CN109743600B - Wearable field operation and maintenance self-adaptive video streaming transmission rate control method - Google Patents
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Abstract
According to the wearable on-site operation and maintenance self-adaptive video streaming transmission rate control method, the receiving end transmission rate control comprises S1, filtering the arrival time S2, calculating a self-adaptive threshold S3, carrying out overload detection S4, carrying out remote rate control S5 and carrying out REMB processing; the transmission rate control of the sending end specifically comprises the steps of firstly calculating the sending rate according to the packet loss rate of the video stream contained in the RTCP data packet, then further optimizing and adjusting the sending rate, and finally controlling the sending rate of the current data packet by adopting a video coding and decoding constant-speed rate control method according to the calculated sending rate, so that the effective adaptation and matching of the video data stream rate to the network condition are realized. The invention aims to improve the utilization rate of network bandwidth to a greater extent, improve the sending rate of video stream, and improve the fluency and stable transmission rate of video playing by the self-adaptive dynamic adjustment of the video stream transmission rate along with the change of network link quality.
Description
Technical Field
The invention relates to the technical field of video streaming transmission rate control, in particular to a wearable field operation and maintenance self-adaptive video streaming transmission rate control method.
Background
With the development of internet technology, intelligent wearable devices are increasingly applied to field operation and maintenance scenes of power Communication networks, and WebRTC (Web Real-Time Communication) and wearable operation and maintenance technologies are applied, so that a video call based on P2P connection between a wearable terminal and an operation and maintenance platform can be established, remote guidance of field operation and maintenance work is realized, on the other hand, the field operation and maintenance of the power Communication networks lacks an efficient operation and maintenance data Real-Time interaction means, Real-Time operation and maintenance decisions and accurate execution of the operation and maintenance work cannot be realized, the operation and maintenance efficiency and quality are not high, the WebRTC and wearable operation and maintenance technologies are applied, a multi-party video call based on P2P connection between the wearable terminal and the operation and maintenance platform can be established across platforms, active field operation and maintenance of multi-party cooperation and auxiliary decisions is realized, but the field network conditions of the power Communication networks are complex, and the mobile network channel quality is not stable, the video transmission quality fluctuates greatly along with the network quality, the performance of video call by using WebRTC is poor, the network delay and the packet loss rate are high, and the high-efficiency implementation of wearable operation and maintenance is restricted.
Currently, existing solutions fall into two categories: the first solution is to combine the adaptive methods of video coding and video transmission control systems on a heterogeneous network to calculate the coding rate to code the video and transmit the compressed video, and the second solution proposes a novel RTC congestion control algorithm, which can adapt the video stream transmission rate to a certain extent according to the network conditions based on the main idea of using a Kalman filter to estimate the end-to-end one-way delay variation experienced by a data packet from a transmitting end to a receiving end, but the control methods do not consider the transmission rate control of the video stream under the complex and changeable network environment, and do not discuss how to realize the adaptive transmission of the video stream along with the network bandwidth variation by using the video stream code rate adjustment.
In order to solve the development situation of the prior art, the existing papers and patents are searched, compared and analyzed, and the following technical information with high relevance to the invention is screened out:
the technical scheme 1: a patent of "video coding rate control system and method based on coding and decoding end buffer" with patent number CN102761741B, which relates to a video coding and decoding system and method applied to high-performance video transmission, and is mainly completed through two steps: firstly, simulating the buffer state of the decoding end buffer device at each moment: under the condition of CBR, the step of simulating the buffer state of the buffer device at the decoding end at each moment comprises the following steps: estimating the fullness of the buffer device at the decoding end; the step of controlling the coding rate of the video coding comprises the following steps: when the estimated fullness of the decoding end cache device is greater than or equal to a fullness threshold or the encoding end cache delay is greater than or equal to a delay threshold, constraining the target bit number of the video coding according to the fullness of the encoding end cache device and the estimated fullness of the decoding end cache device; secondly, controlling the coding rate of the video coding according to the buffer state of the coding end buffer device and the buffer state of the simulated decoding end buffer device: and under the condition of VBR, constraining the target bit number of video coding according to the fullness of the buffer device at the encoding end and the estimated fullness of the buffer device at the decoding end. The virtual decoding end cache device is used for simulating the state of the cache device of the decoding end cache device at each moment, and the code rate control device controls the encoding code rate of the video encoding device according to the cache state of the encoding end cache device and the cache state of the decoding end cache device simulated by the virtual decoding end cache device.
