CN100469072C - Multi-source streaming media transmission QoS control method - Google Patents

Abstract

The invention discloses a QoS control method for multi-source streaming media transmission, which is composed of five modules: multi-source streaming media description language interpreter, multi-source streaming media bandwidth allocation, fixed value bandwidth detection, data transmission control and RTP/RTCP. The MSML interpreter obtains the URL and transmission quality requirements of each channel of streaming media according to the MSML file; the multi-source streaming media bandwidth allocation module determines the transmission rate and encoding rate of each channel of streaming media according to the QoS requirements of streaming media transmission and the bandwidth value obtained by the detection module ; The data transmission control module controls the media transmission according to the VCR command and the buffer state; the RTP/RTCP module realizes the real-time and orderly transmission of the streaming media. The present invention can realize the coordinated control of multi-channel and multi-source concurrent streaming media transmission quality to reduce transmission delay and packet loss rate, and is suitable for multi-window video on demand, network video monitoring, network education courseware on demand, multi-angle/multi-scene live broadcast, etc. application field.

Description

Multi-source streaming media transmission QoS control method

technical field

The invention belongs to the technical field of computer design and application, relates to computer software, data transmission technology, multimedia technology and IP network, and particularly relates to a multi-source streaming media transmission QoS control method.

Background technique

At present, the main object of QoS research on streaming media transmission is single-channel streaming media, and the methods to solve the problem are mainly through buffer scheduling, network resource reservation and other methods. When performing single-source streaming media transmission, the effect is better, but when performing multi-source streaming media transmission QoS control, the effect of the above techniques is not ideal.

For the multi-source streaming media transmission QoS control method, the applicant retrieves 4 relatively closely related documents related to the present invention through a novelty search, and they are respectively:

1. Multi-channel video playback based on Qos

2. Research and design of streaming media system structure with QoS guarantee in IP network VOD

3. Realtime multimedia transport using multiple paths

4.An application-level QoS comparison of single-stream and multi-stream approaches in a wireless ad hoc network

The above-mentioned documents respectively proposed the multi-buffer and its scheduling algorithm used in the playback of multi-channel video; the realization of QoS based on RSVP; method for synchronous control. Among them, the multi-buffer control and scheduling algorithm can solve the jitter problem during playback to a certain extent, but it is powerless in network transmission; the resource reservation protocol RSVP can be used to reserve resources in the network between streaming media transmissions, This requires network support and cannot be dynamically adjusted according to changes in the network environment; the use of multi-path transmission methods can shield some network failures, but it does not determine whether the network can meet the transmission requirements of specific streaming media; separate audio and video transmission increases the Additional overhead, which can compound the problem when streaming from multiple sources.

In addition, the existing SMIL (Synchronized Multimedia Integration Language, Synchronized Multimedia Integration Language), which describes multi-source streaming media, focuses on describing the presentation information of multi-channel streaming media, and lacks the description of multi-source streaming media QoS control information. Current network bandwidth test methods, such as Pathchar algorithm or packet pair algorithm, all detect available bandwidth by inputting data into the network. This type of active data input method will affect the test results, especially in high-speed network environments. There will be a large deviation. In addition, such methods can only obtain the current maximum available bandwidth of the network, but cannot obtain the packet loss rate. The current bandwidth allocation method is mainly aimed at a single media stream, and cannot coordinate network bandwidth allocation for multiple concurrently transmitted media streams according to media stream types and QoS requirements. In addition, the existing bandwidth allocation method is based on the maximum available bandwidth, without considering the packet loss rate under the bandwidth.

Therefore, when performing streaming media transmission, especially multi-source streaming media transmission, the existing technology cannot solve the problem of accurately detecting the available bandwidth of the network in various regions and dynamically monitoring the transmission bandwidth of multi-source streaming media when the available network bandwidth changes at any time. The problem of optimizing distribution, which limits the quality of service of streaming media transmission to some extent.

Contents of the invention

The purpose of the present invention is to propose a multi-source streaming media transmission QoS control method, according to the specific conditions of each channel media in the multi-source streaming media and the available bandwidth of the network, the delay and packet loss rate of data transmission are reduced on the whole, which can Optimize the use of network resources to improve the transmission quality of multi-source streaming media.

