CN104349219A - Strict increase loose decrease equal step congestion control algorithm based on mobile communication network - Google Patents
Strict increase loose decrease equal step congestion control algorithm based on mobile communication network Download PDFInfo
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- CN104349219A CN104349219A CN201310321273.1A CN201310321273A CN104349219A CN 104349219 A CN104349219 A CN 104349219A CN 201310321273 A CN201310321273 A CN 201310321273A CN 104349219 A CN104349219 A CN 104349219A
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- packet loss
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/647—Control signaling between network components and server or clients; Network processes for video distribution between server and clients, e.g. controlling the quality of the video stream, by dropping packets, protecting content from unauthorised alteration within the network, monitoring of network load, bridging between two different networks, e.g. between IP and wireless
- H04N21/64784—Data processing by the network
- H04N21/64792—Controlling the complexity of the content stream, e.g. by dropping packets
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/25—Flow control; Congestion control with rate being modified by the source upon detecting a change of network conditions
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
- H04N21/25—Management operations performed by the server for facilitating the content distribution or administrating data related to end-users or client devices, e.g. end-user or client device authentication, learning user preferences for recommending movies
- H04N21/266—Channel or content management, e.g. generation and management of keys and entitlement messages in a conditional access system, merging a VOD unicast channel into a multicast channel
- H04N21/2662—Controlling the complexity of the video stream, e.g. by scaling the resolution or bitrate of the video stream based on the client capabilities
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- Engineering & Computer Science (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Multimedia (AREA)
- Computer Security & Cryptography (AREA)
- Databases & Information Systems (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
- Mobile Radio Communication Systems (AREA)
Abstract
The invention relates to a strict increase loose decrease equal step congestion control algorithm based on a mobile communication network. According to the algorithm, packet loss ratio of the mobile communication network is calculated by adopting a linear prediction algorithm at an application layer. Code rate is regulated by adopting a mode of comparison of the number of times of cumulative abnormal cycles of packet loss ratio and different set values according to the size of packet loss ratio. Sending code rate is up-regulated when the network state is great, and the sending code rate is down-regulated when the network state is poor; and frame quality is influenced to some extent when amplitude of the code rate is down-regulated, while consequences of rising of packet loss ratio and increasing of end-to-end time delay due to network congestion caused by the fact that output code rate surpasses available bandwidth when the code rate is up-regulated are generated so that the condition of code rate up regulation is stricter than that of down regulation (i.e. strict increase loose decrease) Meanwhile, amplitude (step) of code rate regulation each time is fixed, and the code rate is rapidly regulated to a reasonable range so that self-adaptive sending code rate change is realized, video time delay and jittering are reduced and stability of video quality is guaranteed.
Description
Technical field
The present invention relates to a kind of service quality control QoS based on mobile communications network (Quality of Service, QoS) method in mobile communication technology field.Particularly relate to and a kind ofly subtract (Strict Increase Loose Decrease, SILD) unique step congestion avoidance algorithm based on mobile communications network application layer for the Yan Zengsong of real-time video transmission.
Background technology
Along with developing rapidly of mobile communication technology, the QoS in Network Video Transmission has become an important subject.Compared with traditional IP, 3G (3rd-generation, 3G (Third Generation) Moblie), 4G (4rd-generation, forth generation mobile communication) network availability bandwidth is limited, frequency range used is high, by the impact of multipath fading, shadow fading and interference noise, with frequency adjacent frequency serious interference, mobile channel situation affects greatly by surrounding environment change simultaneously, causes error in data and loss to have randomness and paroxysmal feature.Therefore more harsh to the requirement of network service quality, the QoS in research mobile communications network in transmission of video controls to have great importance and application prospect widely.
QoS control method refers to that communication network is quality of service guarantee that user provides when bearer service.The service quality of data network is commonly used the indexs such as transmission delay, delay variation, packet loss and throughput and is weighed.Different business is different for the sensitivity of each index.Video, as a kind of multimedia service with stronger temporal correlation, not only needs to take the larger network bandwidth, and very strict for the requirement of Network Transmission Delays and delay variation index.Video to a small amount of loss of data or mistake insensitive, and can by suitable indemnifying measure hide or recover drop-out.
