CN107026856A - The optimization method and optimization system of a kind of network plug-flow quality - Google Patents

Abstract

The present application provides a method and an optimization system for optimizing the quality of network streaming, wherein the method includes: collecting characteristic information of streaming according to a preset cycle; according to the current average sending time of each frame of TCP and the current buffer data throughput, Adjust the code rate of the next cycle, and feed the adjusted code rate back to the encoder as the target code rate; adjust the frame rate of the next cycle according to the current buffer data throughput, and adjust the frame rate according to the adjusted frame rate to determine the time interval between two frames; based on the determined time interval, the audio and video data entering the encoder is filtered to generate audio and video data that conforms to the adjusted frame rate; according to the adjusted bit rate and the adjusted Encoding and transmission of audio and video data at a certain frame rate. The method and system for optimizing the quality of network streaming provided by this application can smoothly adjust the code rate and frame rate of streaming.

Description

An optimization method and optimization system for network streaming quality

technical field

The present application relates to the technical field of audio and video processing, in particular to a method and system for optimizing the quality of network streaming.

Background technique

In recent years, mobile video live broadcasting has become more and more popular, and has been widely used in entertainment, education, game e-sports, sports and other fields. Various live broadcasting products have fierce competition, and live broadcasting products and services that can provide excellent user experience will eventually become popular in this field. Stand out from the competition.

The factors that affect the viewing experience of live broadcasts are generally divided into two dimensions, one is the clarity of the picture (that is, the picture quality), and the other is the fluency. Image quality is positively correlated with bit rate and resolution, and fluency is positively correlated with frame rate and data bandwidth. When the bandwidth of the network is constant, the fluency has a negative correlation with the code rate, that is, when the code rate exceeds the actual network data throughput rate, the fluency will decrease. Ideally, when both the anchor and the audience have unlimited bandwidth, you only need to choose the best picture quality that a live broadcast device can withstand to push the stream, and you can get a high-definition, smooth and high-quality viewing experience. However, considering that the network environment of the live broadcaster (that is, the host) is very different, packet loss, jitter, and delay often occur on the link from the live broadcast end to the streaming server, which leads to poor streaming of live streaming and a decline in viewing experience . In this case, it is necessary to balance the two dimensions of image quality and fluency to maintain a good viewing experience. A large number of user feedback shows that using lower image quality and high fluency is not the same as using high image quality and frequent freezes. Compared with live broadcast, the former can bring better user viewing experience. At this time, a technical solution that can dynamically adjust the streaming quality according to the network quality of the live broadcast terminal is particularly necessary.

At present, the general protocol standard for live streaming is RTMP (Real Time Messaging Protocol, Real-time Message Transmission Protocol), which is based on TCP for transmission. It is a connection-oriented and reliable transport layer protocol, and internally implements congestion control strategies. In a network environment with a high packet loss rate, TCP's automatic retransmission strategy will occupy a large part of the bandwidth, resulting in a decrease in network data throughput per unit time and causing freezes at the playback end. There are currently three main solutions on the market to deal with weak network streaming:

The anchor chooses the clarity of the push stream according to the feedback from the audience, that is, the push stream end does not automatically judge the network processing;

Judge the network quality at the beginning of the live broadcast. If poor network conditions are detected, a lower streaming configuration will be used to start the live broadcast, and this configuration will always be used throughout the live broadcast process;

Set the streaming quality configuration to preset two levels, high and low, and switch back and forth between these two levels according to the network conditions during the live broadcast.