The technical solution that realizes above-mentioned invention is: a kind of multi-source streaming media transmission QoS (Quality ofService) control method: carry out according to the following steps:

First, the multi-source streaming media language MSML interpreter is used to obtain the metadata information of the multi-source streaming media from the MSML file, and the streaming media type and transmission quality requirements in the information will be used for QoS control in the streaming media data transmission;

Then, use the fixed-value bandwidth detection module to judge the highest data rate that can be transmitted according to the current network conditions according to the RTP/RTCP protocol; according to this result, the multi-source streaming media bandwidth allocation module combines the streaming media type, transmission quality requirements, and supported code rate , to obtain the highest encoding rate and the transmission rate of encoding rate multiples available for each streaming media under the current network conditions; the fixed-value bandwidth detection module and the multi-source streaming media bandwidth allocation module operate alternately, and obtain according to the latest network available bandwidth at any time The highest available encoding rate and multiple transmission rate;

Finally, the data transmission control module sends commands to the RTP/RTCP module to update the encoding rate and multiple transmission rates according to the user's VCR command and bandwidth allocation results; the RTP/RTCP module performs real-time transmission of streaming media under the control of the data transmission control module.

Described multi-source streaming media language MSML is a kind of description language based on XML, supporting multi-source streaming media transmission QoS control and synchronous playback; the information described by MSML includes three types: one, the basic information of each channel streaming media, Including name, URL, type; 2. The transmission quality requirements of each channel of streaming media, including encoding rate range, maximum allowable delay, and maximum allowable packet loss rate; 3. Information presented during the playback of each channel of streaming media, including the initial size of the video window ,Location.

Described fixed-value bandwidth detection refers to adopting RTP/RTCP protocol to detect and obtain the packet loss rate of streaming media transmission at each rate between the lowest rate supported by the media and the highest available transmission rate of the network, and compare this packet loss rate with the streaming media Compared with the maximum packet loss rate allowed by the transmission, if it is smaller than the predetermined packet loss rate, it is considered that the current network condition can meet the data transmission rate at this rate, otherwise it is considered that the current network condition cannot meet the data transmission rate at this rate.

Described multi-source streaming media bandwidth allocation refers to adjusting the encoding rate and transmission rate of each stream media according to the result of fixed-value bandwidth detection, and the adjustment is divided into the following two situations: one, if a certain stream media comes from and other When the streaming media is from different streaming media servers, use the fixed-value bandwidth detection algorithm to detect the encoding rates of the streaming media at all levels from high to low until the highest encoding rate supported by the current network state is found. If the buffer is empty or the media in it The difference between the maximum and minimum current presentation time stamp PTS (PresentationTimeStamp) of the data packet is less than a specific threshold, then detect the multiplier rate of the rate supported by the network state; if the buffer is empty or the difference between the maximum and minimum PTS of the media data packet is greater than For a specific threshold, the encoding rate is used as the multiple rate; finally, the highest encoding rate supported by the current network status is used as the encoding rate of the streaming media, and the multiple rate is used as the actual transmission rate; 2. If several channels of streaming media come from the same stream The media server determines the candidate set of the total encoding rate of these media streams according to the priority of the media type, and selects the highest total encoding rate through the fixed-value bandwidth detection algorithm. If the minimum PTS difference is greater than a specific threshold, it is necessary to use a fixed value bandwidth to detect the corresponding multiple rate, otherwise, the highest total encoding rate is used as the total multiple rate; finally, the highest encoding rate supported by the current network state is used as the encoding rate of these media , and according to the proportion of each streaming media rate in the total encoding rate, the total multiplication rate is correspondingly decomposed into each channel of media as the actual transmission rate.

The present invention adopts the above-mentioned method, and can judge whether the current network state can meet the media transmission at a predetermined rate according to the basic information, synchronization information, media type, QoS requirements and presentation information of each channel in the multi-source streaming media. In the case of limited bandwidth, according to the type of streaming media and QoS requirements, bandwidth is allocated to multiple concurrent media data from different streaming media servers, so as to reduce the delay and packet loss rate of data transmission as a whole. Through the above strategies, network resources are optimally utilized to improve the transmission quality of multi-source streaming media.