Current mobile communications network have employed a series of QoS mechanism to guarantee the reasonable distribution of channel resource and effectively to utilize, and usually adopts differentiated service to carry out QoS control at packet-switched domain.The method has good autgmentability, but needs routers to be configured, and part low side devices can not carry any existing QoS agreement, must carry out HardwareUpgring.In order to ensure realtime video transmission, the deployment of mobile communication system QoS is implemented to be a complexity and difficult engineering, must complete evolution by stages.Namely a kind of method solving this problem unifies upgrading deployment to equipment such as the router between Ge great operator, base stations, the method is operated in below application layer, cannot all realize in a short time, therefore this method is only applicable in the laying of network or upgrading stage; Another kind method is on existing network basis, by embedding related software in application layer, provides the QoS control measure such as congestion control to raise network utilization, ensures video transmission quality.Based on above consideration, the present invention proposes a kind of effective congestion avoidance algorithm be applied on network application layer.
In order to solve the congestion control of transmission of video in mobile communications network, in existing achievement in research, mainly use for reference the technology of comparative maturity in traditional IP.IETF (The Internet Engineering Task Force), Internet Engineering Task group) for IP network real time business, propose for (the User Datagram Protocol of UDP in transport layer, User Datagram Protoco (UDP)) the Data Congestion Control Protocol DCCP (Datagram Congestion Control Protocol, DCCP) that transmits.DCCP thinks that conventional Research of Congestion Control Techniques can be divided into two classes substantially: TCP-Like (Transmission Control Protocol, be similar to transmission control protocol) and TFRC (TCP-Friendly Rate Control, the rate control mechanism of TCP close friend).Wherein the principle of TCP-Like is: when TCP transmit leg experiences end-to-end path without its transmission speed of increase just linear time congested, reduce its transmission speed when perceiving path congestion with regard to multiplicative, therefore this rate adaptation mode is referred to as " add increasing take advantage of subtract ".The advantage of TCP-Like is algorithm principle and realizes all very simple, and at Traditional IP (Internet Protocol), the agreement interconnected between network) widely use in network, and be proved to be the Fast Convergent that can realize speed, there is good stability.But the shortcoming of TCP-Like all can turn down transmission rate after being each packet loss, and it is excessive to change step-length, and cause data flow to be shaken large, be not suitable for the business such as video, being only applicable to requirement of real-time is not very high application.TFRC proposes based on the Real-time multimedia of UDP, increases corresponding congestion control mechanism, to apply and other UDP apply network resource sharing liberally, make network have higher bandwidth availability ratio and lower packet loss with TCP in UDP application.There is the slow problem of startup in TFRC class method of rate control.Namely, when just starting to carry out data transmission, still can not judge network condition, so transmitting terminal cannot calculate the transmission rate started in a period of time, for Real-time multimedia, when receiving terminal just starts, having larger time delay and shake.Therefore mainly there is following problem in the congestion control of traditional IP: control in transport layer instead of application layer; Real-time, rapidity and video quality stability requirement cannot be met simultaneously.
In addition some achievements in research it is also proposed the congestion avoidance algorithm being directed to mobile communications network application layer, with delay be index to weigh the situation of network, when network delay is less than lowest latency, increase transmission rate, when network delay is greater than maximum delay, reduce transmission rate.But algorithm does not propose concrete effective execution mode, do not consider the difference of the change of speed on the impact of video quality and increase and minimizing transmission rate yet.Because the big ups and downs of video frequency output code check can cause the parameters such as destination time delay and shake to change thereupon, affect video pictures stability, image quality marked change may be caused simultaneously.Therefore should according to the prediction of network state, rate adjust should be avoided fluctuating widely as far as possible, controls in suitable scope, the stability of guarantee real-time video image quality.
