CN1946079A - Selective frame losing method for network bandwidth self adaptive flow medium transmission - Google Patents
Selective frame losing method for network bandwidth self adaptive flow medium transmission Download PDFInfo
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Abstract
This invention relates to a selective frame lost method for transmitting network bandwidth adaptive flow media characterizing in computing a fixed sending time window according to the play time interval of GOP before sending each GOP of a video sequence in a flow media server or the applied layer node of network, every time window of GOP is determined by a start point and an end point, applying TCP transmission protocol to transmit part or total video frame data contained in the GOP selectively in the GOP time window and all RTP packets of the selected video frames are transmitted and all RTP packets of the not-selected video frames are lost.
Description
Technical field
The present invention relates to the internet stream media transmission field, particularly about the Streaming Media transmitting selective frame losing method of the network bandwidth self adaptive at a kind of streaming media server and network intermediate application node layer place.
Background technology
In recent years, along with the fast development of technology such as the Internet and multimedia compression, internet stream media arises at the historic moment.Stream media technology makes media datas such as audio frequency and video the form with a kind of " stream " in real time, continuously and stably to propagate in based on the Internet of IP agreement, be that the recipient almost can just begin to play while receiving data when transmit leg sends data immediately, and needn't wait for that all data start anew to play after all downloading to local storage medium again.
Typical stream media system is by a streaming media server, and an IP internet and a Streaming Media client are formed, and allows other intermediate application node layer of in chain interpolation, for example acting server or P2P (peer-to-peer network) intermediate node etc.As shown in Figure 1, in a typical streaming media server, receive (reading) thread from the video file that Real-Time Video Encoder or precoding are stored, after receiving/read media data and being packaged into the packet of RTP (RTP) form, put into the RTP cyclic buffer; Send thread and from the RTP cyclic buffer, read the RTP bag, and send it to client (or intermediate node) by the Internet.In a typical network intermediate application node layer, send thread and receive the RTP bag and put into the RTP cyclic buffer from upstream flow media server (or other intermediate node); Sending thread reads the RTP bag and sends it to client (or other intermediate node) by the Internet from the RTP cyclic buffer.
The Streaming Media transmission has characteristics such as real-time, continuity and data dependency.Each video requency frame data that real-time requires to transmit in the network must arrive the recipient before the specific broadcast time limit; Continuity requires each video requency frame data to play in a certain order.The video data of compression has two types data dependency: interframe relies on and require recipient's higher frame of video of other priority that it relied on of must decoding before the lower frame of video of priority of decoding; Rely in the frame and decode again after requiring the recipient to be preferably in all to receive all packets of forming a frame, thereby otherwise can cause that decoding error causes phenomenons such as display frame image fragmentation, even may cause the decoder program crashing.
A good stream media transmission system should be able to satisfy the corresponding requirements that above-mentioned these characteristics are brought.Yet present the Internet also mainly is the network of " doing one's best ", and its channel behavior such as bandwidth, time delay and packet loss etc. have certain dynamic and unpredictability, can't provide Streaming Media to transmit needed service quality (QoS) and guarantee.One of them distinct issues is exactly that the both sides up and down bandwidth (code check) that are present in the RTP cyclic buffer place of streaming media server and network intermediate application node layer do not match.Because the isomerism of different sub-network bandwidth and the dynamic fluctuation of the network bandwidth in the Internet, the code check that writes data when the receiving thread at streaming media server or intermediate node place in the RTP cyclic buffer may cause the RTP data-bag lost in this node place randomness when sending thread and transmit the code check of data to downstream node.As shown in Figure 2, when beginning sent No. 25 RTP bag to downstream node, present node was receiving No. 34 RTP bag of sending from upstream node; As shown in Figure 3, when sending No. 25 RTP end-of-packet to downstream node, the RTP bag that receives from upstream node has circulated and has begun to receive the RTP bag No. 44 behind the circle, to follow No. 26 and No. 27 RTP coating lid that does not send as yet after the current RTP bag closely, thereby cause these two RTP bags can't correctly be sent to downstream node.How to select this moment the next grouping that needs to send that two kinds of schemes are arranged: the one, directly increase progressively the pointer of circular buffering formation and send No. 42 RTP grouping that receives recently; The 2nd, search for backward and find the current smallest sequence number that is not capped as yet (No. 28) RTP grouping to send.Yet no matter adopt which kind of scheme, the packet sequence number that all can cause downstream node to receive is no longer continuous, has lost 26-41 number grouping in scheme one, has lost in scheme two No. 26 and No. 27 grouping, the RTP data-bag lost of randomness promptly occurs.
The RTP data-bag lost of above-mentioned randomness has serious consequences for the client decoder.For example losing of some RTP bag data not only can make and phenomenons such as mosaic, image fragmentation occur in the decoded picture of present frame in I frame or the P frame, and can cause that analogue all appears in the decoding of follow-up all P frames and B frame among the current GOP (frame group), even may cause the collapse of decoder process.In order to address this problem, need to introduce suitable adaptability transmission controlling mechanism.
