CN1852429A - Video-code-flow grouped transmission method and system - Google Patents

Abstract

The present invention relates to a video code stream packet transmission method, comprising the following steps: (a) encapsulating the code stream generated by an encoder into RTP packets at the server end according to network conditions, and transmitting the RTP packets to the receiving end; (b) ) The receiving end decodes the received RTP packet to obtain video data. The invention also provides a corresponding video code stream packet transmission system. The invention improves the transmission efficiency by encapsulating code streams into RTP packets of different sizes according to network conditions, minimizes end-to-end delay, and improves video transmission quality. In addition, the present invention ensures the video quality by setting important level parameters.

Description

Video stream packet transmission method and system

technical field

The present invention relates to the field of data transmission, and more specifically, to a method and system for packet transmission of video streams.

Background technique

With the development of IP network technology and video and audio coding technology, more and more video and audio content in various coding formats are transmitted through the IP network.

Existing video coding standard technologies include H.261, H.263, H.264 of ITU-T and Part 2 of MPEG1, Part 2 of MPEG2 and Part 2 of MPEG4 of ISO/IEC. Existing video transmission systems include video conferencing and video telephony. These systems include ITU-T recommendations H.320, H.323, and H.324, etc., which respectively stipulate the operation methods of video conferencing and video telephone system transmission on the network.

AVS (Audio Videocoding Standard, digital audio and video codec technology standard) is formulated to meet the needs of motion image compression technology in applications such as digital TV broadcasting, Internet streaming media, and multimedia communications. Part 2 (P2) involves video Coding technology. AVS-P2 itself divides the original video data into strips, takes macroblock as the basic coding unit, and compresses the output code stream through functional modules such as prediction, motion estimation, differential transformation, frequency domain transformation, quantization and entropy coding. Grammatically, a layered structure is adopted, which can be divided into image sequence layer, image layer, strip layer, and macroblock layer. At each syntax level, there is a 4-byte start code identifier. A stripe is a unit for recovering from data loss or corruption. This code stream also includes codec parameters, such as image size, reference image list, initial quantization parameters, etc. In remote video communication, the compressed AVS-P2 video code stream is transmitted to the receiving end through the network, and the receiving end decodes and plays it according to the parameters in the code stream.

In order to adapt to the characteristics of the IP network and effectively transmit the above-mentioned video and audio content, the IP transmission grouping method of the MPEG2 coded content, the IP transmission grouping method of the MPEG4 coded content, and the IP transmission grouping method of the H.263 and H.263+ coded content have emerged , IP transmission grouping method of H.264 coded content, etc. These technologies minimize the end-to-end delay by adapting to the characteristics of the video content, thereby improving the quality of video communication.

For example, RFC 2250 provides a video packetization scheme primarily for MPEG2. But the main disadvantage of this scheme is that there are too many information header parameters in the video packet, which affects the further improvement of transmission efficiency. In addition, the Act mainly relies on the underlying transmission protocol to ensure the quality of transmission, and lacks measures such as key content protection for video characteristics.

RFC 3984 provides a video grouping scheme mainly for the H.264 standard, which makes full use of the network abstraction layer unit header (NALU) specified by the H.264 standard to form packets of different structures. However, this solution is mainly aimed at video data that conforms to the H.264 syntax. When it is used to transmit non-H.264 standard video data, the parameters of the information header are relatively small, which affects the effective transmission of video data, especially some video extension data. transmission. Before grouping, non-H.264 video data must be converted to a certain extent, and the decoder at the receiving end must also reconstruct these data.

Contents of the invention

The technical problem to be solved by the present invention is to provide a new video stream packet transmission method and system for the above-mentioned defects of low streaming media transmission efficiency in the prior art.

The technical solution adopted by the present invention to solve its technical problems is: construct a kind of video stream grouping transmission method, comprise the following steps:

(a) encapsulating the code stream generated by the encoder into RTP packets according to network conditions at the server end, and sending the RTP packets to the receiving end;

(b) The receiving end decodes the received RTP packet to obtain video data.