The above solutions can alleviate to a certain extent the problem of unsmooth streaming at the live broadcast end caused by network fluctuations or weak networks, but they also have a series of defects. The specific analysis is as follows:

Relying on the anchor to choose the clarity of the stream based on audience feedback is too dependent on the experience of the anchor, and the feedback from the audience cannot accurately determine the root cause of the live broadcast stuttering. Poor network at the anchor end or problems with the audience's own network may cause live broadcast Negative feedback from experience;

Factors of network fluctuations are not taken into account. It is possible that the network is fluctuating at the beginning of the live broadcast, resulting in low-quality streaming, and then the network becomes stable during the live broadcast, but the live broadcast can only always use low-quality images, which cannot achieve the best viewing experience. optimization. In addition, high-quality streaming may be used at the beginning, and network fluctuations during the live broadcast will also cause frequent freezes;

Using high and low image quality configurations to switch back and forth according to network conditions will bring two problems. The first problem is that the playback end can clearly perceive the jitter of the image quality, and the second is that when frequently switching from low image quality to high image quality, it may increase the network burden instantaneously, causing the transmitted bit rate to exceed the actual bandwidth, causing freezes, and in short, it cannot be done. to smooth switching.

It should be noted that the above introduction to the technical background is only for the convenience of a clear and complete description of the technical solution of the present application, and for the convenience of understanding by those skilled in the art. It cannot be considered that the above technical solutions are known to those skilled in the art just because these solutions are described in the background technology section of this application.

Contents of the invention

The purpose of the embodiments of the present application is to provide a method and system for optimizing the quality of network streaming, which can smoothly adjust the code rate and frame rate of streaming.

In order to achieve the above purpose, the present application provides a method for optimizing the quality of network push stream on the one hand, the method includes: collecting feature information of push stream according to a preset period, and the feature information includes average code rate, average frame rate, buffer Area data throughput, TCP average sending time of each frame; according to the current average TCP sending time of each frame and the current buffer data throughput, the code rate of the next cycle is adjusted, and the adjusted code rate is used as the target code rate feedback to the encoder; adjust the frame rate of the next cycle according to the current buffer data throughput, and determine the time interval between two frames according to the adjusted frame rate; The audio and video data is filtered to generate audio and video data conforming to the adjusted frame rate; the audio and video data is encoded and transmitted according to the adjusted bit rate and the adjusted frame rate.

Further, according to the current average TCP transmission duration of each frame and the current buffer data throughput, adjusting the code rate of the next cycle specifically includes: when the current average TCP transmission duration of each frame is greater than the first preset risk value, Decrease the current code rate according to the preset code rate step size, and use the reduced code rate as the code rate for the next cycle; when the average transmission duration of each current TCP frame is less than or equal to the first preset risk value, Judging whether the current average TCP transmission duration of each frame is greater than a second preset risk value; if the current average TCP transmission duration of each frame is greater than the second preset risk value and the data occupancy rate of the current buffer is greater than a preset threshold, Decrease the current code rate according to the preset code rate step, and use the reduced code rate as the code rate for the next cycle.

Further, the method further includes: judging whether the code rate adjustment performed last time is to decrease the code rate, and if so, increasing the preset code rate step size to a preset value, and decreasing the current code rate according to the increased preset value. code rate.

Further, according to the current average sending time of each frame of TCP and the current buffer data throughput, adjusting the code rate of the next cycle specifically includes: judging the relationship between the amount of output data and the amount of ingested data in the current buffer, when When the amount of output data is greater than or equal to the amount of ingested data, add one to the count value of the counter; when the amount of output data is less than the amount of ingested data, decrease the count value of the counter by one; code rate for one cycle.

Further, according to the size of the count value in the current counter, adjusting the code rate of the next cycle specifically includes: when the count value in the counter is smaller than the first determination threshold, decreasing the current code rate according to the preset code rate step size, and Use the reduced code rate as the code rate of the next cycle, and simultaneously reset the count value of the counter to zero; when the count value in the counter is greater than the second judgment threshold, increase the current code rate according to the preset code rate step rate, and use the increased code rate as the code rate for the next period, and reset the count value of the counter to zero.

Further, the method further includes: judging whether the code rate adjustment performed last time is to decrease the code rate, and if so, increasing the preset code rate step size to a preset value, and decreasing the current code rate according to the increased preset value. Code rate: determine whether the code rate adjustment performed last time is to increase the code rate, if so, increase the preset code rate step size to a preset value, and increase the current code rate according to the increased preset value.

Further, the push stream is generated in the following manner: encode the pre-collected audio and video data through an encoder, and transmit the encoded audio and video data through the real-time message transmission protocol RTMP after being packaged, so as to generate the Push flow.

