmission method based on the internet according to claim 4 is characterized in that said adjusting quantization step and frame-skipping coding are that two packet loss P are set
LossThreshold value P
ThlAnd P
Thh, network congestion is divided into slight congested P
Thl＜P
Loss＜P
ThhWith heavy congestion P
Loss＞P
Thh, when detecting network congestion, if P
Thl＜P
Loss＜P
Thh, then consider to strengthen quantization step factor U, but U should be less than maximum step-length U
MaxIf P
Loss＞P
Thh, then adopt to reduce the coding frame per second till the congestion relief, then, carry out the output frame rate of tentative raising encoder, just reduce quantization step afterwards.
6. the MPEG-4 stream adaptive transmission method based on the internet according to claim 5 is characterized in that, works as P
Thl＜P
Loss＜P
ThhIn time, ties up on the original basis the increasing of quantization step factor U and to increase by 20%, but do not surpass maximum step-length U
MAX
7. the MPEG-4 stream adaptive transmission method based on the internet according to claim 5, said P
Loss＞P
ThhThe time, adopt minimizing coding frame per second to be meant the coding of not doing earlier the B frame, consider to abandon the coding of P frame then.
8. the MPEG-4 stream adaptive transmission method based on the internet according to claim 5 is characterized in that, also sets the adjustment direction of Flag sign expression to the quantization step factor and coding frame per second, and Flag=0 represents to increase quantization step and reduces the coding frame per second; Flag=1 represents the lower quantization step-length and increases the coding frame per second.
9. the MPEG-4 stream adaptive transmission method based on the internet according to claim 1 and 2, it is characterized in that, said Rate Control module, it is control code check Shaping Module in the store video VOD system, so that according to the network information, adjust the transmission of encoded video streams with autgmentability.
10. the MPEG-4 stream adaptive transmission method based on the internet according to claim 9, it is characterized in that, saidly go to control the transmission that the code check Shaping Module is adjusted encoded video streams by the Rate Control module, its method is to occur when congested at network, adapts to the network broadband of dynamic change by selective frame losing.
11. the MPEG-4 stream adaptive transmission method based on the internet according to claim 10 is characterized in that said selective frame losing is the packet loss P that sends here by receiving terminal
LossTwo threshold value P are set
ThlAnd P
Thh, with P
Thl＜P
Loss＜P
ThhThe expression Network Transmission is slightly congested, with P
Loss＞P
ThhP is worked as in the heavy congestion of expression Network Transmission
Thl＜P
Loss＜P
ThhThe time lose the B frame earlier, if still be in congestion state, just by far away and near-earth, by less and the important place abandons the P frame, when congested the slowing down of transmission, just increase frame per second at leisure.
12. the MPEG-4 stream adaptive transmission method based on the internet according to claim 11 is characterized in that, also is provided with to indicate that Flag indicates the increase and decrease direction to frame, Flag=0 subtracts frame, and Flag=1 then adds frame.
13. the MPEG-4 stream adaptive transmission method based on the internet according to claim 1 and 2 is characterized in that, also from the RTP sequence number, the packet that is later than thereafter the 3rd packet arrival or still no show is thought the packet of losing.
14. the MPEG-4 stream adaptive transmission method based on the internet according to claim 13 is characterized in that, after the decoder of receiving terminal detects data-bag lost, just the corresponding therewith pictorial data of front reconstruct is copied into the packet of losing; Transmitting terminal can be in the coding or the transmission that reduce B frame even P frame according to the packet loss information of feedback.
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