In the video code stream packet transmission method of the present invention, said step (a) includes:

(a1) At the server end, according to the service request of the receiving end, the code stream from the encoder is placed into the packaging buffer located at the server end;

(a2) monitor the network condition through the application controller at the server end, and use the packer to encapsulate the code stream generated by the encoder into RTP packets according to the network condition;

(a3) Transmitting the RTP packet to the receiving end.

In the video code stream packet transmission method of the present invention, said step (b) includes:

(b1) placing the received RTP packet into the receiving buffer at the receiving end at the receiving end and waiting for a specified time;

(b2) Send the RTP packet in the receiving buffer to the decoder for decoding.

In the video code stream packet transmission method of the present invention, said step (a) includes the steps of adding an importance level parameter to the RTP packet and transmitting the RTP packet according to the importance level, and said step (b) includes the step of adding an important level parameter according to the RTP The importance level parameter in the packet preferentially decodes the RTP packet with high importance level.

In the video code stream packet transmission method of the present invention, the code stream is an AVS-P2 code stream, and the RTP packet includes RTP header files, syntax layer elements of the AVS-P2 code stream and media data.

In the video code stream packet transmission method of the present invention, the network conditions include one or more of network bandwidth, code rate and packet loss rate, and in the step (a), the RTP generated by the server The size of the packet increases with the increase of the network bandwidth, increases with the increase of the code rate, and decreases with the increase of the packet loss rate.

In the video code stream packet transmission method of the present invention, the RTP packet includes an RTP header and an RTP payload, wherein the RTP payload includes RTP payload header information and media data.

In the video code stream packet transmission method of the present invention, the RTP payload header information includes first field priority flag, important first field flag and progressive frame flag parameters.

In the video code stream packet transmission method of the present invention, the step (a) includes the steps of using the transfer encoding time information and multi-video data packet encapsulation to perform interleaved grouping and video source switching.

The present invention also provides a video code stream grouping transmission system, including a server and a receiving end connected through a network, the server includes a code stream source for providing a code stream, an application controller for monitoring network conditions, and a The code stream from the code stream source is encapsulated into a packer for RTP packets, and the receiving end includes a decoder for decoding the RTP packets from the server, wherein the packer converts the code stream according to the results of monitoring the network of the application controller Streams are encapsulated into RTP packets of a certain size.

In the video code stream packet transmission system of the present invention, the code stream source is an encoder or an encoder group that encodes a video file into a code stream, or a storage device storing a code stream.

In the video code stream packet transmission system of the present invention, the receiving end includes a receiving buffer for buffering RTP packets from the server, and the receiving buffer buffers the received RTP packets for a predetermined time and sends them to the decoder device decoding.

In the video code stream packet transmission system of the present invention, the RTP packet includes an RTP header and an RTP payload, wherein the RTP payload includes RTP payload header information and media data.

In the video stream packet transmission system of the present invention, the RTP packet includes a parameter identifying the importance level of the RTP packet, and the encoder preferentially decodes the RTP packet with a high importance level.

In the video code stream packet transmission system of the present invention, the code stream is an AVS-P2 code stream, and the RTP packet includes RTP header files, syntax layer elements and media data of the AVS-P2 code stream

In the video code stream packet transmission system according to the present invention, the RTP packet includes parameters identifying the occurrence of extended data, and the network conditions include one or more of network bandwidth, code rate and packet loss rate, and the server The size of the generated RTP packet increases with the increase of the network bandwidth, increases with the increase of the code rate, and decreases with the increase of the packet loss rate.

The video code stream packet transmission and system of the present invention, by encapsulating the code stream into RTP packets of different sizes according to network conditions, improves transmission efficiency, minimizes end-to-end delay, and improves video transmission quality. In addition, the present invention ensures the video quality by setting important level parameters.

In addition, the present invention improves the error concealment and the load balancing ability of the service system by transmitting the encoding time information and encapsulating multiple video data packets to perform interleaved grouping and video source switching.