Summary of the invention
In order to solve the congestion control of transmission of video in mobile communications network, rate adjust is controlled in suitable scope, the present invention proposes and a kind ofly subtract (Strict Increase Loose Decrease, SILD) unique step congestion avoidance algorithm based on mobile communications network application layer for the Yan Zengsong of real-time video transmission.This algorithm adopts linear prediction algorithm to calculate the packet loss of mobile communications network in application layer, according to the size of packet loss, the strategy that employing packet loss continuous abnormal moment number statistical value compares with different settings is to regulate code check, when network state is relatively good, raise and send code check, when network state is poor, lowers and send code check; Consider that the amplitude of code check is lowered, only can produce certain influence to image quality, and code check rise likely causes network congestion because bit rate output exceedes available bandwidth, cause the consequences such as packet loss rising and end to end time delay increase, therefore code check raises condition than lowering strict (namely Yan Zengsong subtracts); Simultaneously the amplitude (step-length) of each adjustment code check is fixing, fast by rate adjust to rational scope, achieve adaptive change transmission code check, decrease video delay time and shake, ensure that video quality stability.
The present invention solves the problems of the technologies described above adopted technical scheme: using packet loss as the index weighing network QoS.The strategy that employing packet loss continuous abnormal moment number statistical value compares with different settings is to regulate code check, and setting packet loss lower limit is set to L
min, higher limit is set to L
max, set transmission limit bit rate value and the lower limit of transmitting terminal simultaneously.Receiving terminal adopts the method for linear prediction to dope the packet loss in next moment, and the higher limit of this packet loss and formulation and lower limit are compared, if N continuous (rise preset sends the continuous moment number statistical value that code check needs) individual moment packet loss is less than lower limit, then raises and send code check; If continuous N (downward preset sends the continuous moment number statistical value that code check needs) individual moment packet loss is greater than higher limit, then lowers and send code check.Between the higher limit that transmission code check after adjustment is in the code check of setting and lower limit.Wherein N > M, namely needs to observe that the more moment meets packet loss and is less than lower limit, just can raise transmission code check; And N and M value be greater than 1 integer, doing so avoids once detect that packet loss change just changes the drawback sending code check and cause code check to fluctuate widely, and controls rate adjust in suitable scope.
The invention has the beneficial effects as follows: Yan Zengsong subtracts the situation of change that unique step congestion avoidance algorithm can detect current network state in real time, rate adjust is controlled in suitable scope, fast and effeciently restrain bit rate output, achieve the stability of real-time video image quality, ensure that the fairness of same tcp data stream competition bandwidth, meet TCP friendly feature.
Accompanying drawing explanation
Fig. 1 subtracts unique step congestion avoidance algorithm structured flowchart based on the Yan Zengsong of transmission of video under mobile communications network
In FIG, 1 represents video transmit leg, is responsible for sending video data according to the code check of feedback; 2 represent mobile communications network channel; 3 represent recipient's (mobile terminal); 4 represent mobile terminal screen display; 5 represent packet loss rate measurement module; 6 represent linear prediction module, the packet loss in prediction next moment; 7 represent judging module, according to decision algorithm, make rise, lower or keep the judgement of code check; The entirety of 8 expressions 6 and 7, for Yan Zengsong subtracts unique step congestion avoidance algorithm implementation section.
The flow chart that Fig. 2 is " Yan Zengsong subtracts " algorithm is the detailed description to 7 in Fig. 1.
The wherein packet loss L in the next moment through linear prediction of 9 expression inputs; 10 expressions judge whether L meets L
max> L > L
min(L
minfor packet loss lower limit, L
maxfor packet loss higher limit); 11 for meeting processing module when 10 condition, and namely belowMinCount is (continuously lower than L
minnumber of times measuring period), aboveMaxCount (is greater than L continuously
maxmeasurement moment number of times) empty zero setting; 12 expressions judge whether L meets L≤L
min; 13 represent processing module when meeting 12 condition, and namely belowMinCount adds 1, aboveMaxCount and empties zero setting; Processing module during 14 foot 12 condition with thumb down, namely aboveMaxCount adds 1, belowMinCount and empties zero setting; 15 expressions judge whether belowMinCount meets belowMinCount=N; When 16 expressions meet 15 condition, raise and send code check to R=min{R+R
step, R
max, belowMinCount empties zero setting simultaneously; 17 represent whether aboveMaxCount meets aboveMaxCount=M; 18 represent when meeting 17 condition, lower and send code check to R=max{R-R
step, R
min, aboveMaxCount empties zero setting simultaneously.