The technical scheme 2 is as follows: a patent of "a macroblock layer code rate control method based on h.264 video coding standard" with patent number CN103067720B, which relates to a macroblock layer code rate control method based on h.264 video coding standard, and is mainly completed through five steps: firstly, calculating the target bit number of a current frame by using a fluid blockage model and a linear tracking theory; secondly, distributing the bit number of the macro block layer; thirdly, predicting the average absolute error ratio of the brightness of the current macro block image in the current frame; fourthly, calculating the quantization parameter of the current macro block in the current frame; fifthly, the rate distortion optimization of each macro block in the current frame is realized by using the parameter values obtained from the step four. The method can obtain better subjective video quality while controlling the code rate precision, thereby improving the continuity and stability of the video image. The method is simple, is easy to realize by hardware, and has good application prospect.
Technical scheme 3: a code rate control method for stabilizing video quality, which is disclosed in patent No. CN101754003B, and relates to a code rate control method for stabilizing video quality, which mainly comprises the following four steps: firstly, estimating the complexity of a P frame type image or the complexity of an I frame type image; secondly, obtaining the pre-allocation code rate of the k frame image; thirdly, calculating a quantization parameter of the kth frame according to the complexity of the obtained P frame type image and the pre-allocation code rate of the kth frame image or the complexity of the I frame type image and the pre-allocation code rate of the kth frame image; fourthly, the encoder encodes and adjusts the compression rate according to the quantization parameters, and then the video output code rate is controlled.
The invention relates to a macroblock layer code rate control method based on H.264 video coding standard, which comprises the following steps that A, a target bit number of a current frame is calculated by utilizing a fluid blocking model and a linear tracking theory; B. and C, allocating the bit number of the macro block layer, and predicting the average absolute error ratio of the current macro block in the current frame. D. Calculating quantization parameters of corresponding macro blocks; E. and D, realizing the rate distortion optimization of each macro block in the current frame by using the parameter values obtained from the step D. The beneficial effects are as follows: the macro block layer code rate control method can improve the structural similarity of video output sequences while enabling the generated code rate to be similar to the target code rate, thereby obtaining better subjective video quality. The method has the advantages of simplicity, easiness in hardware implementation and the like, and has a good application prospect. But the algorithm is complex, large resource overhead is brought, and the calculation time is long.
Technical scheme 3 relates to an image transmission technology, in particular to a code rate control method for stabilizing video quality. The method is realized by the following technical means, and comprises the following steps of A, estimating the complexity of a P frame type image or the complexity of an I frame type image; B. obtaining the pre-allocation code rate of the kth frame image; C. calculating a quantization parameter of the kth frame according to the complexity of the obtained P frame type image and the pre-allocation code rate of the kth frame image or the complexity of the I frame type image and the pre-allocation code rate of the kth frame image; D. and the encoder encodes and adjusts the compression rate according to the quantization parameters so as to control the video output code rate. The method has the advantages of smoothly regulating and controlling the stable video quality and meeting the video requirements of many application occasions, but the method has the defects of higher implementation difficulty and higher additional cost.
Disclosure of Invention
In view of the above situation, to overcome the defects of the prior art, the present invention aims to improve the utilization rate of network bandwidth, improve the sending rate of video stream, and improve the fluency and stable transmission rate of video playing to a greater extent by adaptively and dynamically adjusting the video stream transmission rate along with the quality change of a network link.