Description of drawings

The present invention will be further described below in conjunction with accompanying drawing and embodiment, in the accompanying drawing:

Fig. 1 is the structural representation of video code stream packet transmission system of the present invention;

Fig. 2 is the structural representation of the first embodiment of the RTP packet in the video code stream packet transmission system of the present invention;

Fig. 3 is the second embodiment structural representation of the RTP packet in the video code stream packet transmission system of the present invention;

Fig. 4 is the RTP packet schematic diagram of switching video source in the video stream packet transmission system of the present invention;

Fig. 5 is the schematic diagram of the RTP payload header in the video stream packet transmission system of the present invention;

Fig. 6 is the flow chart of the video code stream packet transmission method of the present invention;

Fig. 7 is the flow chart of encapsulation AVS-P2 stream step among Fig. 6;

FIG. 8 is a flow chart of the steps of receiving and decoding the playing RTP packet in FIG. 6 .

Detailed ways

As shown in FIG. 1 , it is a schematic structural diagram of the video code stream packet transmission system of the present invention. In the following, an AVS-P2 code stream is taken as an example to illustrate the video code stream packet transmission system of the present invention.

In the present embodiment, the video stream packet transmission system of the present invention is based on the AVS-P2 video communication system, which includes an AVS-P2 video Internet service system (abbreviated as the server) 11 and a plurality of devices connected to the server through the network. The AVS-P2 video Internet receiving terminal system (abbreviated as the receiving end, only one is shown in the figure) 12. Among them, the server 11 is used to send streaming media according to the request of the receiving end, and the receiving end 12 decodes the streaming media from the server 11 to play video or audio files. The server 11 may also include multiple backup video Internet service systems.

The server 11 includes an encoder 112 , a packager 113 and an application server 111 . The encoder 112 is used to encode the original image into an AVS-P2 code stream, and send the generated code stream to the packer 113 . The application controller 111 is used to monitor the status of the network 13, and the network status may be one or more of the bandwidth, code rate, and packet loss rate of the network 13. The packer 113 is used to package the code stream from the encoder 112 according to the monitoring results of the application controller 111 into RTP (Real Time Transport Protocol, real-time transport protocol) packets. For example, when the application controller 111 detects that the network bandwidth is insufficient, each RTP packet generated by the packer 113 contains a small amount of data, and when the network bandwidth is large, each RTP packet generated by the packer 113 contains a large amount of data. After the encapsulation is completed, the server 11 sends the encapsulated RTP packet to the receiving end 12 through the network 13 . In this embodiment, the size of the RTP packet generated by the packer 113 increases with the increase of the network bandwidth, increases with the increase of the code rate, and decreases with the increase of the packet loss rate. The structure of the RTP packet will be described in detail in Figure 2 and Figure 3. In addition, the server 11 can improve the correct decoding capability of the receiving end 12 through repeated transmission.

Of course, the server 11 may also include multiple encoders; or the encoder 112 is not included, and the AVS-P2 code stream is directly stored in the memory, and the packer 113 directly packs the AVS-P2 code stream in the memory.

A packing buffer may also be set between the packer 113 and the encoder 112 for buffering the code stream, and the packer 113 sequentially takes out the code stream from the packing buffer for packing.

The receiver 12 includes a decoder 122 and a player 123 . The decoder 122 is used to decode the received RTP packet, and transmit the decoded video data to the player 123 for playing.

In order to improve the playback effect of the player 123, a receiving buffer 121 can be set at the receiving end 12. Through the receiving buffer 121, the RTP packet from the server 11 is buffered for a predetermined time (can be set according to network conditions), and then sent to the decoder 122 for decoding, thereby avoiding the playback caused by network delay and packet loss. The pause ensures the continuity of video data playback.

Since the encoding time of the RTP packets received by the receiving end 11 may be interleaved, the memory required by the receiving buffer 121 may be larger than the decoding buffer specified by the corresponding profile level in the AVS-P2 video standard. The initial size of the receiving buffer 121 can be estimated by the server 11 according to the packet generation method, and then transmitted to the receiving end 12 in the form of MIME parameters. After the decoder 122 obtains the RTP packets buffered in the receiving buffer 121, it will decode and reassemble them in time order according to the accompanying parameter information.

