CN101562615A - Transmission method for MPEG-4 code based multimedia data stream self-adapting network bandwidth - Google Patents
Transmission method for MPEG-4 code based multimedia data stream self-adapting network bandwidth Download PDFInfo
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Abstract
The invention relates to a transmission method for MPEG-4 code based multimedia data stream self-adapting network bandwidth. The method is characterized by comprising the following steps that: firstly, a transmission end intercepts a receiver report RR provided in an RTCP data package, and works out the current packet loss rate of the network; according to the current packet loss rate of the network, whether the current bandwidth using state of the network is in the congested state or light load state is determined; and the MPEG-4 coded multimedia data stream from the transmission end is encapsulated, and the number of macro blocks in the data packet transferred by the RTP in next transmission is self-adaptively updated according to the current bandwidth using state of the network. Compared with the prior art, the method has the advantages of knowing the current packet loss rate of the network in real time, self-adaptively adjusting the size of the data packet transferred by the RTP according to the change of network conditions (such as bandwidth, congestion degree and the like), further providing better service, controlling the packet loss rate to a medium or low degree, simultaneously improving the utilization rate of bandwidth, and not wasting limited bandwidth resources.
Description
Technical field
The present invention relates to a kind of transmission method of the multimedia data stream self-adapting network bandwidth based on MPEG-4 coding.
Background technology
The RTP RTP is to be used for internet (Internet) to go up host-host protocol at a kind of real-time application of multimedia data stream, and transport services end to end can be provided, and is the reasonable way that solves the real-time video transmission problem.Closely-related with RTP is the RTCP RTCP Real-time Transport Control Protocol, and both are inseparable, are used.The RTCP RTCP Real-time Transport Control Protocol can provide feedback information for the service quality (QoS) of the RTP data that transmit: as packet loss, delay jitter or the like, people can calculate according to the feedback information among the Receiver Report RR that provides in the RTCP packet, thereby judge the behaviour in service of the network bandwidth.The RTCP packet has the bag type of following five kinds of different control informations: Sender Report SR, and Receiver Report RR, Source Description SDES withdraws from indication BYE and application-specific message APP.The RTCP packet uses transmission report SR and reception report RR that service quality (QoS) feedback is provided respectively at transmitting terminal and receiving terminal.Transmitting terminal calculates packet loss, the shake of packet arrival interval and loop time etc. according to Receiver Report RR, determines the situation of offered load, differentiates the network channel situation.
Because the network bandwidth generally is conditional, particularly the bandwidth of radio network information channel is extremely limited, and therefore the signal characteristic that multimedia data stream (as high clear video image etc.) has magnanimity, transmits high-resolution multimedia data stream and need consume massive band width.MPEG-4 is based on the video encoding standard of object, and its bit stream provides the hierarchical description of visual scene.Because the MPEG-4 coding has the interactivity of content, compressibility and general visit efficiently, and become multimedia data stream the most frequently used coded system before carrying out the RTP transmission.MPEG-4 when multimedia data stream is encoded, use be the notion of video object plane VOP, each object (object) that is about on some pictures cuts out, indivedual compressions, same picture is made up of many VOP packets.With transmission resolution is that the video image of the VGA size of 640*480 is an example, if be input as the RGB initial data of per second 30 frames, then the data volume of per second is about 221Mbit before the coding, the compression ratio of supposing common MPEG-4 encoder is 50: 1, and then the per second behind the coding must data quantity transmitted be about 4.4Mbit.This data volume is sizable, can cause very big load to network path (particularly wireless network).Simultaneously, the channel of wireless network have again be subject to disturb, the error rate height is with the characteristic of timely change and asymmetric propagation, therefore, the packet loss of high resolution video image in wireless network is just higher.Generally, packet drop is divided into 3 kinds of statistics ranks in the wireless network: 15%～20% belongs to high packet loss, and 5%～10% belongs to middle packet loss, and 1%～3% belongs to low packet loss ratio.