Description of drawings

Fig. 1 is the composition of control mechanism of the present invention;

Fig. 2 is the working mechanism of the control mechanism of the present invention.

Detailed ways

In order to understand the present invention more clearly, the present invention will be further described in detail below in conjunction with the accompanying drawings.

Referring to Figure 1, first, the multi-source streaming language MSML interpreter is used to obtain the metadata information of the multi-source streaming media from the MSML file, and the streaming media type and transmission quality requirements in this information will be used in the streaming media data transmission QoS control;

Then, the fixed-value bandwidth detection module uses the RTP/RTCP protocol to judge the highest rate of data that can be transmitted under the current network conditions; according to this result, the multi-source streaming media bandwidth allocation module adjusts the bandwidth of each streaming media in combination with the streaming media type and transmission quality requirements. The encoding rate and the transmission rate of the multiple of the encoding rate; the fixed-value bandwidth detection module and the multi-source streaming media bandwidth allocation module operate alternately, and bandwidth allocation is performed according to the latest network conditions at any time;

Finally, the data transmission control module controls the streaming media transmission according to the user's VCR command and bandwidth allocation results; the RTP/RTCP module performs real-time transmission of the streaming media under the control of the data transmission control module, and completes it according to the amount of data in the buffer Retransmission of lost data.

1. Composition of multi-source streaming media transmission QoS control mechanism

The multi-source streaming media transmission QoS control mechanism is composed of five modules: MSML interpreter, constant value bandwidth detection, multi-source streaming media bandwidth allocation, data transmission control and RTP/RTCP.

The MSML interpreter is used to obtain the metadata information of multi-source streaming media from the MSML file; among them, the streaming media type and transmission quality requirements will be used for QoS control during transmission. The fixed value bandwidth detection module is used to judge whether the current network condition can meet the data transmission of a certain rate. The multi-source streaming media bandwidth allocation module determines the encoding rate and transmission rate of the media according to the fixed value bandwidth detection result, streaming media type, and transmission quality requirements. The data transmission control module controls the media transmission according to the VCR command and the bandwidth allocation result. The RTP/RTCP protocol module is used to realize real-time and orderly transmission of streaming media and retransmission of lost data packets.

2. Working mechanism of multi-source streaming media transmission QoS control mechanism

Referring to Figure 2, the multi-source streaming media transmission QoS control mechanism mainly works in an event-driven manner. After obtaining the streaming media type and QoS requirements information in the MSML file, the data transmission control module is the core, and the multi-source streaming media transmission QoS control is completed by processing three types of events: user VCR control, buffer status and timer; Among them, the transmission control module calls the multi-source streaming media distribution module and the RTP/RTCP module. The event types and processing procedures are as follows:

■VCR control

1) When starting to play, use the RTSP or MMS protocol to obtain the encoding rates of each streaming media at all levels, and notify the media server to transmit data at the highest rate, and perform multi-source streaming media bandwidth allocation at the same time;

2) When the playback is paused, the RTSP or MMS protocol is used to notify the media server to suspend the transmission of data;

3) When continuing to play, use RTSP or MMS protocol to notify the media server to continue to transmit data at the current encoding rate and transmission rate;

4) When the playback is stopped, the RTSP or MMS protocol is used to notify the media server to stop transmitting data and close the channel.

5) When dragging to a certain moment to play, clear the buffer, and notify the media server to transmit data at the highest rate, and perform multi-source streaming media bandwidth allocation at the same time.

■Buffer status

1) When the media delay in the buffer capacity is less than a specific threshold (5 seconds), re-execute multi-source streaming media bandwidth allocation;

2) When the delay of the media in the buffer capacity is greater than a specific threshold (30 seconds), the transmission rate of each channel of the multi-source streaming media is set as the encoding rate.

■Timer

Periodically re-execute multi-source streaming media bandwidth allocation.