The common adaptability transmission controlling mechanism at the bandwidth mismatch problem mainly contains radio hookup, transcoding and scalable coding etc.Yet radio hookup is only applicable to server and can't be used for intermediate node, and can't realize the switching of arbitrary bit rate; The computation complexity of real-time transcoding is higher, and relates to the bag of group again of RTP bag and unpack etc., has increased the complexity of application program management greatly; Scalable coding need adopt new coding techniques and standard, can't be applicable to existing a large amount of non-scalable coding standard that exists.
Summary of the invention
At the problems referred to above, the Streaming Media transmitting selective frame losing method that the purpose of this invention is to provide a kind of network bandwidth self adaptive, use the inventive method, real-time is good and simple, even the network bandwidth do not match and the situation of dynamic change in time under, the result of broadcast of client also can obtain coherent, smooth and complete clear picture.
For achieving the above object, the present invention takes following technical scheme: a kind of Streaming Media transmitting selective frame losing method of network bandwidth self adaptive, it is characterized in that: in streaming media server or network intermediate application node layer, send before each GOP of a video sequence, reproduction time according to GOP is spaced apart fixing transmitting time window of its calculating, and each GOP time window is determined by a starting point and an end point; Adopt TCP host-host protocol part or all of video requency frame data of being comprised of this GOP of reliable transmission only selectively in a GOP time window, wherein all RTP bags of selecteed frame of video all are transmitted, and non-selected frame of video RTP bag all is dropped.
Before being ready for sending first RTP bag of current GOP, at first predict and tentatively determine the selective frame losing strategy of current GOP according to the transmission situation of a last GOP, if the current time does not surpass the starting point of current GOP time window, the arbitrary frame of video that does not abandon among the current GOP is set then; If the current time has surpassed the starting point of current GOP time window, and just having sent the frame of video that finishes is an I frame, all frame of video that abandon among the current GOP then are set, if the current time has surpassed the starting point of current GOP time window and just sent the frame of video that finishes is not the I frame, all B frames that abandon among the current GOP are set then.
After all RTP bags of each selected frame of video are finished in using the reliable current GOP of transmission of Transmission Control Protocol, whether the selective frame losing strategy of current GOP is adjusted according to the current time decision, if the current time does not surpass the end point of current GOP time window, then do not adjust the frame losing strategy and continue to send next frame of video by original frame losing strategy; If the current time has surpassed the end point of current GOP time window, then be provided with and abandon follow-up all P frames and B frame among the current GOP, directly change transmission over to video requency frame data bag among the next GOP.
The present invention is owing to take above technical scheme, therefore it is compared with other solution unmatched Streaming Media transmission of network bandwidth adaptive control mechanisms, have the following advantages: network bandwidth self adaptive selective frame losing method complexity 1, proposed by the invention is low, real-time is good, can realize the switching of arbitrary bit rate, and do not need to change original buffer data structure in streaming media server and the network intermediate application node layer, simple, be convenient to promote the use of.2, the present invention is owing to carry out selective frame losing according to the priority of each frame of video in constraint of the reproduction time of GOP and the GOP, therefore can guarantee all RTP bags that each frame of video comprises all safety send to receiving terminal or all be dropped, and received each frame of video of client is all complete and can be correctly decoded, and does not need to depend on any selected frame that abandons.3, the present invention can guarantee under the unmatched situation of bandwidth, though the viewed play frame rate of client may descend to some extent, but still can be continuously, continue to play glibly, and each frame picture is all still intact.The inventive method is applicable to video encoding standards such as MPEG series and AVS, can be widely used in the transfer of data at various internet stream media servers or network intermediate application node layer place.
Description of drawings
Fig. 1 is typical stream media system structural representation
Fig. 2 is that streaming media server or network intermediate application node layer send and receive RTP bag schematic diagram
Fig. 3 is streaming media server or network intermediate application node layer RTP bag random loss schematic diagram
Fig. 4 is the flow chart of network bandwidth self adaptive selective frame losing method provided by the invention
Embodiment
Below in conjunction with drawings and Examples, the present invention is described in detail.
The inventive method comprises following content: when the upstream and downstream network bandwidth (code check) when not matching, the transmission thread can abandon the lower frame of video of part priority selectively and not send, thereby guarantees to utilize limited bandwidth resources that the frame of video that other has higher priority is sent to downstream node smoothly in official hour.The selected data that abandon should not constitute any influence to sending being correctly decoded of data, thereby make client also can be continuously under the situation that only receives partial data, swimmingly with complete, picture is play clearly, though may descend to some extent in part-time section frame per second.