In order to achieve the above purpose, another embodiment of the present application also provides a system for optimizing the quality of network streaming, the system includes: a streaming data collection control module, used to collect characteristic information of streaming according to a preset period, and the Feature information includes average code rate, average frame rate, buffer data throughput, and average TCP sending time per frame; the adaptive code rate control module is used to control the The code rate of the next cycle is adjusted, and the adjusted code rate is fed back to the encoder as the target code rate; the dynamic frame rate control module is used to adjust the frame rate of the next cycle according to the current buffer data throughput , and determine the time interval between two frames according to the adjusted frame rate; filter the audio and video data entering the encoder based on the determined time interval to generate audio and video data that conforms to the adjusted frame rate; transmission control The module is used for encoding and transmitting audio and video data according to the adjusted code rate and the adjusted frame rate.

Further, the adaptive code rate control module specifically includes: a first judging unit, configured to decrease the current code rate according to a preset code rate step when the current average TCP transmission duration of each frame is greater than a first preset risk value. rate, and use the reduced code rate as the code rate of the next cycle; the second judging unit is used to judge the Whether the current average sending time of TCP per frame is greater than the second preset risk value; the third judging unit is used to if the current average sending time of TCP per frame is longer than the second preset risk value and the data occupancy rate of the current buffer is greater than the preset risk value When the threshold is set, the current code rate is decreased according to the preset code rate step size, and the reduced code rate is used as the code rate for the next cycle.

Further, the adaptive code rate control module specifically includes: a throughput judging unit for judging the relationship between the amount of output data and the amount of data ingested in the current buffer, when the amount of output data is greater than or equal to the amount of data ingested When the count value of the counter is increased by one; when the amount of output data is less than the amount of data swallowed, the count value of the counter is decreased by one; the code rate adjustment unit is used to adjust the value of the next cycle according to the size of the count value in the current counter code rate.

The embodiment of the present invention utilizes the interaction between the code rate and the frame rate adjustment to effectively control the amount of audio and video data sent in the case of network fluctuations on the one hand, relieve network congestion, and at the same time bring a smooth viewing experience to the playback terminal; on the other hand When the network quality is restored, it can ensure that the bit rate and frame rate of audio and video are pushed with the optimal quality configuration set, and it can switch between different bit rate and frame rate configurations in a smoother way, without It brings an abrupt sense of image quality fluctuations to the viewing end.

With reference to the following description and accompanying drawings, specific embodiments of the present application are disclosed in detail, indicating the manner in which the principles of the application may be employed. It should be understood that the embodiments of the present application are not limited thereby in scope. Embodiments of the present application encompass many changes, modifications and equivalents within the spirit and scope of the appended claims.

Features described and/or illustrated with respect to one embodiment can be used in the same or similar manner in one or more other embodiments, in combination with, or instead of features in other embodiments .

It should be emphasized that the term "comprising/comprising" when used herein refers to the presence of a feature, integer, step or component, but does not exclude the presence or addition of one or more other features, integers, steps or components.

Description of drawings

The included drawings are used to provide further understanding of the embodiments of the present application, which constitute a part of the specification, are used to illustrate the embodiments of the present application, and explain the principles of the present application together with the text description. Apparently, the drawings in the following description are only some embodiments of the present application, and those skilled in the art can obtain other drawings according to these drawings without any creative effort. In the attached picture:

FIG. 1 is a flow chart of a method for optimizing the quality of network streaming provided by an embodiment of the present application;

FIG. 2 is a schematic diagram of network push quality optimization in an embodiment of the present application;

Fig. 3 is the flowchart of code rate adjustment in the embodiment of the present application;

FIG. 4 is a functional block diagram of a system for optimizing the quality of network streaming provided by an embodiment of the present application.

detailed description

In order to enable those skilled in the art to better understand the technical solutions in the present application, the technical solutions in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the described The implementations are only some of the implementations of the present application, not all of them. Based on the implementation manners in this application, all other implementation manners obtained by persons of ordinary skill in the art without creative efforts shall fall within the scope of protection of this application.