In the process of bandwidth allocation for multi-source streaming media, the fixed-value bandwidth detection module is called to check whether the current network status meets the bandwidth allocated for each channel of media, and the bandwidth is re-allocated for each channel of media according to the detection results.

During the processing of events such as "start playing", "drag to a certain moment to play", and "buffer capacity media delay is less than a certain threshold", the transmission rate allocated for streaming media is generally a multiple of the transmitted media encoding rate, so The buffer can be filled quickly to reduce user waiting time.

3. The interface of each module of the multi-source streaming media transmission QoS control mechanism

■Interface of multi-source streaming media bandwidth allocation module

Bandinfo[]AllotBand(Mediainfo MediaArray[]);

This function determines the transfer rate of the multiplexed media and the encoding rate of the transferred media.

The parameter of the function AllotBand is an array of Mediainfo type, and the Mediainfo type is defined as follows:

Struct Mediainfo{

char[]url,

int Mediatype,

int Minband,

int Maxlost

}

Among them, url is the URL of the streaming media; Mediatype is the media type, including audio, video, and screen; Minband and Maxlost are the QoS requirements of the media, which are the allowed minimum bandwidth and maximum packet loss rate respectively.

The function AllotBand returns an array of the Bandinfo type, and the Bandinfo type is defined as follows:

Struct Bandinfo{

int coding_speed,

int trans_speed

}

Among them, coding_speed is the coding rate; trans_speed is the transmission rate.

■The interface of fixed value bandwidth detection module

Int Bandtest(intb, int lmax, int pmax, int pmin);

This function is used to judge whether the current network status meets the bandwidth requirement.

Among them, b is the fixed value bandwidth of a streaming media that needs to be checked, the maximum packet loss rate lmax allowed by the streaming media, the maximum and minimum PTS pmax and pmin of the media stream in the buffer; the return is the packet loss rate, when When -1 is returned, it means that the current network condition cannot satisfy the bandwidth b. The fixed-value bandwidth detection algorithm is invoked by the multi-source streaming media bandwidth allocation algorithm.

■Interface of RTSP/MMS module

Interface definition Interface Description void get_description(); Get the encoding rate of all levels of streaming media. void setup(); Open the transmission channel between the player and the media server. void vcr_control(vcr_typevcr); Perform VCR control on media transmission, including pause, resume, end, fast forward, positioning, etc. void close(); Close the transmission channel between the player and the media server.

callbackreceive_control(*bytecontrol) Receive control information as a callback function.

■Interface of RTP/RTCP module

4. XML-based multi-source streaming media language MSML

MSML is a simple XML-based markup language that can be used to describe the basic information, synchronization information, media type, QoS requirements and presentation information of each channel in multi-source streaming media.

The display of MSML is divided into two parts. The first part is the Header Section, where the global characteristics of the display are defined, such as the information to be displayed in the information window in the player, the status of the window during the process of each streaming media player, including the initial size and position of the video window.

The second part is the Body Section, which mainly specifies the synchronization information of the streaming media, the basic information of the media, including name, URL, and type, and sets the QoS requirements of the streaming media, including the encoding rate range, maximum delay time and maximum loss. packet rate.

The frame structure of the MSML file is as follows:

<msml>

<head>

<meta/>

                                                                                            ...

<layout>

                                                                                                                                                                                                                                                ...Defines the initialization information required for rendering during streaming playback...

</layout>

</head>

<body>

<par>

...Given the basic information of the streaming media...

...Setting related parameters required for transmission quality...