Studies show that in wireless channel, the RTP length of data package of the utilance of channel width and transmission has direct relation.In certain zone, the utilance of channel increases along with the growth of the RTP length of data package that sends.But the wireless channel finite capacity, during to a certain utilance, can be because the RTP length of data package that sends be excessive, and cause packet loss, channel utilization also decreases.Therefore, the RTP length of data package of transmission is not that the longer the better.
When the multimedia data stream to the MPEG-4 coding carries out the RTP transmission, earlier the MPEG-4 code stream is carried out the RTP packing, when the MPEG-4 code stream was packed, the RTP RTP carried out the soft subpackage as the encapsulation elementary cell with VOP to the data of MPEG-4 code stream earlier; And then the VOP that squeezes into encapsulation elementary cell carried out the hard subpackage of RTP, no matter promptly the content of data, the value that will fix of the machinery MTU MTU of network path (usually with) sends the size of RTP bag as each.Almost each VOP can be above the size of MTU owing to high-resolution MPEG-4 code stream is right, so just need carry out subpackage to each VOP that surpasses the MTU size, being divided into several different RTP wraps and sends, like this, same frame of video may be placed in two different RTP bags, correlation between each packet can be very big, in a single day packet loss takes place in data, just can't take the technology of error concealment to recover at receiving terminal, because the data that receive might be incomplete, do not know which frame has been dropped yet.Therefore, this method might cause the blank screen of receiving terminal picture or discontinuous, has influenced the quality of playing greatly.
Summary of the invention
Technical problem to be solved by this invention is the transmission method that a kind of multimedia data stream self-adapting network bandwidth based on MPEG-4 coding is provided at above-mentioned prior art, this method can be according to the behaviour in service of current network bandwidth, adjust the size that sends the RTP bag adaptively, to be controlled at middle low degree to network packet loss rate, simultaneously can improve the bandwidth utilization rate, not waste limited bandwidth resources.
The present invention solves the problems of the technologies described above the technical scheme that is adopted: be somebody's turn to do the transmission method based on the multimedia data stream self-adapting network bandwidth of MPEG-4 coding, transmitting terminal will adopt RTP RTP and RTCP real time control protocol to give receiving terminal by Network Transmission through the multimedia data stream behind the MPEG-4 coding, it is characterized in that: may further comprise the steps:
Step 1: the size that network path MTU MTU is set;
Step 2: the Receiver Report RR that in transmission end intercepts RTCP packet, provides, and the packet loss of calculating current network;
Step 3: determine that according to the packet loss of current network the bandwidth user mode of current network is in congestion state or is in light condition;
Step 4: the multimedia data stream through behind the MPEG-4 coding of transmitting terminal is carried out the RTP encapsulation, read in an encapsulation unit VOP, if the size of this encapsulation unit VOP is less than or equal to the size of network path MTU MTU, then directly the data encapsulation among this encapsulation unit VOP is become a RTP to send packet and transmit; If the size of this encapsulation unit VOP is greater than the size of network path MTU MTU, then according to the bandwidth user mode of current network, to a current encapsulation unit VOP who squeezes into is that unit carries out subpackage with the macro block, being divided into different RTP transmission packets transmits, and according to the bandwidth user mode of current network, the RTP of real-time update transmission next time sends the number of macroblocks in the packet, if that is: the bandwidth user mode of current network is in congestion state, then send the number of macroblocks of packet, reduce the number of macroblocks in the next RTP transmission packet that transmits according to the RTP of this transmission; If the bandwidth user mode of current network is in light condition, then send the number of macroblocks of packet according to the RTP of this transmission, increase the number of macroblocks in the RTP transmission packet that transmits next time.
As improvement, in the described step 3, judge by the following method current network the bandwidth user mode be in congestion state or be in light condition:
Step (3-1), the threshold value of a network packet loss rate is set, the threshold value scope of this network packet loss rate is [3%, 10%];
Step (3-2) is if the packet loss of current network greater than the threshold value of network packet loss rate, judges that then current network is in congestion state; Otherwise judge that then current network is in light condition.