The technical proposal for solving the problem is that the method comprises the transmission rate control of a receiving end and the transmission rate control of a transmitting end, and is characterized in that the transmission rate control of the receiving end comprises the following steps,
s1, filtering the arrival time, calculating the estimation of the one-way time delay gradient by using a Kalman filter, and defining d (t)i) Comprises the following steps: d (t)i)＝(ti-ti-1)-(ts i-ts i-1) Wherein, ti s,ti-1 sRespectively, the start time, t, of the transmission of the i, i-1 th frame of video datai,ti-1Respectively the deadline of all received video data of the ith frame and the ith frame-1;
s2, calculating an adaptive threshold, and dynamically adjusting the initial value gamma (t) of the threshold based on the RRTCC algorithm0) And a threshold value gamma (t)i) To meet the requirement of reasonable delay tolerance;
s3, carrying out overload detection, wherein the overload detection is carried out according to the estimated value d (t) of the one-way time delay gradient every time one video stream is receivedi) And a threshold value gamma (t)i) Relative size, and current state retention time TkeepThe trigger state drives signal S, which has three states: overload (indicating that the current network is congested, which causes a large video stream delay and a high packet loss rate), unCause (indicating that the current network has fewer waiting transmission queues and abundant available bandwidth resources), normal (a state between the two);
overload detection process: when d (t)i)>γ(ti) And T iskeep>TsTriggering an override signal, TsIndicating the lower limit of the current state holding time if Tkeep<TsThen the overload detection signal S is not triggered;
when d (t)i)<0, and Tkeep>TsTriggering an undersuse signal;
when 0 is present<d(ti)<γ(ti) And T iskeep>TsTriggering a normal signal;
in order to meet the requirement of reasonable delay tolerance,
the current network link state should be adapted to,
as follows:
is the RRTCC algorithm trigger signal time threshold, b is
The dynamic adjustment factor is used for adjusting the dynamic adjustment factor,
with d (t)i) Dynamically changing;
s4, remote rate control, calculating the receiving rate R of the receiving end according to the state driving signal Sr(ti) When S is over, the reception rate is lowered to balance the delay, as shown in the following formula; when S is underserve, the receiving rate is increased to improve the bandwidth utilization rate,
wherein
Is the average received rate, λ, within the last 500ms1And λ2A receiving rate reduction factor and an increase factor, respectively;
in addition, according to the bandwidth active detection method, the receiving rate R when the signal S is in the normal stater(ti) Adjusting, detecting corresponding increment according to the rate variation trend, and queuing delay d (t)i) Adjusting the receiving rate R in three casesr(ti) As shown in the following formula,
wherein:
when the value of the queuing delay is lower than the lower limit, the receiving rate is increased, and the increase factor is reduced along with the increase of the queuing delay; when the value of the queuing delay is higher than the upper limit, the receiving rate is reduced, and the reduction factor is increased along with the increase of the queuing delay; when the queuing delay takes the value of the middle area, the receiving rate and ti-1Reception rate of time Rr(ti-1) The consistency is achieved;
s5, REMB processing, receiving rate Rr(ti) The REMB message is sent to a video sending end along with an RTCP packet, under the normal condition, the REMB message is sent once every 1s, once the rate R is reachedr(ti)<0.97Rr(ti-1) I.e. the receiving rate Rr(ti) Attenuation is more than 3%, and REMB can be sent immediately;
the transmission rate control of the sending end is specifically that video is contained according to RTCP data packetsPacket loss rate PLR (t) of streami) Calculating a sending rate Rr(ti) As shown in the following formula:
wherein b isu,blAre each Rr(ti) PLR (t) in staged optimizationi) B is taken as a value thresholdu＝0.1，bl0.02, which can be specifically adjusted according to business requirements, bm＝(bu+bl)/2，ω1,ω2Respectively a reduction factor and an increase factor, omega, of the transmission rate1∈[0.1,1]，ω2∈[0.01,0.1]Dynamically set according to business requirements, R0Detecting the bandwidth for a constant sending rate, and when the packet loss rate is greater than buWhen the packet loss rate is less than b, the transmission rate is increasedlWhen the packet loss rate is b, the transmission rate is decreaseduAnd blIn the meantime, the sending rate is further optimally adjusted, and the formula is as follows:
when b ism<PLR(ti)≤buAccording to the packet loss rate PLR (t)i) Value of (1) in R0And the half of the reference reduction is used for carrying out speed reduction detection, and the reduction is increased along with the increase of the packet loss rate. When b isl≤PLR(ti)≤bmAccording to the packet loss rate PLR (t)i) Value of (1) in R0Half of the reference decrease amount is used as a reference decrease amount for speed-up detection, and the increase amount is reduced along with the increase of the packet loss rate;
r calculated by combining the rate control strategy based on the time delayr(ti) And calculating the data rate of the sending end by using a rate control method based on the packet loss rate
Then, the current transmission is calculatedThe actual video data stream transmission rate R is given by the following formula:
and controlling the sending rate of the current data packet by adopting a rate control method such as video coding and decoding according to the calculated sending rate R, and realizing effective adaptation and matching of the video data stream rate to the network condition.