Embodiment
1. the packet loss in linear prediction next moment
The present invention uses packet loss as judging the characteristic quantity mark whether mobile communications network is congested.Packet loss lower limit is set to L
min, higher limit is set to L
max.When packet loss is less than L
mintime, think that current network is underload; When packet loss is greater than L
minbe less than L
maxtime, think network full-load run; When packet loss is greater than L
maxtime, then think network over loading.
Consider that in network, packet loss change has short-term correlation, adopt the linear prediction method shown in module 6 in Fig. 1 to calculate packet loss herein.If packet loss numerical value is L in the n-th measuring period
n, then L
ncan be expressed as
L
n=α
1L
n-1+α
2L
n-2+α
3L
n-3
Wherein α
1, α
2, α
3for coefficient correlation, L
n-1, L
n-2, L
n-3be when first first of the pre-test moment, packet loss in front second and front 3rd measuring period respectively, predict subsequent time network packet loss rate L with this
n.
2. quantize to send rate adjust step-length
First will set the maximum bit rate output of transmitting terminal is R
max, minimum bit rate output is R
min, R
maxwith R
minbetween be quantified as some rank, step-length is R
step.The present invention, first for current network, measures the packet loss of network during a series of transmission code check by experiment, and from the transmission code check-packet loss curve drawn, finding out minimum bit rate output is R
min, namely smooth the and network major part of video pictures is in that underload (packet loss is less than L
min) time maximum transmission code check; Same reason finds maximum bit rate output R
max, namely smooth the and network major part of video pictures is in that fully loaded (packet loss is greater than L
minbe less than L
max) time minimum transmission code check.Determine R
minand R
maxafterwards, then experiment measuring goes out a series of step-length R
steppacket loss change during value, selects R
stepfor making video pictures smooth and the maximum step value of packet loss change relatively slow (video pictures is stablized).The R measured under heterogeneous networks status condition
min, R
maxand R
stepgenerally different.
3, implement " Yan Zengsong subtracts " algorithm to control to send code check
What in Fig. 1, module 7 described is " Yan Zengsong subtracts " algorithm, and the detailed process of this algorithm as shown in Figure 2.When transmission of video starts, setting transmitting terminal bit rate output is R=R
min.The packet loss L that algorithm obtains using linear prediction as input, if L
max> L > L
min, keep current code check, the packet loss starting the next moment receives, simultaneously respectively to packet loss continuously lower than L
minmeasurement moment number of times belowMinCount and be greater than L
maxmeasurement moment number of times aboveMaxCount carry out zero setting, as shown in 10,11 branches in Fig. 2; If L≤L
minthen enter 13 of flow chart in Fig. 2,15,16 branches, namely belowMinCount adds 1, aboveMaxCount empties zero setting, and if the value of belowMinCount is greater than N (rise preset sends the continuous moment number statistical value of code check needs), then raises and sends code check to R=min{R+R
step, R
max, restart to add up belowMinCount simultaneously; If L>=L
maxthen enter 14 of flow chart in Fig. 2,17,18 branches, namely aboveMaxCount adds 1, belovMinCount empties zero setting, and if aboveMaxCount is greater than M (downward preset sends the continuous moment number statistical value of code check needs), then lowers and sends code check to R=max{R-R
step, R
min, again add up aboveMaxCount simultaneously.Wherein N > M, namely needs the more continuous moment L≤L to be detected
min, just can raise code check, the rise of code check is stricter than lowering, the implication place that Here it is " Yan Zengsong subtracts ".