As shown in FIG. 2 , FIG. 3 , and FIG. 5 , they are schematic structural diagrams of RTP packets in the video code stream packet transmission system of the present invention.

After receiving the service request sent by the receiving end 12, the server 11 obtains the AVS-P2 code stream from the encoder 112 (or obtains the pre-generated AVS-P2 code stream from the memory). Since the image of AVS-P2 is generally relatively large, and the network bandwidth is limited, it is necessary to transmit the code stream in packets. Normally, the code stream will be divided into several RTP packets. If the characteristics of AVS-P2 compression coding are not considered, blind grouping will affect the efficiency and quality of transmission. It may not be able to adapt to the size of the underlying packet data unit (PDU) and be forced to re-segment, resulting in low transmission efficiency. The loss of previously transmitted data will affect the playback of subsequent data. Therefore, the generated AVS-P2 stream enters the packetizer 113 before being sent to the network.

Since the AVS-P2 code stream has a hierarchical structure, different packet structures can be used, and parameters are used to indicate the content of the code stream in the group. In each RTP packet, it uses a combination of parameters to indicate whether the contents of the packet contain AVS-P2 syntax layer elements, such as sequence header information, image header information, slice data, or multiple Various types of content, etc., are grouped in the form of header parameter information plus subsequent media data. In this way, the bandwidth can be fully utilized and the transmission efficiency can be improved.

When network resources are relatively tight, a code stream (or even a strip) can be divided into several packets for transmission to prevent large data from being unable to be carried due to insufficient bandwidth. As shown in Figure 2, parameter 1, parameter 2, etc. are the syntax elements of the AVS-P2 code stream (such as sequence header information, image header information, slice data, etc.), and the media information is the code stream corresponding to the combination of the above syntax elements A fragment contains video data, in this case an RTP packet may contain one or more parameters. When bandwidth permits, multiple code streams can also be encapsulated in one RTP packet, which can also be indicated by parameters. As shown in Figure 3, parameters 1', 2', etc. are video data serial numbers, while media information 1', 2' are video data corresponding to the code stream. This real-time encapsulation based on network conditions improves data transmission efficiency.

When the bandwidth allows and the processing memory of the service segment 11 and the receiving end 12 is sufficient, it is even possible to encapsulate multiple code streams with different encoding timings in one RTP packet, and indicate the decoding of the data by a time parameter before the code stream order. For example: under normal circumstances, the content in the packet is encapsulated in sequence according to the encoding time of the original code stream and then sent to the network, "1, 2, 3, 4, 5, 6, 7, 8, 9, ..."; To improve the error concealment ability, by setting parameters, the encapsulation code stream can be interleaved in time, such as sending packets in the following encoding order: "1, 4, 2, 5, 3, 6, 7, 10, 8, ..." .

When there are multiple service requests, in order to balance the load of the server 11, when the video source needs to be switched, the AVS-P2 code stream can also be continuously relayed at any starting point by periodically transmitting the sequence header information, such as Figure 4 shows. The code stream following the sequence header is responsible for interpretation by the sequence header.

As shown in FIG. 5 , it is a schematic diagram of a specific RTP payload header. The RTP payload consists of the RTP payload header and media data. The parameters in the RTP payload header are described as follows:

PD: It is used to identify the display time sequence of the current image (8 bits), and its value ranges from 0 to 255. For all RTP packets of the same image, the value of this parameter is the same.

AN: It is used to start bit N (1 bit). When the subsequent bit N is used to convey the change of the image header information in the AVS-P2 payload, this bit is set to 1; when the subsequent bit N is not used, this bit is set to 1. bit is set to 0.