Due to the adoption of the technical scheme, compared with the prior art, the invention has the following advantages;
1, a new scheme for dynamically adjusting a threshold is provided, so that the threshold is adaptive to the current network environment, the size of the threshold is dynamically adjusted in real time, and the real-time monitoring of the current link load is realized;
r calculated based on rate control strategy of time delayr(ti) And calculating the data rate of the sending end by using a rate control method based on the packet loss rate
Then, the actual video data stream sending rate R of the current sending end is calculated, a basis is provided for the sending rate control of the current data packet, the self-adaptive dynamic adjustment of the video stream transmission rate along with the quality change of a network link is realized, and the smoothness and the stability of video playing are improved;
drawings
FIG. 1 is a graph of the computation of the one-way delay gradient according to the present invention.
FIG. 2 is a diagram of transmission rates in a wired network environment according to the present invention.
FIG. 3 is a diagram of transmission rates in a mobile network environment according to the present invention.
FIG. 4 is a video frame rate diagram under a cable network environment according to the present invention.
FIG. 5 is a video frame rate diagram under a mobile network environment according to the present invention.
Detailed Description
The foregoing and other technical matters, features and effects of the present invention will be apparent from the following detailed description of the embodiments, which is to be read in connection with the accompanying drawings of fig. 1 to 5. The structural contents mentioned in the following embodiments are all referred to the attached drawings of the specification.
The first embodiment of the invention relates to a wearable on-site operation and maintenance self-adaptive video streaming transmission rate control method, which comprises a receiving end transmission rate control and a sending end transmission rate control, and is characterized in that the receiving end transmission rate control is realized by the following steps,
s1, filtering the arrival time, calculating the estimation of the one-way time delay gradient by using a Kalman filter, and defining d (t)i) Comprises the following steps: d (t)i)＝(ti-ti-1)-(ts i-ts i-1) Wherein, ti s,ti-1 sRespectively, the start time, t, of the transmission of the i, i-1 th frame of video datai,ti-1Respectively the deadline of all received video data of the ith frame and the ith frame-1;
s2, calculating an adaptive threshold, and dynamically adjusting the initial value gamma (t) of the threshold based on the RRTCC algorithm0) And a threshold value gamma (t)i) To meet the requirement of reasonable delay tolerance;
s3, carrying out overload detection, wherein the overload detection is carried out according to the estimated value d (t) of the one-way time delay gradient every time one video stream is receivedi) And a threshold value gamma (t)i) Relative size, and current state retention time TkeepThe trigger state drives signal S, which has three states: overuse (indicating that the current network is congested, which causes a large video stream delay and a high packet loss rate), underuse (indicating that the current network has fewer waiting transmission queues and rich available bandwidth resources), normal (a state between the two);
overload detection process: when d (t)i)>γ(ti) And T iskeep>TsTriggering an override signal, TsIndicating the lower limit of the current state holding time if Tkeep<TsThen the overload detection signal S is not triggered;
when d (t)i)<0, and Tkeep>TsTriggering an undersuse signal;
when 0 is present<d(ti)<γ(ti) And T iskeep>TsTriggering a normal signal;
in order to meet the requirement of reasonable delay tolerance,
the current network link state should be adapted to,
as follows:
is the RRTCC algorithm trigger signal time threshold, b is
The dynamic adjustment factor is used for adjusting the dynamic adjustment factor,
with d (t)i) Dynamically changing;
s4, remote rate control, calculating the receiving rate R of the receiving end according to the state driving signal Sr(ti) When S is over, the reception rate is lowered to balance the delay, as shown in the following formula; when S is underserve, the receiving rate is increased to improve the bandwidth utilization rate,
wherein
Is the average received rate, λ, within the last 500ms1And λ2A receiving rate reduction factor and an increase factor, respectively;
in addition, according to the bandwidth active detection method, the receiving rate R when the signal S is in the normal stater(ti) Make an adjustmentAccording to its rate change trend, making correspondent incremental detection, according to queuing time delay d (t)i) Adjusting the receiving rate R in three casesr(ti) As shown in the following formula,
wherein:
when the value of the queuing delay is lower than the lower limit, the receiving rate is increased, and the increase factor is reduced along with the increase of the queuing delay; when the value of the queuing delay is higher than the upper limit, the receiving rate is reduced, and the reduction factor is increased along with the increase of the queuing delay; when the queuing delay takes the value of the middle area, the receiving rate and ti-1Reception rate of time Rr(ti-1) The consistency is achieved;
s5, REMB processing, receiving rate Rr(ti) The REMB message is sent to a video sending end along with an RTCP packet, under the normal condition, the REMB message is sent once every 1s, once the rate R is reachedr(ti)<0.97Rr(ti-1) I.e. the receiving rate Rr(ti) Attenuation is more than 3%, and REMB can be sent immediately;
the transmission rate control of the sending end is specifically that according to the packet loss rate PLR (t) of the RTCP data packet containing video streami) Calculating a sending rate Rr(ti) As shown in the following formula:
wherein b isu,blAre each Rr(ti) PLR (t) in staged optimizationi) B is taken as a value thresholdu＝0.1，bl0.02, which can be specifically adjusted according to business requirements, bm＝(bu+bl)/2，ω1,ω2Respectively a reduction factor and an increase factor, omega, of the transmission rate1∈[0.1,1]，ω2∈[0.01,0.1]Dynamically set according to business requirements, R0Detecting the bandwidth for a constant sending rate, and when the packet loss rate is greater than buWhen the packet loss rate is less than b, the transmission rate is increasedlWhen the packet loss rate is b, the transmission rate is decreaseduAnd blIn the meantime, the sending rate is further optimally adjusted, and the formula is as follows:
when b ism<PLR(ti)≤buAccording to the packet loss rate PLR (t)i) Value of (1) in R0And the half of the reference reduction is used for carrying out speed reduction detection, and the reduction is increased along with the increase of the packet loss rate. When b isl≤PLR(ti)≤bmAccording to the packet loss rate PLR (t)i) Value of (1) in R0Half of the reference decrease amount is used as a reference decrease amount for speed-up detection, and the increase amount is reduced along with the increase of the packet loss rate;
r calculated by combining the rate control strategy based on the time delayr(ti) And calculating the data rate of the sending end by using a rate control method based on the packet loss rate
Then, the current actual video data stream transmission rate R of the transmitting end is calculated, and the formula is as follows:
and controlling the sending rate of the current data packet by adopting a rate control method such as video coding and decoding (the specific control process is the prior art and is not detailed herein) according to the calculated sending rate R, and realizing the effective adaptation and matching of the video data stream rate to the network condition.
Example two in the exampleOn the basis of one, the adaptive threshold is calculated in the step S2, and the patent further dynamically adjusts the initial threshold value γ (t) based on the RRTCC algorithm0) And a threshold value gamma (t)i) To meet the requirement of reasonable delay tolerance, firstly, in the network environment with poor link quality, the threshold initial value is increased, and gamma (t) is defined0) Comprises the following steps:
in the formula, gamma0(t0) For the initial value of the threshold used by the RRTCC algorithm,
the mean value of the one-way delay gradient estimation of the RRTCC algorithm under the network environment with the delay lower than 200ms,
is the one-way time delay gradient estimation value of the former network.
Secondly, in the case of complex and variable network link quality, the threshold γ (t) needs to be dynamically adjustedi) As shown in formula (3).
In the formula, kuAnd ksRepresenting the speed of increase or decrease, k, respectively, of the thresholduSlightly greater than 1, while 0<ks<1。
Adaptive threshold based on an estimate of the one-way delay gradient d (t)i) And the last time threshold value gamma (t)i-1) The new threshold value gamma (t) is calculatedi) When d (t)i)>γ(ti-1) When the network link quality is reduced and the link is overloaded, the method enters an addition fast increase stage, and the threshold value is adjusted to be slightly larger than the estimated value d (t)i) In the state of (1), the algorithm is guided to enter a 'micro-drop' stage, the delay tolerance capacity of the network is increased, the link is prevented from being overloaded all the time, and the threshold value is reducedThe adjustment period of the link quality; when d (t)i)<γ(ti-1) And d (t)i)>When 0, the link quality is stable, and the subtraction micro-drop stage is entered to keep the current network stable state; when d (t)i)<And 0, indicating that the current link quality is better and the bandwidth utilization rate is lower, entering a multiplication fast-decreasing stage, rapidly decreasing the current threshold, guiding the sending end to send more video frame data, and increasing the network bandwidth utilization rate.