Claims (5)
1. the Yan Zengsong based on mobile communications network subtracts unique step congestion avoidance algorithm, it is characterized in that adopting linear prediction algorithm to calculate the packet loss of mobile network in application layer, Yan Zengsong is adopted to subtract unique step congestion avoidance algorithm, the strategy using packet loss continuous abnormal moment number statistical value to compare with different settings carries out Automatic adjusument to video frequency output code check, namely when continuous multiple moment packet loss is less than packet loss lower limit, raise and send code check, when continuous multiple moment packet loss is greater than packet loss higher limit, lower and send code check, but the condition raised than lowering is strict.The step-length of each rise or downward is equal.
2. the Yan Zengsong based on mobile communications network according to claim 1 subtracts unique step congestion avoidance algorithm, is further characterized in that: packet loss lower limit is set to L
min, higher limit is set to L
max.When packet loss is less than L
mintime, think that current network is underload; When packet loss is greater than L
minbe less than L
maxtime, think network full-load run; When packet loss is greater than L
maxtime, then think network over loading, consider that in network, packet loss change has short-term correlation, adopts linear prediction method to calculate packet loss herein.If packet loss numerical value is L in the n-th measurement moment
n, then L
ncan be expressed as
L
n=α
1L
n-1+α
2L
n-2+α
3L
n-3
Wherein α
1, α
2, α
3for coefficient correlation, L
n-1, L
n-2, L
n-3be first first, front second and front 3rd packet loss measured in the moment when the pre-test moment respectively, predict subsequent time network packet loss rate L with this
n.
3. the Yan Zengsong based on mobile communications network according to claim 1,2 subtracts unique step congestion avoidance algorithm, be further characterized in that: according to the size of packet loss, the strategy that employing packet loss continuous abnormal moment number statistical value compares with different settings is to regulate code check, if N continuous (rise preset sends the continuous moment number statistical value that code check needs) individual moment packet loss is less than lower limit, then raises and send code check; If continuous N (downward preset sends the continuous moment number statistical value that code check needs) individual moment packet loss is greater than higher limit, then lowers and send code check.Between the higher limit that transmission code check after adjustment is in the code check of setting and lower limit.Wherein N > M, namely needs to observe that the more moment meets packet loss and is less than lower limit, just can raise transmission code check; And N and M value be greater than 1 integer.
4. the Yan Zengsong based on mobile communications network according to claim 1,2 and 3 subtracts unique step congestion avoidance algorithm, is further characterized in that: set the maximum bit rate output of transmitting terminal as R
max, minimum bit rate output is R
min, R
maxwith R
minbetween be quantified as some rank, step-length is R
step, when transmission of video starts, setting transmitting terminal bit rate output is R=R
min, when each rise or downward send code check, the step-length of adjustment is all R
step, namely adjustment is unique step, R
min, R
maxand R
stepvalue must be measured value, the R measured under heterogeneous networks status condition
min, R
maxand R
stepdifferent.
5. the Yan Zengsong based on mobile communications network according to claim 1,2,3 and 4 subtracts unique step congestion avoidance algorithm, be further characterized in that: the transmission code check self adaptation realizing video under mobile communications network, reduce network congestion, improve the utilance of network.