N: Used to identify the presence of a new picture header (1 bit). This bit is set to 1 when the information contained in the previous picture header cannot be used to reconstruct the picture header for the current picture (ie, the previous picture of the same type was encoded with a different parameter set than the current picture). This bit remains unchanged for all RTP packets belonging to the same image. In this way, when reconstructing an image (one field or one frame), it can be judged whether the information required for reconstruction is contained in the current image (N=1) or before by detecting the bit N in any RTP packet that makes up the image. in the image (N=0).

S: Used to identify whether the sequence header appears (1 bit). Normally set to 0, this bit is only set to 1 each time the AVS-P2 sequence header is present.

B: Used to identify the start of a stripe (1 bit). This bit is set to 1 when the beginning of the RTP payload is a slice start code, or when the slice start code is preceded by only one or more sequence headers and/or picture headers.

E: used to identify the end of the stripe (1 bit). This bit is set to 1 when the last byte of the RTP payload is the slice end code for AVS-P2.

P: Used to identify the image type (3 bits). For example, a value of 1 indicates an I frame, a value of 2 indicates a P frame, and a value of 3 indicates a B frame. This value is constant for each RTP packet of a given image. A value of 000 is prohibited, and a value of 100-111 is reserved to support future extensions specified by AVS.

PS: Used to identify the image structure (2 bits). When PS is "0", it means that the encoded data of the two fields of the current image appears sequentially; when the value is "1", it means that the encoded data of the two fields of the current image appears alternately (please refer to the explanation of picture_structure in AVS-P2).

TF: First field priority flag (1 bit) (please refer to the explanation of the top_field_first field in AVS-P2).

R: important first field flag (1 bit) (please refer to the explanation of the repeat_first_field field in AVS-P2).

G: Progressive frame flag (1 bit). A value of "0" indicates that the two lines of the frame are interlaced fields with a field time gap between them. If the bit value is "1", it means that the two fields of the frame actually come from the same moment (please refer to the provisions of the progressive_frame field in AVS-P2).

PHD: The flag bit (1 bit) appears in the image header. Normally set to 0, this bit is only set to 1 every time an AVS-P2 picture header (i_picture_header or pb_picture_header) appears. This bit is used to detect whether the image header is present in the RTP packet.

T: Extended data occurrence flag (1 bit), this parameter is related to the size of the RTP packet. AVS-P2 stipulated sequence_display_extension, copyright_extenstion, camera_parameter_extenstion and other extended signs. When these extended data appear in the RTP payload, this bit is set to 1. Although extension data can improve video performance, their inclusion in the RTP payload is optional.

I: Interleaving mode flag bit (1 bit). Indicates that after the current payload header and payload media data in the same RTP packet, another payload header and media data will follow.

RSV: Reserved (6 bits). Reserved for future extended applications.

In this embodiment, through parameters such as AN, N, S, B, E and PD, PS, TF, R, G, PHD, T, I, PR, etc., the RTP packet content can be fully understood on the basis of effective transmission And help the receiving end to decode. Through the combination of the assigned values of parameters S, PHD, B, E, T and I, the video data is encapsulated into packets of appropriate size in real time to improve transmission efficiency. For example, when these parameters are all set to 1, a video sequence and some user data may be encapsulated in one RTP packet. Through the combination of S, PD and I parameters, multiple video data units can be encapsulated in one RTP packet, and the payload information before each video data unit indicates the decoding time of the code stream in PD. For example, the RTP packet is encapsulated in the following form: "RTP header + sequence header 1 + sequence 1 + code stream 1 + sequence 2 + code stream 2 + sequence header 2 + sequence 3 + code stream 3". In this way, multi-source video data can be encapsulated in one RTP packet to perform interleaved packet switching and video source switching. In this way, the error concealment capability of transmission and the load balancing capability of the server are improved. Repeated transmission is realized through TF, R, and G parameters, thereby improving the ability of correct decoding.