When the invention is used, in order to enable technical personnel in the field to better understand the scheme, the WebRTC module in the Chromium browser is modified to realize the data transmission mechanism provided by the patent, and an API provided by the WebRTC official is utilized to build a real-time streaming media server to serve as an experimental platform to carry out simulation experiment. The video session is synchronously opened in the environment of the wired network and the mobile 4G network with the bandwidth of 100M respectively, and the fluctuation curve graph of the transmission rate along with the time is recorded, and the results are shown in fig. 2 and fig. 3. Then, the average value and the standard deviation of the transmission rate after the video call is stabilized are calculated, respectively, and the results are shown in table 1.
Figure 4 shows ATCS algorithm transmission rates in a wired network environment. In the session connection establishment stage, the transmission rate can be rapidly increased. After the transmission rate is stabilized, it can be seen from table 1 that the ATCS algorithm increases the average transmission rate by 10.8% and reduces the standard deviation by 56.1%. Figure 5 shows ATCS algorithm transmission rates in a mobile network environment. The experimental result shows that the ATCS algorithm effectively reduces the transmission rate adjustment period, not only improves the transmission rate of the video stream, but also reduces the transmission rate fluctuation of the video stream, and the effect is obvious particularly under the condition of a mobile network.
The patent uses the video frame rate to compare the video fluency, and uses the algorithm to transmit a standard video sequence with the frame rate of 15 Frames Per Second under the conditions of a wired network and a mobile network, and checks the received video frame rate at a receiving end. The results are shown in FIGS. 4 and 5. Then, the average and standard deviation of the video frame rates were calculated, respectively, and the results are shown in table 2.
The experimental result shows that the ATCS algorithm enables the receiving frame rate of the video stream to be more stable, and meanwhile, the picture quality of the video is improved.
The experimental result shows that the ATCS algorithm has more remarkable effect in a mobile environment, because the RRTCC algorithm has good performance, high transmission rate and generally high receiving frame rate under the wired network condition of high network link quality and low time delay, and therefore, the improvement effect is limited. However, in a mobile network environment, the adaptive threshold policy of the ATCS algorithm effectively reduces the transmission rate adjustment period, greatly improves the network bandwidth utilization rate, and improves the sending rate of the video stream, so the adaptive video stream transmission rate control policy provided by the patent can effectively cope with different network environments, and improves the fluency of video playing and the stable transmission rate.
While the invention has been described in further detail with reference to specific embodiments thereof, it is not intended that the invention be limited to the specific embodiments thereof; for those skilled in the art to which the present invention pertains and related technologies, the extension, operation method and data replacement should fall within the protection scope of the present invention based on the technical solution of the present invention.
Claims (2)
1. A wearable on-site operation and maintenance self-adaptive video streaming transmission rate control method comprises a receiving end transmission rate control and a sending end transmission rate control, and is characterized in that the receiving end transmission rate control is realized by the following steps,
s1, filtering the arrival time, calculating the estimation of the one-way time delay gradient by using a Kalman filter, and defining d (t)i) Comprises the following steps: d (t)i)＝(ti-ti-1)-(ts i-ts i-1) Wherein, ti s,ti-1 sRespectively, the start time, t, of the transmission of the i, i-1 th frame of video datai,ti-1Respectively the deadline of all received video data of the ith frame and the ith frame-1;
s2, calculating an adaptive threshold, and dynamically adjusting the initial value gamma (t) of the threshold based on the RRTCC algorithm0) And a threshold value gamma (t)i) To meet the requirement of reasonable delay tolerance;
s3, carrying out overload detection, wherein the overload detection is carried out according to the estimated value d (t) of the one-way time delay gradient every time one video stream is receivedi) And a threshold value gamma (t)i) Relative size, and current state retention time TkeepThe trigger state drives signal S, which has three states: the method comprises the steps of overuse, undersuse and normal, wherein overuse represents that the current network is congested, so that the video stream delay is large, the packet loss rate is high, undersuse represents that the current network has fewer waiting transmission queues and abundant available bandwidth resources, and normal is a state between overuse and undersuse;