The present invention is applicable to the video encoding standard with GOP (frame group) structured coding, for example MPEG1, MPEG2, MPEG4, H.264 with AVS etc.In the video encoding standard based on gop structure, each GOP must begin with an I frame, follows a series of P frames and B frame thereafter closely.The I frame is intraframe coding, does not need during decoding with reference to other any frame; The P frame is a forward predictive coded, needs during decoding with reference to being positioned at its I frame or P frame before; The B frame is a bi-directional predictive coding, needs during decoding with reference to the I frame or the P frame that are positioned at its both sides, front and back.For DISPLAY ORDER is " IBBPBBP ... " the GOP of form, its transmission and decoding order is " IPBBPBB ... "When decision abandoned certain frame, all other frames that directly or indirectly depend on this frame also all must be dropped among this GOP, otherwise can cause decoder error or failure.According to above-mentioned frame losing principle and interframe dependence, each B frame can abandon at any time; When abandoning a P frame, be positioned at this P frame all other P frames and B frame afterwards by decoding order among the current GOP and all should abandon; When abandoning an I frame, other frame of all among the current GOP also all should abandon.
In a stream media transmission system, the transmission code rate of media data in network should be as far as possible to consume the speed of these data consistent with player.Because each frame of video all must be put a regular time and finish broadcast after client terminal start-up is play in stream media system, so the time interval of a GOP of player consumption (broadcast) is confirmable.The present invention selects to allow network finish transmission to all video requency frame datas among this GOP at interval at a GOP reproduction time for this reason.If, then should select to abandon partial frame, and after the prescribed time-limit arrives, skip current GOP and directly enter transmission next GOP because bandwidth constraints can't be finished the transmission of all these frames at the appointed time.
Can safe and reliable in order arrival receiving terminal for the RTP bag that guarantees all selected transmissions, the present invention selects to adopt Transmission Control Protocol (transmission control protocol) to send these groupings.
The present invention's carry is mainly predicted frame losing strategy among the current GOP according to the transmission situation of a last GOP based on the network self-adapting selective frame losing method of TCP host-host protocol and GOP time window constraint, and according to real network bandwidth throughput situation the frame losing strategy of current GOP is suitably adjusted after each frame in sending current GOP again.As shown in Figure 3, the concrete implementation step of this selective frame losing method is as follows:
S1. by out of band signal instruction or check that the modes such as timestamp of adjacent two I frames obtain the GOP reproduction time spacing value of current video code stream.
S2. with the current time datum mark, according to the starting point and the end point of the transmitting time window of each GOP of GOP reproduction time spacing value calculated for subsequent.
S3. before first RTP of the I frame that is ready for sending a GOP grouping, judge that the current time is whether above the time window starting point of this GOP:
If S3.1. do not surpass, then adopt Transmission Control Protocol to send all frames among this GOP successively, comprise each RTP bag of I frame, P frame and B frame;
S3.2. otherwise further judge the last whether I frame of the frame that finishes that just sent:
S3.2.1. if then setting abandons all frames among this GOP, comprises I frame, P frame and B frame;
S3.2.2. otherwise be provided with and abandon all B frames among this GOP, adopt Transmission Control Protocol only to send I frame and follow-up all P frames among this GOP successively.
S4. in sending current GOP after all RTP end-of-packet of each frame of video, judge that the current time is whether above the time window end point of this GOP:
If S4.1. do not surpass, continue then set by step that determined frame losing strategy adopts Transmission Control Protocol to send the frame of video that the next one should send successively among the S3;
S4.2. otherwise abandon follow-up all P frames and B frame among this GOP, directly enter transmission to next GOP.
Claims (3)
1, a kind of Streaming Media transmitting selective frame losing method of network bandwidth self adaptive, it is characterized in that: in streaming media server or network intermediate application node layer, send before each GOP of a video sequence, reproduction time according to GOP is spaced apart fixing transmitting time window of its calculating, and each GOP time window is determined by a starting point and an end point; Adopt TCP host-host protocol part or all of video requency frame data of being comprised of this GOP of reliable transmission only selectively in a GOP time window, wherein all RTP bags of selecteed frame of video all are transmitted, and non-selected frame of video RTP bag all is dropped.
2, the Streaming Media transmitting selective frame losing method of network bandwidth self adaptive as claimed in claim 1, it is characterized in that: before being ready for sending first RTP bag of current GOP, at first predict and tentatively determine the selective frame losing strategy of current GOP according to the transmission situation of a last GOP, if the current time does not surpass the starting point of current GOP time window, the arbitrary frame of video that does not abandon among the current GOP is set then; If the current time has surpassed the starting point of current GOP time window, and just having sent the frame of video that finishes is an I frame, all frame of video that abandon among the current GOP then are set, if the current time has surpassed the starting point of current GOP time window and just sent the frame of video that finishes is not the I frame, all B frames that abandon among the current GOP are set then.
3, press the Streaming Media transmitting selective frame losing method of claim 1 or 2 described network bandwidth self adaptives, it is characterized in that: after all RTP bags of each selected frame of video are finished in using the reliable current GOP of transmission of Transmission Control Protocol, whether the selective frame losing strategy of current GOP is adjusted according to the current time decision, if the current time does not surpass the end point of current GOP time window, then do not adjust the frame losing strategy and continue to send next frame of video by original frame losing strategy; If the current time has surpassed the end point of current GOP time window, then be provided with and abandon follow-up all P frames and B frame among the current GOP, directly change transmission over to video requency frame data bag among the next GOP.
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