Since AVS-P2 is a moving image compression technology, the time dependence between images is relatively strong. When an error occurs in the previous image, it may affect the decoding of subsequent images. Usually the B frame does not participate in the prediction, so its damage will not cause error diffusion, but the I frame and P frame are different, their damage will cause error diffusion, thus affecting the playback quality. When watching a video, subjective feeling is a very important indicator. In an image, the audience's interest points for each part are different. In a sequence of images, some images have a high degree of similarity, and their damage or loss has relatively little impact on subjective experience, while some images are scene transition points or points of interest of the audience, which are more important. Therefore, when grouping, according to the judgment of the grouping controller on the content, a parameter can be used to mark the importance of the grouping. For packets with a relatively high level of importance, the transmission system sends the transmission first, or repeats the transmission to the receiving end within a certain time range.

Therefore, the priority flag bit PR (2 bits) can be set in the RTP packet according to the importance of the content and the point of interest, so as to adapt to the network condition and balance the load of the receiving end. For example, when level 00 is the highest priority, level 11 is the lowest priority. When network resources are tight, the parameters and video data that meet the basic transmission bandwidth (higher priority data) will be transmitted first; in the network queue or when the receiving end is overloaded, try to process high priority data first and discard low priority data level data. At the receiving end 12, after the receiving buffer 121 receives the packets marked with a high importance level, they will be given priority for decoding and playing. When the delay exceeds the buffer size, the packets marked as lower priority can be discarded to ensure real-time decoding and playback of high priority packets.

Of course, besides the above-mentioned forms, other naming forms may also be used for the naming of each parameter in the above grouping format.

The video code stream packet transmission method of the present invention is described below by taking an AVS-P2 code stream as an example.

As shown in FIG. 6 , it is a flow chart of the video code stream packet transmission method of the present invention.

First, at the service end 11, according to the service request of the receiving end 12 (i.e., the request for playing video data), use the encoder 112 to encode the specified video file into an AVS-P2 code stream, or directly from the stored AVS-P2 code stream. The stream storage device takes out the AVS-P2 code stream, and encapsulates the above code stream into RTP packets according to the "AVS-P2RTP Payload Format" according to the network conditions. Then, the server 11 transmits the generated RTP packet to the receiving end 12 through the network (step S61).

After receiving the RTP packet, the receiver 12 decodes the received RTP packet into video data, and plays the video data through the player (step S62).

As shown in FIG. 7 , it is a flow chart of the steps of encapsulating the AVS-P2 code stream in FIG. 6 .

At first service end 11 according to the service request from receiving end 12, the video file of request is encoded as AVS-P2 code stream; Or directly take out AVS-P2 code stream from storage device, and code stream is put into packing buffer (step S71).

The server 11 also obtains the real-time status of the network 13 through the application controller 111, specifically including one or more of network bandwidth, code rate, and packet loss rate (step S72).

Then, the server 11 uses the packetizer 113 to encapsulate the AVS-P2 code stream in the packet buffer into RTP packets with different sizes according to the RTP payload format according to the network conditions monitored by the application controller 111 . For example, when the application controller 111 detects that the network bandwidth is insufficient, each RTP packet generated by the packer 113 contains a small amount of data, and when the network bandwidth is large, each RTP packet generated by the packer 113 contains a large amount of data. After the encapsulation is completed, the server 11 sends the encapsulated RTP packet to the receiving end 12 through the network 13 . The structure of the RTP packet is shown in Fig. 2, Fig. 3 and Fig. 5 (step S73). In addition, the correct decoding capability of the receiving end 12 can also be improved by repeated transmission.

Finally, the server 11 transmits the generated RTP packet to the receiving end 12 through the network 13 (step S74).

As shown in FIG. 8 , it is a flowchart of the steps of receiving and decoding the RTP packet in FIG. 6 .

First, the receiving end 12 puts the RTP packet from the server 11 into the receiving buffer 121, and waits for a predetermined time. The predetermined time avoids the playback pause caused by network delay and packet loss, etc., and ensures the continuity of the playback (direct playback may cause the decoding to be suspended because the subsequent RTP packet is not received), and its value can be determined according to the network. The situation is preset, for example, when the network bandwidth is small, the waiting time is longer, and when the network bandwidth is large, the waiting time is short (step S81).