<par>

</body>

</msml>

The label description is as follows:

Label Function <meta/> Allows designers to set media metadata

<layout></layout> If layout data is specified, it must be contained between these two tags <root-layout/> It is included between <layout> and </layout>, specifying the displayed window size <region> Set the window area displayed by each streaming video <par></par> Play the streaming media files between <par> and </par> in parallel <video> Give the relevant information of the video file and set the transmission quality of the video <audio> Give the relevant information of the audio file and set the transmission quality of the audio <screen> Give information about the screen and set the transmission quality of the screen playback

The attributes of the meta tag are set as follows:

The attributes of the layout tag are set as follows:

Attributes value Function id any string specify the name of the window width Number or percentage in pixels Specifies the width of the window height Number or percentage in pixels Specifies the height of the window left, top Number or percentage in pixels Determine the position of the area relative to the upper left corner of the parent window bottom, right Number or percentage in pixels Determine the position of the area relative to the upper right corner of the parent window z-index any value Determine what level each window is in

Property settings for streaming media files

Attributes value Function url filename or full URL path Specify the location of the streaming media file region Display any zone names defined in Assign play area for streaming codespeed_from any valid value Specify the starting point of the encoding rate value codespeed_to any valid value Specifies the end point of the encoding rate value max_delay any valid value Specifies the maximum delay time

max_lose any valid value Specifies the maximum packet loss rate begin any legally scheduled time Specifies the playback time of the file end any legally scheduled time Specifies the end time of the file dur any legally scheduled time Specifies the duration of file playback id any string Assign the corresponding id in meta to this file

5. Fixed value bandwidth detection

During the on-demand process, since the encoding rate of media data is known, we only need to check whether the packet loss rate of media data transmitted at this rate is less than the maximum allowable packet loss rate in the streaming media QoS requirement under the current network conditions. Then the transmission effect of the media data can be judged.

In view of this, we design a fixed-value bandwidth estimation algorithm to detect the packet loss rate of a specific rate media stream under the condition of allowing a limited number of data retransmissions. The fixed value bandwidth estimation algorithm is as follows:

Algorithm input: the fixed value bandwidth b _v of a streaming media to be checked, the maximum packet loss rate l _max allowed by the streaming media, the maximum and minimum PTS p _max and p _min of the media stream in the buffer;

Algorithm output: packet loss rate 1; when -1 is returned, it means that the current network condition cannot satisfy the bandwidth b _v , that is, l _max < l _r ;

Intermediate variables: the duration of transmission t _s , the number of RTP packets received at time t _s m, the size of RTP packets s, the number of allowed retransmissions n, and the fixed delay in retransmission t _c .

The algorithm is described as follows:

STEP1. If the buffer is empty, or (p _max -p _min )/(t _c +s/b _v )<1, then n=0; otherwise n=[(p _max -p _min )/(t _c + s/b _v )], f(x)=[x] is a Gaussian function, and n≥1 indicates that data retransmission is allowed.

STEP2. If n=0, within t _s time, the number of media data packets sent by the media server to the player with a rate of b _v is (t _s ×b _v )/s, and the packet loss rate is l=1-(t _s ×b _v )/(s×m); if l≤l _max , the current network condition meets the transmission rate of media data, return l, otherwise, return -1;

STEP3. If n≥1, introduce a retransmission mechanism. At this time, the packet loss rate is l ⁿ . If l ⁿ ≤ l _max , the current network conditions can meet the transmission rate of media data, and return l ⁿ ; if l ⁿ > l _max , the current network condition does not meet the transmission rate of media data, return -1.

Compared with other algorithms, the main features of this method are:

1) Obtain the ability of the link to transmit a predetermined rate bandwidth without estimating the entire available bandwidth of the link;

2) Do not actively input data to the network, and the test process is non-disturbing to the network;

3) The measurement process truly reflects the transmission process of link data;

4) There is no problem of being interfered by other traffic on the link.

6. Multi-source streaming media bandwidth allocation

The various media that make up the multi-source streaming media may come from multiple media servers, and because of the different media types, the requirements for media transmission are also different. For audio streaming media, low packet loss rate and jitter are required, and it occupies an important position in streaming media; for screen streaming media, high resolution is required, but the frame rate requirement is not high, and it occupies more in streaming media than video. important position. In addition, each channel of media has specific QoS requirements.

In order to obtain good performance quality under limited network conditions, we have designed a multi-source streaming media bandwidth allocation algorithm to allocate bandwidth to multiple concurrent media data from different streaming media servers, thereby reducing the overall data transmission cost. Latency and packet loss rate.