In the described step 4, be that unit carries out subpackage with the macro block to a current encapsulation unit VOP who squeezes into by following steps:
Step (4-1), a number of macroblocks initial value is set, this number of macroblocks initial value is made as the maximum number of macroblocks that is no more than network path MTU MTU;
Step (4-2), with MB
NewIndividual macro block data is packaged into a RTP transmission packet and transmits MB
NewInitial value be the number of macroblocks initial value;
Step (4-3), judge that whether the current encapsulation unit VOP that reads in remains macro block, as do not have in addition, then current encapsulation unit VOP subpackage processing of reading in finishes, and returns step 4, reads in next encapsulation unit VOP; If remain macro block in addition, then judge the bandwidth user mode of current network;
Step (4-4) is upgraded the number of macroblocks in the RTP transmission packet that transmits next time in the following ways, and is returned step (4-2) if the bandwidth user mode of current network is in congestion state:
MB
new＝MIN{[MB
pre·α]，MB
suiplus}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreFor the RTP of current this transmission sends the number of macroblocks of packet, α is the property the taken advantage of minimizing factor, and the span of α is: 0＜α＜1, MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP;
Step (4-5) is if when current network is in light condition, upgrades RTP in the following ways and sends number of macroblocks in the packet, and return step (4-2):
MB
new＝MIN{[MB
pre+AIR]，MB
suiplus}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreFor the RTP of current this transmission sends the number of macroblocks of packet, AIR is the linear incremental factor, and the span of AIR is: 0＜AIR＜50, MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP.
Like this, when carrying out subpackage for every encapsulation unit VOP, the present invention has adopted the method for similar AIMD, i.e. the property taken advantage of minimizing, and additivity increases, and this is a kind of strategy that rises slowly that falls soon, can load improves to current network rapidly.
In the described step 2, the packet loss of current network calculates by the following method:
Step (2-1), calculate accumulative total poor that the RTP that loses among two Receiver Report RR that receive before and after sending time slots of transmitting terminal sends packet, obtain the RTP that sending time slots of transmitting terminal loses and send packet number L (n), be with equation expression:
L (n)=C (n)-C (n-1), wherein C (n) is the accumulative total that the RTP that loses altogether behind n sending time slots of transmitting terminal sends packet;
Highest serial number is poor among two Receiver Report RR that receive before and after step (2-2), sending time slots of calculating transmitting terminal, obtains the number R (n) that the receivable RTP that arrives of sending time slots of receiving terminal sends packet, with equation expression is:
R (n)=H (n)-H (n-1), wherein, H (n) is a highest serial number among the Receiver Report RR that receives behind n sending time slots of transmitting terminal;
The loss rate F (n) of network when step (2-3), n sending time slots of calculating, the RTP that loses for n sending time slots of transmitting terminal sends the ratio of packet number L (n) and the number R (n) of the receivable RTP transmission packet that arrives of n sending time slots of receiving terminal, with equation expression is:
F(n)＝L(n)/R(n)；
Step (2-4), the loss rate F (n) of network carries out the packet loss P (n) that smoothing processing obtains current network during to n sending time slots,
P (n)=(1-β) * P (n-1)+β * F (n), wherein, β is a smoothing factor, 0.5≤β≤0.8.
Compared with prior art, the invention has the advantages that:
1, the Receiver Report RR by providing in transmission end intercepts RTCP packet has set up the feedback mechanism of a network bandwidth user mode information at transmitting terminal, can understand the packet loss state of current network in real time;
2, the multimedia data stream after the present invention encodes to process MPEG-4 is taked reasonably, dynamic RTP subpackage method, can adjust RTP adaptively according to the variation of network condition (as bandwidth, Congestion Level SPCC etc.) and send the packet size, provide better service quality with this, packet loss is controlled at middle low degree.Simultaneously can improve the bandwidth utilization rate, not waste limited bandwidth resources.