FIG. 1 is a flow chart of a method for optimizing the quality of network streaming provided by an embodiment of the present application. Although the processes described below include multiple operations occurring in a particular order, it should be clearly understood that these processes may include more or fewer operations, which may be performed sequentially or in parallel (e.g., using parallel processors or a multi-threaded environment). . As shown in Figure 1, the method may include:

Step S1: Collect feature information of streaming according to a preset period, and the feature information includes average code rate, average frame rate, buffer data throughput, and average TCP sending time of each frame.

In this embodiment, data collection may be performed on the push stream of the current system for a period of time to obtain historical data of the push stream. Specifically, referring to FIG. 2 , the source audio and video files can be sampled through the audio and video acquisition module, so that audio and video data can be collected from the source audio and video files. In the actual application process, the audio and video acquisition module can collect audio and video data according to the preset sampling period.

In this embodiment, the collected audio and video data can enter the encoder and be encoded with the encoding format and bit rate set in the encoder. After the encoded audio and video frames are packaged, they can be transmitted through the Real Time Messaging Protocol (RTMP) in the transmission control module to generate the push stream. The audio and video data to be transmitted can be temporarily stored in the sending buffer in the transmission control module, so that the real-time transmission of the audio and video data can be guaranteed.

In this embodiment, the stream data collection control module can be used to collect the feature information of the push stream according to the preset period, and the feature information can include the average bit rate, the average frame rate, the buffer data throughput, and the average TCP sending time of each frame .

Step S2: According to the current average TCP transmission time of each frame and the current buffer data throughput, the code rate of the next cycle is adjusted, and the adjusted code rate is fed back to the encoder as the target code rate.

In this embodiment, in order to optimize the streaming quality, the collected current feature information can be analyzed, so as to predict whether the code rate and frame rate of streaming need to be adjusted in the next cycle. Specifically, the code rate of the next period may be adjusted according to the current average TCP transmission duration of each frame and the current buffer data throughput. Please refer to Fig. 3, when the current average TCP sending duration of each frame is greater than the first preset risk value, it indicates that frame loss in the buffer may be about to occur, and the current code rate can be decreased according to the preset code rate step rate, and use the reduced code rate as the code rate for the next cycle.

When the current average TCP transmission duration per frame is less than or equal to the first preset risk value, it may continue to determine whether the current average TCP transmission duration per frame is greater than a second preset risk value. Wherein, the second preset risk value may be smaller than the first preset risk value.

If the current TCP average sending time of each frame is greater than the second preset risk value and the data occupancy rate of the current buffer is greater than the preset threshold, the situation of frame loss in the buffer may also occur. At this time, the preset code rate can be used The step size decreases the current code rate, and the reduced code rate is used as the code rate for the next cycle.

In this embodiment, the code rate in each cycle can be analyzed, and the code rate in the next cycle can be predicted in real time. In actual application, when the network fluctuates, it may cause the code rate to decrease for several consecutive cycles. In this case, in order to reduce the number of code rate drops, the rate of code rate drops can be increased accordingly. Specifically, it can be determined whether the code rate adjustment performed last time is a decrease in the code rate, and if so, it indicates that the current network condition is in a situation in which the code rate continues to decrease, so that the preset code rate step size can be increased to the preset value , so as to increase the speed of bit rate decrease, and decrease the current bit rate according to the increased preset value.

In this embodiment, the buffer data throughput may be analyzed to determine whether the code rate of the next cycle needs to be adjusted. Specifically, the relationship between the output data volume and the ingested data volume of the current buffer can be judged. When the output data volume is greater than or equal to the ingested data volume, the count value of the counter can be increased by one; when the output data volume is less than the ingested data volume When the amount of data is input, the count value of the counter can be decremented by one. In this way, according to the size of the count value in the counter, the relationship between the amount of output data and the amount of data swallowed in the previous few cycles can be known, so that the code rate of the next cycle can be adjusted according to the size of the count value in the current counter.