The multi-source streaming media bandwidth allocation algorithm classifies the bandwidth for each channel of media in a heuristic manner according to the type of media, QoS requirements, buffer and network status. Multi-source streaming media bandwidth allocation algorithm is as follows:

Algorithm input: Streaming media URL _i , media type Mediatype _i , minimum bandwidth Minband _i , maximum packet loss rate Maxlost _i , current coding rate CodingSpeed _{i, t} , multiples of current transmission rate and coding rate for each channel of media in n channels of streaming media Times _{i, t} , the current packet loss rate Lost _{i, t} of each channel in the n-channel streaming media, the time delay in the buffer zone d _{i, t} , d _{i, t} = p _{max, i, t} -p _{min, i, t} , p _{max, i, t} and p _{min, i, t} are the maximum and minimum PTS of the media stream currently in the buffer, i∈[l,n].

Algorithm output: CodingSpeed _{i, t+1} of each channel of media in n channels of streaming media, times _{i, t+1} of transmission rate and encoding rate, i∈[l, n].

STEP1. Let M be the collection of n-channel streaming media, is a set that satisfies the following conditions: for any streaming media in M _E , the source of any streaming media in M is different from the media server. Execute the following operations in parallel for all streams in M _E :

Let the serial number of the media stream be j.

STEP1.1 When the time delay d _{j, t} of the media in the buffer zone is 0, then CodingSpeed _{j, t+1} is set to the highest coding rate supported by the media, and Times _{j, t+1} is set to 2.

STEP1.2 When the delay of the media in the buffer is 0<d _{j, t} ≤ d:

If Lost _{i, t} ≥ Maxlost _i and Times _{i, t} > 1.5, set CodingSpeed _{j, t+1} to CodingSpeed _{j, t} , set Times _{j, t+1} to Times _{j, t} -0.5;

If Lost _{i, t} ≥ Maxlost _i and Times _{i, t} ≤ 1.5 and CodingSpeed _{j, t} > Minband _i , set CodingSpeed _{j, t+1} as the coding rate one level lower than CodingSpeed _{j, t} , set Times _{j, t+ 1} is 2;

If Lost _{i, t} ≥ Maxlost _i and Times _{i, t} ≤ 1.5 and CodingSpeed _{j, t} ≤ Minband _i , set CodingSpeed _{j, t+1} to Minband _i , set Times _{j, t+1} to 1;

If Lost _{i, t} ≤ Maxlost _i and Times _{i, t} ≤ 2.5, set CodingSpeed _{j, t+1} to CodingSpeed _{j, t} , set Times _{j, t+1} to Times _{j, t} +0.5;

If Lost _{i, t} ≤ Maxlost _i and Times _{i, t} > 2.5 and CodingSpeed _{j, t} is not the highest coding rate, set CodingSpeed _{j, t+1} as a higher coding rate than CodingSpeed _{j, t} , set Times _{j, t+1} is 2;

If Lost _{i, t} ≤ Maxlost _i and Times _{i, t} >2.5 and CodingSpeed _{j, t} is the highest coding rate, set CodingSpeed _{j, t+1} to CodingSpeed _{j, t} , set Times _{j, t+1} to Times _{i, t} ;

When the delay of the media in the buffer d<d _{j, t} :

If Lost _{i, t} ≥ Maxlos _t i and CodingSpeed _{j, t} >Minband _i , set CodingSpeed _{j, t+1} to a coding rate lower than CodingSpeed _{j, t} , set Times _{j, i+1} to 1;

If Lost _{i, t} ≥ Maxlos _t i and CodingSpeed _{j, t} ≤ Minband _i , set CodingSpeed _{j, t+1} to Minband _i , set Times _{j, t+1} to 1;

If Lost _{i, t} ≤ Maxlost _i and CodingSpeed _{j, t} is not the highest coding rate, set CodingSpeed _{j, t+1} to a higher coding rate than CodingSpeed _{j, t} , set Times _{j, t+1} to 1;

If Lost _{i, t} ≤ Maxlost _i and Times _{i, t} > 2.5 and CodingSpeed _{j, t} is the highest coding rate, set CodingSpeed _{j, t+1} to CodingSpeed _{j, t} , and set Times _{j, t+1} to 1.