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Cited By (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN105430532A (en) * | 2015-11-18 | 2016-03-23 | 南京创维信息技术研究院有限公司 | Control method and system for adaptive adjustment of video data transmission |
| CN107295395A (en) * | 2017-06-27 | 2017-10-24 | 武汉斗鱼网络科技有限公司 | Code check adaptive regulation method, device and electronic equipment |
| CN109698928A (en) * | 2018-11-15 | 2019-04-30 | 贵阳朗玛信息技术股份有限公司 | The method and device of video flowing in a kind of adjusting video conferencing system |
| CN109788316A (en) * | 2019-02-27 | 2019-05-21 | 腾讯科技（深圳）有限公司 | Code rate control, video transcoding method and device, computer equipment and storage medium |
| CN111385574A (en) * | 2018-12-28 | 2020-07-07 | 北京字节跳动网络技术有限公司 | Code rate control method and device in video coding, mobile terminal and storage medium |
| CN112203138A (en) * | 2020-10-16 | 2021-01-08 | 深圳乐播科技有限公司 | Projection screen data transmission method, device, equipment and storage medium based on UDP protocol |
| CN115914682A (en) * | 2022-11-24 | 2023-04-04 | 合肥移瑞通信技术有限公司 | Video code rate adjusting method, system, server and storage medium |
| CN117714757A (en) * | 2024-02-04 | 2024-03-15 | 北京搜狐新动力信息技术有限公司 | Code rate adjusting method and device, electronic equipment and storage medium |
Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2005003973A1 (en) * | 2003-06-17 | 2005-01-13 | Qq Technology, Inc. | Digital controlled adaptive equalization designed for high-speed clock and data recovery applications |
| CN101505202A (en) * | 2009-03-16 | 2009-08-12 | 华中科技大学 | Adaptive error correction method for stream media transmission |
| CN101909208A (en) * | 2010-08-05 | 2010-12-08 | 浙江工业大学 | Video wireless transmission control method suitable for CDMA2000 |
| CN202841192U (en) * | 2012-09-01 | 2013-03-27 | 林宏刚 | High-speed reliable one-way transmission system supporting wide area network |
-
2013
- 2013-07-29 CN CN201310321273.1A patent/CN104349219A/en active Pending
Patent Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2005003973A1 (en) * | 2003-06-17 | 2005-01-13 | Qq Technology, Inc. | Digital controlled adaptive equalization designed for high-speed clock and data recovery applications |
| CN101505202A (en) * | 2009-03-16 | 2009-08-12 | 华中科技大学 | Adaptive error correction method for stream media transmission |
| CN101909208A (en) * | 2010-08-05 | 2010-12-08 | 浙江工业大学 | Video wireless transmission control method suitable for CDMA2000 |
| CN202841192U (en) * | 2012-09-01 | 2013-03-27 | 林宏刚 | High-speed reliable one-way transmission system supporting wide area network |
Non-Patent Citations (2)
| Title |
|---|
| 蔡凌等: "《基于丢包率预测的多路径网络负载均衡策略》", 《东北大学学报》 * |
| 黄文等: "《基于丢包率预测的P2P流传输拥塞控制研究》", 《湖南科技学院学报》 * |
Cited By (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN105430532A (en) * | 2015-11-18 | 2016-03-23 | 南京创维信息技术研究院有限公司 | Control method and system for adaptive adjustment of video data transmission |
| CN105430532B (en) * | 2015-11-18 | 2019-04-16 | 南京创维信息技术研究院有限公司 | A kind of control method and its system that video data transmission adaptively adjusts |
| CN107295395A (en) * | 2017-06-27 | 2017-10-24 | 武汉斗鱼网络科技有限公司 | Code check adaptive regulation method, device and electronic equipment |
| CN109698928A (en) * | 2018-11-15 | 2019-04-30 | 贵阳朗玛信息技术股份有限公司 | The method and device of video flowing in a kind of adjusting video conferencing system |
| CN109698928B (en) * | 2018-11-15 | 2021-04-13 | 贵阳朗玛信息技术股份有限公司 | Method and device for adjusting video stream in video conference system |
| CN111385574A (en) * | 2018-12-28 | 2020-07-07 | 北京字节跳动网络技术有限公司 | Code rate control method and device in video coding, mobile terminal and storage medium |
| CN109788316A (en) * | 2019-02-27 | 2019-05-21 | 腾讯科技（深圳）有限公司 | Code rate control, video transcoding method and device, computer equipment and storage medium |
| CN112203138A (en) * | 2020-10-16 | 2021-01-08 | 深圳乐播科技有限公司 | Projection screen data transmission method, device, equipment and storage medium based on UDP protocol |
| CN115914682A (en) * | 2022-11-24 | 2023-04-04 | 合肥移瑞通信技术有限公司 | Video code rate adjusting method, system, server and storage medium |
| CN117714757A (en) * | 2024-02-04 | 2024-03-15 | 北京搜狐新动力信息技术有限公司 | Code rate adjusting method and device, electronic equipment and storage medium |
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