overload detection process: when d (t)i)>γ(ti) And T iskeep>TsTriggering an override signal, TsIndicating the lower limit of the current state holding time if Tkeep<TsThen the overload detection signal S is not triggered;
when d (t)i)<0, and Tkeep>TsTriggering an undersuse signal;
when 0 is present<d(ti)<γ(ti) And T iskeep>TsTriggering a normal signal;
in order to meet the requirement of reasonable delay tolerance,
the current network link state should be adapted to,
as follows:
is the RRTCC algorithm trigger signal time threshold, b is
The dynamic adjustment factor is used for adjusting the dynamic adjustment factor,
with d (t)i) Dynamically changing;
s4, remote rate control, calculating the receiving rate R of the receiving end according to the state driving signal Sr(ti) When S is over, the reception rate is lowered to balance the delay, as shown in the following formula; when S is underserve, the receiving rate is increased to improve the bandwidth utilization rate,
wherein
Is the average received rate, λ, within the last 500ms1And λ2A receiving rate reduction factor and an increase factor, respectively;
in addition, according to the bandwidth active detection method, the receiving rate R when the signal S is in the normal stater(ti) Adjusting, detecting corresponding increment according to the rate variation trend, and queuing delay d (t)i) Adjusting the receiving rate R in three casesr(ti) As shown in the following formula,
wherein:
when the value of the queuing delay is lower than the lower limit, the receiving rate is increased, and the increase factor is reduced along with the increase of the queuing delay; when the value of the queuing delay is higher than the upper limit, the receiving rate is reduced, and the reduction factor is increased along with the increase of the queuing delay; when the queuing delay takes the value of the middle area, the receiving rate and ti-1Reception rate of time Rr(ti-1) The consistency is achieved;
s5, REMB processing, receiving rate Rr(ti) The REMB message is sent to a video sending end along with an RTCP packet, under the normal condition, the REMB message is sent once every 1s, once the rate R is reachedr(ti)<0.97Rr(ti-1) I.e. the receiving rate Rr(ti) Attenuation is more than 3%, and REMB can be sent immediately;
the transmission rate control of the sending end is specifically that according to the packet loss rate PLR (t) of the RTCP data packet containing video streami) Calculating a sending rate
As shown in the following formula:
wherein b isu,blAre respectively as
PLR (t) in staged optimizationi) B is taken as a value thresholdu＝0.1，bl0.02, which can be specifically adjusted according to business requirements, bm＝(bu+bl)/2，ω1,ω2Respectively a reduction factor and an increase factor, omega, of the transmission rate1∈[0.1,1]，ω2∈[0.01,0.1]Dynamically according to business requirementsSet up of R0Detecting the bandwidth for a constant sending rate, and when the packet loss rate is greater than buWhen the packet loss rate is less than b, the transmission rate is increasedlWhen the packet loss rate is b, the transmission rate is decreaseduAnd blIn the meantime, the sending rate is further optimally adjusted, and the formula is as follows:
when b ism＜PLR(ti)≤buAccording to the packet loss rate PLR (t)i) Value of (1) in R0The half of the reference reduction amount is used for carrying out speed reduction detection, and the reduction amount is increased along with the increase of the packet loss rate; when b isl≤PLR(ti)≤bmAccording to the packet loss rate PLR (t)i) Value of (1) in R0Half of the reference decrease amount is used as a reference decrease amount for speed-up detection, and the increase amount is reduced along with the increase of the packet loss rate;
r calculated by combining the rate control strategy based on the time delayr(ti) And calculating the data rate of the sending end by using a rate control method based on the packet loss rate
Then, the current actual video data stream transmission rate R of the transmitting end is calculated, and the formula is as follows:
2. the wearable live operation and maintenance adaptive video streaming rate control method according to claim 1, wherein the adaptive threshold value calculated in step S2 is specifically:
s21, in the network environment with poor link quality, increasing the initial value of the threshold value and defining gamma (t)0) Comprises the following steps:
in the formula, gamma0(t0) For the initial value of the threshold used by the RRTCC algorithm,
the mean value of the one-way delay gradient estimation of the RRTCC algorithm under the network environment with the delay lower than 200ms,
the estimated value of the unidirectional time delay gradient of the former network is obtained;
s22, when the network link quality is complicated and variable, the threshold γ (t) needs to be dynamically adjustedi) As shown in the following formula:
in the formula, kuAnd ksRepresenting the rate of increase or decrease, respectively, of the threshold, while 0<ks<1。
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