Specifically, when the count value in the counter is smaller than the first determination threshold, it indicates that the amount of data ingested in the buffer is greater than the amount of data output for several consecutive times. For example, the first determination threshold may be -3, so that it may indicate that the data ingested into the buffer is greater than the data outputted for more than three consecutive times. In this case, there will be a risk of frame loss, so the current code rate can be reduced according to the preset code rate step, and the reduced code rate will be used as the code rate for the next cycle, and the count value of the counter will be reset to zero .

When the count value in the counter is greater than the second determination threshold, it indicates that the data output volume of the buffer is greater than the data ingestion volume for several consecutive times. For example, the second determination threshold may be 3, so that it may indicate that the amount of data output from the buffer is greater than the amount of data ingestion for more than three consecutive times. In this case, it indicates that the network is in good condition and the bandwidth is sufficient, so the current code rate can be increased according to the preset code rate step, and the increased code rate can be used as the code rate for the next cycle, and the count value of the counter can be returned to zero.

In this embodiment, when judging the buffer data throughput, it can be judged whether the code rate adjustment performed last time is to reduce the code rate. Increase the preset code rate step to a preset value to increase the rate of code rate decrease, and decrease the current code rate according to the increased preset value;

In addition, it can also be judged whether the code rate adjustment performed last time was to increase the code rate. If so, it indicates that the current network condition is in a situation where the code rate continues to rise, so the preset code rate step size can be increased to a preset value , to increase the bit rate increase rate, and increase the current bit rate according to the increased preset value.

In this way, by analyzing the current average TCP transmission time of each frame and the current buffer data throughput, the transmission code rate of the next cycle can be predicted in real time, and the code rate can be switched flexibly. Since the analysis is performed every cycle, it also avoids excessive jumps in bit rate switching, ensuring a smooth viewing experience for users.

Step S3: Adjust the frame rate of the next cycle according to the current buffer data throughput, and determine the time interval between two frames according to the adjusted frame rate; The video data is filtered to generate audio and video data conforming to the adjusted frame rate.

In this embodiment, the current buffer data throughput can be analyzed in a manner similar to that in step S2, so that the frame rate of the next cycle can be adjusted. The adjusted frame rate can indicate the time interval between two frames, and through the time interval, the audio and video data entering the encoder can be filtered to generate audio and video data conforming to the adjusted frame rate.

It should be noted that, in this embodiment, the code rate adjustment process can be performed in the adaptive code rate control module, and the frame rate adjustment process can be performed in the dynamic frame rate control module. In various comparison processes, the thresholds for each comparison may be determined according to a TCP sending time prediction operator or a buffer size prediction operator. The adaptive code rate control module and the dynamic frame rate control module may be managed by an adaptive stream quality control module. The adaptive code rate module and the dynamic frame rate control module mentioned above can be turned on or off independently, and users can decide the application form of the technical solution according to their own needs. If the use scene requires to keep the high-quality streaming as possible, you can use the dynamic frame rate control module to give priority to adjusting the frame rate to control the amount of data entering the encoder per unit time, so as to control the amount of transmitted data; otherwise, the frame rate is more demanding. In high scenes, the adaptive bit rate control module can be used preferentially.

Step S4: Encoding and transmitting audio and video data according to the adjusted code rate and adjusted frame rate.

In this embodiment, the adjusted frame rate can adjust the audio and video data collected by the audio and video acquisition module, and the adjusted code rate can be applied to the encoder to adjust the data processed in the encoder. In this way, audio and video data can be encoded and transmitted according to the adjusted code rate and adjusted frame rate.

Please refer to Figure 4, the embodiment of the present application also provides a system for optimizing the quality of network streaming, and the system includes:

The stream data collection control module 100 is used to collect feature information of the push stream according to a preset period, and the feature information includes an average code rate, an average frame rate, buffer data throughput, and an average TCP transmission time per frame;

The adaptive code rate control module 200 is used to adjust the code rate of the next cycle according to the current average TCP transmission time of each frame and the current buffer data throughput, and feed back the adjusted code rate as the target code rate to the coder device;

The dynamic frame rate control module 300 is used to adjust the frame rate of the next cycle according to the current buffer data throughput, and determine the time interval between two frames according to the adjusted frame rate; based on the determined time interval Filter the audio and video data entering the encoder to generate audio and video data that conforms to the adjusted frame rate;

The transmission control module 400 is used for encoding and transmitting audio and video data according to the adjusted code rate and adjusted frame rate.