3, adopt transmission method of the present invention, receiving terminal can adopt the error concealment technology when decoding, and the mistake of covering as far as possible and reducing to occur owing to packet loss when making decoding makes the visual effect of image approach original effect.
Description of drawings
Fig. 1 is in the embodiment of the invention, the transmission mechanism schematic diagram between transmitting terminal and the receiving terminal;
Fig. 2 is in the embodiment of the invention, the flow chart that an encapsulation unit VOP is carried out subpackage according to the bandwidth user mode of current network.
Embodiment
Embodiment describes in further detail the utility model below in conjunction with accompanying drawing.
As shown in Figure 1, in Radio Network System, transmitting terminal will adopt RTP RTP and RTCP real time control protocol to give receiving terminal by Network Transmission through the multimedia data stream behind the MPEG-4 coding, and use the RTCP packet to set up feedback, make it dynamic size of distributing the RTP transmission packet of transmission, it may further comprise the steps:
Step 1: the size of network path MTU MTU is set, and preferred value is 512 bytes;
Step 2: the Receiver Report RR that in transmission end intercepts RTCP packet, provides, and the packet loss of calculating current network; Here the packet loss of current network calculates by the following method:
Step (2-1), calculate accumulative total poor that the RTP that loses among two Receiver Report RR that receive before and after sending time slots of transmitting terminal sends packet, obtain the RTP that sending time slots of transmitting terminal loses and send packet number L (n), be with equation expression:
L (n)=C (n)-C (n-1), wherein C (n) is the accumulative total that the RTP that loses altogether behind n sending time slots of transmitting terminal sends packet;
Highest serial number is poor among two Receiver Report RR that receive before and after step (2-2), sending time slots of calculating transmitting terminal, obtains the number R (n) that the receivable RTP that arrives of sending time slots of receiving terminal sends packet, with equation expression is:
R (n)=H (n)-H (n-1), wherein, H (n) is a highest serial number among the Receiver Report RR that receives behind n sending time slots of transmitting terminal;
The loss rate F (n) of network when step (2-3), n sending time slots of calculating, the RTP that loses for n sending time slots of transmitting terminal sends the ratio of packet number L (n) and the number R (n) of the receivable RTP transmission packet that arrives of n sending time slots of receiving terminal, with equation expression is:
F(n)＝L(n)/R(n)；
Step (2-4), the loss rate F (n) of network carries out the packet loss P (n) that smoothing processing obtains current network during to n sending time slots,
P (n)=(1-β) * P (n-1)+β * F (n), wherein, β is a smoothing factor, 0.5≤β≤0.8.
Here, the loss rate F (n) of network is the index that transmitting terminal is estimated the radio network information channel situation during n sending time slots, but the loss rate F (n) of network judges the network channel situation and adjusts the size that transmission RTP sends packet in view of the above in the time of can not directly utilizing n sending time slots, because, this size that can make transmission RTP send packet changes too frequently, also makes the picture quality instability of receiving terminal.When utilizing n sending time slots, before loss rate F (n) the estimation network channel conditions of network, earlier it is made smoothing processing; When the β value increased, the loss rate F (n) of network increased level and smooth P as a result (n) influence during n sending time slots; Get when low when the β value, former level and smooth P as a result (n-1) increases level and smooth P as a result (n) influence of current generation.Experimental statistics is determined smoothing factor β value, and the span of general β is [0.5,0.8]; The optimal selection value that the present invention adopts is 0.6.The judgement that the packet loss of employing after with smoothing processing carries out the network channel situation helps rationally adjusting the size that RTP sends packet;
Step 3: determine that according to the packet loss of current network the bandwidth user mode of current network is in congestion state or is in light condition; Here, the bandwidth user mode of current network is judged by the following method:
Step (3-1), the threshold value of a network packet loss rate is set, the threshold value scope of this network packet loss rate is [3%, 10%], is preferably 5% here;
Step (3-2) is if the packet loss of current network greater than the threshold value of network packet loss rate, judges that then current network is in congestion state; Otherwise judge that then current network is in light condition;