STEP2. Let M _M =MM _E , divide M _M into several subsets M _{M, k} , k∈[l, m]. The media in each subset comes from the same media server, and the media in each subset comes from different media servers.

STEP3. In M _{M, k} , each medium should have the same packet loss rate, the same multiplier rate, and the same delay in the buffer zone. Let the current packet loss rate of M _{M, k} be Lost _{M, k, t} , the multiple of the transmission rate and the coding rate is Times _{M, k, t} , and the time delay is the sum of the current coding rates of each media in d _{M, k, t} , M _{M, k} CodingSpeed _{M, k, t} . Lost _{M, k, t} , d _{M, k, t} are the same as the packet loss rate and buffer delay of any media in M _{M, k} . For each M _M,k do the following in parallel:

STEP3.1 Determine the priority of each channel of media in M _M,k , where the priority of audio stream is 3, the priority of screen media stream is 2, the priority of video media stream is 1, and the same type of media has the same priority.

STEP3.2 When the time delay of the media in the buffer zone d _{M, k, t} = 0, for any media j in M _{M, k} , set CodingSpeed _{j, t+1} as the highest coding rate supported by the media , Set Times _{j, t+1} to 2.

STEP3.3 When the delay of the media in the buffer is 0<d _{M, k, t} ≤ d:

If there is a channel of media j in M _{M, k} , so that Lost _{M, k, t} ≥ Maxlost _j and the current Times _{j, t} > 1.5, keep the transmission rate of all media in M _{M, k} unchanged, set Times _{j, t +1} for Times _j,t -0.5;

If there is a media j in M _{M, k} , so that Lost _{M, k, t} ≥ Maxlost _j and Times _{i, t} ≤ 1.5, then find the media stream with the lowest priority among all media streams whose current encoding rate is greater than the minimum allowable rate , assuming it is media l, set CodingSpeed _{l, t+1} to be a coding rate one level lower than CodingSpeed _{l, t} , set M _{M, k} to double the rate of all media. If M _{M, k} does not have a media stream whose encoding rate is greater than the minimum allowable rate, then set the encoding rate of all media in M _{M, k} to the corresponding minimum allowable rate, and set the multiplication rate of all media in M _{M, k} to 1.

If any media j in M _{M, k} satisfies Lost _{M, k, t} ≥ Maxlost _j , and Times _{i, t} ≤ 2.5, set CodingSpeed _{j, t+1} to CodingSpeed _{j, t} , set Times _{j, t+ 1} for Times _{j, t} +0.5;

If any media j in M _{M, k} satisfies Lost _{M, k, t} ≥ Maxlost _j , and Times _{i, t} > 2.5, then find the highest priority among all media streams whose current encoding rate is not the highest encoding rate Media stream, assuming it is media l, set CodingSpeed _{l, t+1} to be a coding rate one level higher than CodingSpeed _{l, t} , set M _{M, k} to 2 for all media multipliers. If M _{M, k,} the current encoding rate of all media is the highest encoding rate, keep the encoding rate and multiplication rate of each channel unchanged. When the delay of the media in the buffer d<d _{M, k, t} :

If there is a media j in M _{M, k} , so that Lost _{M, k, t} ≥ Maxlost _j , then find the media stream with the lowest priority among all the media streams whose current encoding rate is greater than the minimum allowable rate, assuming it is media l, set CodingSpeed _{l, t+1} is the coding rate one level lower than CodingSpeed _{l, t} , set M _{M, k} to be 1 for all media. If M _{M, k} does not have a media stream whose encoding rate is greater than the minimum allowable rate, then set the encoding rate of all media in M _{M, k} to the corresponding minimum allowable rate, and set the multiplication rate of all media in M _{M, k} to 1.

If any media j in M _{M, k} satisfies Lost _{M, k, t} ≥ Maxlost _j , then find the media stream with the highest priority among all the media streams whose current encoding rate is not the highest encoding rate, assuming it is media l, Set CodingSpeed _{l, t+1} as the coding rate one level higher than CodingSpeed _{l, t} , and set M _{M, k} to 1 for all media multipliers. If the current encoding rate of all media _{in M, k} is the highest encoding rate, keep the encoding rate of each media, and set the multiplication rate of all media in M _{, k} to 1.