In the implementation manner of this application, the adaptive code rate control module 200 specifically includes:

The first judging unit is used to decrease the current code rate according to the preset code rate step when the current TCP average transmission duration of each frame is greater than the first preset risk value, and use the code rate after the decrease as the next period's code rate code rate;

A second judging unit, configured to judge whether the current average TCP transmission duration per frame is greater than a second preset risk value when the current average TCP transmission duration per frame is less than or equal to the first preset risk value;

The third judging unit is configured to decrease the current code rate according to the preset code rate step if the current average TCP transmission duration of each frame is greater than the second preset risk value and the data occupancy rate of the current buffer is greater than the preset threshold, And use the reduced code rate as the code rate for the next cycle.

In the implementation manner of this application, the adaptive code rate control module 200 specifically includes:

The throughput judging unit is used to judge the relationship between the output data volume and the ingested data volume of the current buffer. When the output data volume is greater than or equal to the ingested data volume, the count value of the counter is increased by one; when the output data volume When it is less than the amount of swallowed data, decrement the count value of the counter by one;

The code rate adjustment unit is configured to adjust the code rate of the next period according to the value of the count value in the current counter.

The embodiment of the present invention utilizes the interaction between the code rate and the frame rate adjustment to effectively control the amount of audio and video data sent in the case of network fluctuations on the one hand, relieve network congestion, and at the same time bring a smooth viewing experience to the playback terminal; on the other hand When the network quality is restored, it can ensure that the bit rate and frame rate of audio and video are pushed with the optimal quality configuration set, and it can switch between different bit rate and frame rate configurations in a smoother way, without It brings an abrupt sense of image quality fluctuations to the viewing end.

The foregoing description of various embodiments of the present application is provided for those skilled in the art for purposes of illustration. It is not intended to be exhaustive or to limit the invention to a single disclosed embodiment. As described above, various alterations and modifications of the present application will be apparent to those skilled in the art to which the above technologies pertain. Thus, while a few alternative implementations have been discussed in detail, other implementations will be apparent, or relatively readily arrived at, by those skilled in the art. This application is intended to cover all alternatives, modifications, and variations of the invention that have been discussed herein, as well as other embodiments that fall within the spirit and scope of the above application.

Claims (10)

1. a kind of optimization method of network plug-flow quality, it is characterised in that methods described includes：

The characteristic information of plug-flow is gathered according to predetermined period, the characteristic information includes average bit rate, average frame per second, number of buffer Duration is averagely sent according to handling capacity, per frame TCP；

Duration and current buffer data throughout are averagely sent according to current every frame TCP, the code check in next cycle is carried out Adjustment, and the code check after adjustment is fed back into encoder as target bit rate；

According to current buffer data throughout, the frame per second to next cycle is adjusted, and is determined according to the frame per second after adjustment Time interval between two frames；The time interval based on determination is filtered to the audio, video data for entering encoder, with Generation meets the audio, video data of the frame per second after adjustment；

Coding and the transmission of audio, video data are carried out according to the code check after adjustment and the frame per second after adjustment.

2. the optimization method of network plug-flow quality according to claim 1, it is characterised in that put down according to current per frame TCP Duration and current buffer data throughout are sent, the code check in next cycle is adjusted and specifically included：

When current every frame TCP averagely sends duration more than the first default value-at-risk, decline according to default code check step-length and work as Preceding code check, and using the code check after decline as next cycle code check；

When current every frame TCP averagely sends duration less than or equal to the described first default value-at-risk, judgement is described ought It is preceding averagely to send whether duration is more than the second default value-at-risk per frame TCP；

If the current duration that averagely sent per frame TCP is more than the second default value-at-risk and the data occupancy of current buffer During more than predetermined threshold value, current code check is declined according to default code check step-length, and using the code check after decline as next cycle code Rate.