Step 4: the multimedia data stream through behind the MPEG-4 coding of transmitting terminal is carried out the RTP encapsulation: realize by following steps: referring to shown in Figure 2:
Step (4-1), read in an encapsulation unit VOP;
Step (4-2), at first judge whether this encapsulation unit VOP size of data has exceeded the size of the MTU MTU of network path, and the preferred value of MTU is elected 512 bytes as; If the size of this encapsulation unit VOP is less than or equal to the size of network path MTU MTU, then directly the data encapsulation among this encapsulation unit VOP being become a RTP to send packet transmits, current encapsulation unit VOP subpackage processing of reading in finishes, and returns step (4-1);
The size of step (4-3), this encapsulation unit VOP is greater than the size of network path MTU MTU, a number of macroblocks initial value at first is set, this number of macroblocks initial value is made as the maximum number of macroblocks that is no more than network path MTU MTU, is made as 100 macro blocks here;
Step (4-4), with MB
NewIndividual macro block data is packaged into a RTP transmission packet and transmits MB
NewInitial value be the number of macroblocks initial value;
Step (4-5), judge that whether the current encapsulation unit VOP that reads in remains macro block, as do not have in addition, then current encapsulation unit VOP subpackage processing of reading in finishes, and returns step (4-1), reads in next encapsulation unit VOP; If remain macro block in addition, then judge the bandwidth user mode of current network;
Step (4-6) is upgraded the number of macroblocks in the RTP transmission packet that transmits next time in the following ways, and is returned step (4-4) if the bandwidth user mode of current network is in congestion state:
MB
new＝MIN{[MB
pre·α]，MB
suiplus}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreFor the RTP of current this transmission sends the number of macroblocks of packet, α is the property the taken advantage of minimizing factor, and the span of α is: 0＜α＜1, MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP; The use property taken advantage of mode reduces the macro block number in the next RTP transmission packet apace, makes it to adapt to as soon as possible the packet loss situation of network;
Step (4-7) is if when current network is in light condition, upgrades RTP in the following ways and sends number of macroblocks in the packet, and return step (4-4):
MB
new＝MIN{[MB
pre+AIR]，MB
suiplus}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreSend the number of macroblocks of packet for the RTP of current this transmission, AIR is the linear incremental factor, the span of AIR is: 0＜AIR＜50, because network condition is better, by adding a linear incremental factors A IR, use linear mode to increase next RTP gradually and send macro block number in the packet; MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP.
Claims (4)
1, a kind of transmission method of the multimedia data stream self-adapting network bandwidth based on MPEG-4 coding, transmitting terminal will adopt RTP RTP and RTCP real time control protocol to give receiving terminal by Network Transmission through the multimedia data stream behind the MPEG-4 coding, it is characterized in that: may further comprise the steps:
Step 1: the size that network path MTU MTU is set;
Step 2: the Receiver Report RR that in transmission end intercepts RTCP packet, provides, and the packet loss of calculating current network;
Step 3: determine that according to the packet loss of current network the bandwidth user mode of current network is in congestion state or is in light condition;
Step 4: the multimedia data stream through behind the MPEG-4 coding of transmitting terminal is carried out the RTP encapsulation, read in an encapsulation unit VOP, if the size of this encapsulation unit VOP is less than or equal to the size of network path MTU MTU, then directly the data encapsulation among this encapsulation unit VOP is become a RTP to send packet and transmit; If the size of this encapsulation unit VOP is greater than the size of network path MTU MTU, then according to the bandwidth user mode of current network, to a current encapsulation unit VOP who squeezes into is that unit carries out subpackage with the macro block, being divided into different RTP transmission packets transmits, and according to the bandwidth user mode of current network, the RTP of real-time update transmission next time sends the number of macroblocks in the packet, if that is: the bandwidth user mode of current network is in congestion state, then send the number of macroblocks of packet, reduce the number of macroblocks in the next RTP transmission packet that transmits according to the RTP of this transmission; If the bandwidth user mode of current network is in light condition, then send the number of macroblocks of packet according to the RTP of this transmission, increase the number of macroblocks in the RTP transmission packet that transmits next time.