Claims (4)

1, a kind of multi-source flow media transmission QoS (Quality of Service) control method is characterized in that, carries out according to the following steps:

At first, adopt multi-source flow media language MSML interpreter to obtain the metadata information of multi-source flow media from the MSML file, the Streaming Media type in this information, transmission quality requirements will be used for the QoS control of transmission of flow media data;

Then, adopt the definite value bandwidth detection can transmit the flank speed of data according to RTP/RTCP agreement judgement current network conditions; According to this result, the multi-source flow media bandwidth allocation module is in conjunction with Streaming Media type, transmission quality requirements, the code rate supported, obtains the available the highest code rate under current network conditions of every road Streaming Media and the transmission rate of code rate multiple; Definite value bandwidth detection and multi-source flow media bandwidth allocation module alternate run at any time according to the latest network available bandwidth, obtain the highest available code rate and multiple transmission rate;

At last, the Data Transmission Controlling module sends renewal code rate and the instruction of multiple transmission rate according to user's VCR order and allocated bandwidth result to the RTP/RTCP module; The RTP/RTCP module is carried out the real-time Transmission of Streaming Media under the control of Data Transmission Controlling module.

2, multi-source flow media transmission QoS control method according to claim 1 is characterized in that: described multi-source flow media language MSML is a kind of descriptive language based on XML, the control of support multi-source flow media transmission QoS and synchronous playing; The information that MSML describes comprises three types: one, the essential information of each road Streaming Media comprises title, URL, type; Two, the transmission quality requirements of each road Streaming Media comprises code rate scope, maximum delay, the maximum packet loss that allows of allowing; Three, the presentation information in each road streaming media playing process comprises video window initial size, position.

3, multi-source flow media transmission QoS control method according to claim 1, it is characterized in that: described definite value bandwidth detection is meant that the minimum code rate that adopts the RTP/RTCP protocol detection and obtain to support from Streaming Media is to the dirty media delivery packet loss of each code rate the highest available transmission rate of network, and the maximum packet loss that the transmission of this packet loss and Streaming Media is allowed compares, if it is littler than predetermined packet loss, think that then current network conditions can satisfy the transfer of data under this speed, otherwise just think that current network conditions can not satisfy the transfer of data under this speed.

4, multi-source flow media transmission QoS control method according to claim 1, it is characterized in that: described multi-source flow media allocated bandwidth is meant the result according to the definite value bandwidth detection, adjust the code rate and the transmission rate of each road Streaming Media, adjust and divide following two kinds of situations to handle: one, when if certain road Streaming Media comes from the streaming media server different with other Streaming Media, then adopt definite value bandwidth detection algorithm to detect the code rates at different levels of this road Streaming Media from high to low, up to the highest code rate that finds current network state to support, if the difference that buffering area is maximum empty or wherein media data packet and minimum current presentative time stabs PTS (Presentation TimeStamp) then detects times speed of this speed that network state is supported less than certain threshold level; If the difference of the maximum of media data packet and minimum PTS is greater than certain threshold level, with this speed as a times speed; At last, the highest code rate that current network state is supported is as the code rate of this road Streaming Media, and times speed is as actual transfer rate; Two, if wherein several roads Streaming Media comes from same streaming media server, then determine the candidate set of the total code rate of these Streaming Medias according to the priority of medium type, and by the highest total code rate of definite value bandwidth detection algorithm selection, if when buffering area be sky or wherein the difference of the maximum of media data packet and minimum PTS less than certain threshold level, adopt corresponding times of speed of definite value bandwidth detection, if the difference of the maximum of media data packet and minimum PTS is greater than certain threshold level, with the highest total code rate as total times of speed; At last, the highest code rate that current network state is supported is as the code rate of these Streaming Medias, and according to the shared proportion of each road Streaming Media speed in total code rate total times of speed correspondingly decomposed each road Streaming Media as actual transfer rate.

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