3. the optimization method of network plug-flow quality according to claim 2, it is characterised in that methods described also includes：

Judge that the code check of last progress is regulateed whether to decline code check, if so, the default code check step-length is increased to default Value, and decline current code check according to the preset value after increase.

4. the optimization method of network plug-flow quality according to claim 1, it is characterised in that put down according to current per frame TCP Duration and current buffer data throughout are sent, the code check in next cycle is adjusted and specifically included：

The relation for judging the discharge data volume of current buffer and swallowing between data volume, when discharge data volume is more than or equal to When swallowing data volume, the count value of counter is added one；When discharge data volume, which is less than, swallows data volume, by the counting of counter Value subtracts one；

According to the size of nonce counter count value, the code check in next cycle is adjusted.

5. the optimization method of network plug-flow quality according to claim 4, it is characterised in that fallen into a trap according to nonce counter The size of numerical value, the code check for adjusting next cycle is specifically included：

When the count value in the counter is less than the first decision threshold, current code check is declined according to default code check step-length, and Using the code check after decline as next cycle code check, while by the counting value returns of the counter；

When the count value in the counter is more than the second decision threshold, current code check is improved according to default code check step-length, and Using the code check after raising as next cycle code check, while by the counting value returns of the counter.

6. the optimization method of network plug-flow quality according to claim 5, it is characterised in that methods described also includes：

Judge that the code check of last progress is regulateed whether to decline code check, if so, the default code check step-length is increased to default Value, and decline current code check according to the preset value after increase；

Judge that the code check of last progress is regulateed whether to improve code check, if so, the default code check step-length is increased to default Value, and improve current code check according to the preset value after increase.

7. the optimization method of network plug-flow quality according to claim 1, it is characterised in that the plug-flow is according to following sides Formula is generated：

The encoded device of the audio, video data gathered in advance is encoded, and by the audio, video data after coding after package It is transmitted by real-time messages host-host protocol RTMP, to generate the plug-flow.

8. a kind of optimization system of network plug-flow quality, it is characterised in that the system includes：

Flow data collects control module, the characteristic information for gathering plug-flow according to predetermined period, and the characteristic information includes flat Equal code check, average frame per second, buffer data handling capacity, every frame TCP averagely send duration；

Adaptive rate control module, for averagely sending duration and current buffer data throughput according to current every frame TCP Amount, is adjusted, and the code check after adjustment is fed back into encoder as target bit rate to the code check in next cycle；

Dynamic frame rate control module, for according to current buffer data throughout, the frame per second to next cycle to be adjusted, and Time interval between two frames is determined according to the frame per second after adjustment；The time interval based on determination is to the sound into encoder Video data is filtered, to generate the audio, video data for meeting the frame per second after adjustment；

Transmission control module, coding and biography for carrying out audio, video data according to the code check after adjustment and the frame per second after adjustment It is defeated.

9. the optimization system of network plug-flow quality according to claim 8, it is characterised in that the adaptive rate control Module is specifically included：

First judging unit, for when current every frame TCP averagely sends duration more than the first default value-at-risk, according to pre- If code check step-length declines current code check, and using the code check after decline as next cycle code check；

Second judging unit, for when the current duration that averagely sent per frame TCP is less than or equal to the described first default wind During the value of danger, judge the current duration that averagely sent per frame TCP whether more than the second default value-at-risk；

3rd judging unit, if presetting value-at-risk more than second and current for the current duration that averagely sent per frame TCP When the data occupancy of buffering area is more than predetermined threshold value, current code check is declined according to default code check step-length, and by the code after decline Rate as next cycle code check.

10. the optimization system of network plug-flow quality according to claim 8, it is characterised in that the self-adaption code rate control Molding block is specifically included：

Handling capacity judging unit, for the relation for judging the discharge data volume of current buffer and swallowing between data volume, when telling Go out data volume more than or equal to when swallowing data volume, the count value of counter is added one；Number is swallowed when discharge data volume is less than During according to amount, the count value of counter is subtracted one；

Code check adjustment unit, for the size according to nonce counter count value, adjusts the code check in next cycle.