2, the transmission method of the multimedia data stream of network bandwidth adaptive according to claim 1 is characterized in that: in the described step 3, judge by the following method current network the bandwidth user mode be in congestion state or be in light condition:
Step (3-1), the threshold value of a network packet loss rate is set, the threshold value scope of this network packet loss rate is [3%, 10%];
Step (3-2) is if the packet loss of current network greater than the threshold value of network packet loss rate, judges that then current network is in congestion state; Otherwise judge that then current network is in light condition.
3, the transmission method of the multimedia data stream of network bandwidth adaptive according to claim 1 is characterized in that: in the described step 4, be that unit carries out subpackage by following steps with the macro block to a current encapsulation unit VOP who squeezes into:
Step (4-1), a number of macroblocks initial value is set, this number of macroblocks initial value is made as the maximum number of macroblocks that is no more than network path MTU MTU;
Step (4-2), with MB
NewIndividual macro block data is packaged into a RTP transmission packet and transmits MB
NewInitial value be the number of macroblocks initial value;
Step (4-3), judge that whether the current encapsulation unit VOP that reads in remains macro block, as do not have in addition, then current encapsulation unit VOP subpackage processing of reading in finishes, and returns step 4, reads in next encapsulation unit VOP; If remain macro block in addition, then judge the bandwidth user mode of current network;
Step (4-4) is upgraded the number of macroblocks in the RTP transmission packet that transmits next time in the following ways, and is returned step (4-2) if the bandwidth user mode of current network is in congestion state:
MB
new＝MIN{[MB
pre·α]，MB
suiplus?}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreFor the RTP of current this transmission sends the number of macroblocks of packet, α is the property the taken advantage of minimizing factor, and the span of α is: 0＜α＜1, MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP;
Step (4-5) is if when current network is in light condition, upgrades RTP in the following ways and sends number of macroblocks in the packet, and return step (4-2):
MB
new＝MIN{[MB
pre+AIR]，MB
suiplus}
Wherein: MB
NewBe the number of macroblocks in the RTP transmission packet that upgrades back transmission next time according to the bandwidth user mode of current network, MB
PreFor the RTP of current this transmission sends the number of macroblocks of packet, AIR is the linear incremental factor, and the span of AIR is: 0＜AIR＜50, MB
SuiplusIt is macro block number remaining among the encapsulation unit VOP.
4, the transmission method of the multimedia data stream of network bandwidth adaptive according to claim 1 is characterized in that: in the described step 2, the packet loss of current network calculates by the following method:
Step (2-1), calculate accumulative total poor that the RTP that loses among two Receiver Report RR that receive before and after sending time slots of transmitting terminal sends packet, obtain the RTP that sending time slots of transmitting terminal loses and send packet number L (n), be with equation expression:
L (n)=C (n)-C (n-1), wherein C (n) is the accumulative total that the RTP that loses altogether behind n sending time slots of transmitting terminal sends packet;
Highest serial number is poor among two Receiver Report RR that receive before and after step (2-2), sending time slots of calculating transmitting terminal, obtains the number R (n) that the receivable RTP that arrives of sending time slots of receiving terminal sends packet, with equation expression is:
R (n)=H (n)-H (n-1), wherein, H (n) is a highest serial number among the Receiver Report RR that receives behind n sending time slots of transmitting terminal;
The loss rate F (n) of network when step (2-3), n sending time slots of calculating, the RTP that loses for n sending time slots of transmitting terminal sends the ratio of packet number L (n) and the number R (n) of the receivable RTP transmission packet that arrives of n sending time slots of receiving terminal, with equation expression is:
F(n)＝L(n)/R(n)；
Step (2-4), the loss rate F (n) of network carries out the packet loss P (n) that smoothing processing obtains current network during to n sending time slots,
P (n)=(1-β) * P (n-1)+β * F (n), wherein, β is a smoothing factor, 0.5≤β≤0.8.
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