Then send the RTP packet in the receiving buffer 121 to the decoder 122 for decoding, so as to obtain video data, and use the player 123 to play (step S82).

The above is only a preferred embodiment of the present invention, but the scope of protection of the present invention is not limited thereto. Any person skilled in the art within the technical scope disclosed in the present invention can easily think of changes or Replacement should be covered within the protection scope of the present invention. Therefore, the protection scope of the present invention should be determined by the protection scope of the claims.

Claims (16)

1, a kind of video-code-flow grouped transmission method is characterized in that, may further comprise the steps:

(a) in service end the code stream that encoder generates is encapsulated as the RTP grouping according to network condition, and described RTP grouping is sent to receiving terminal;

(b) described receiving terminal obtains video data with the RTP packet decoding that receives.

2, video-code-flow grouped transmission method according to claim 1 is characterized in that, described step (a) comprising:

(a1) insert the packing buffering area that is positioned at service end in service end according to the code stream of the service request own coding device in future of receiving terminal;

(a2) pass through application controller monitoring network situation in service end, and the code stream that uses packing device according to described network condition encoder to be generated is encapsulated as the RTP grouping;

(a3) described RTP grouping is sent to described receiving terminal.

3, video-code-flow grouped transmission method according to claim 1 is characterized in that, described step (b) comprising:

(b1) at described receiving terminal the RTP grouping that receives is inserted the reception buffering area that is positioned at receiving terminal and waited for the fixed time;

(b2) grouping of the RTP in the described reception buffering area is sent to decoder decode.

4, video-code-flow grouped transmission method according to claim 1, it is characterized in that, comprise in the described step (a) in RTP grouping, adding the important level parameter and transmitting the step of RTP grouping that described step (b) includes according to preferentially the decode step of the high RTP grouping of important level of the important level parameter in the RTP grouping according to important level.

5, video-code-flow grouped transmission method according to claim 1 is characterized in that, described code stream is the AVS-P2 code stream, includes the grammer layer element and the media data of RTP header file, AVS-P2 code stream in the described RTP grouping.

6, according to each described video-code-flow grouped transmission method among the claim 1-5, it is characterized in that, described network condition comprises one or more in the network bandwidth, code check and the packet loss, in described step (a), the size of the RTP grouping that service end generated increases along with the increase of the network bandwidth, increases along with the increase of code check, reduces along with the increase of packet loss.

7, video-code-flow grouped transmission method according to claim 1 is characterized in that, described RTP grouping comprises that RTP head and RTP carry only, and wherein RTP included RTP in clean year and carries header and media data only.

8, video-code-flow grouped transmission method according to claim 7, it is characterized in that, also comprise the step that improves the ability that is correctly decoded by repeating to transmit, described RTP carries header only and includes first precedence designation, important first sign and progressive frame flags parameters.

9, video-code-flow grouped transmission method according to claim 1 is characterized in that, comprises in the described step (a) that use transmission code time information and many video data packets encapsulate the step of interlock grouping and video source switching.

10, a kind of video code flow block transmission system, comprise the service end and the receiving terminal that are connected by network, it is characterized in that, the packing device that described service end includes provides the code stream of code stream source, be used for the application controller of monitoring network situation and will be encapsulated as the RTP grouping from the code stream in code stream source, described receiving terminal comprises the decoder of decoding from the described RTP grouping of service end, and wherein said packing device is encapsulated as described code stream according to the result of the monitoring network of application controller the RTP grouping of specific size.

11, video code flow block transmission system according to claim 10 is characterized in that, described code stream source is for being encoded to video file the encoder or the encoder group of code stream, perhaps for storing the storage device of code stream.

12, video code flow block transmission system according to claim 10, it is characterized in that, described receiving terminal includes the reception buffering area of buffer memory from the RTP grouping of service end, and described reception buffering area sends to described decoder decode after the RTP grouping cache scheduled time with reception.

13, video code flow block transmission system according to claim 10 is characterized in that, described RTP grouping comprises that RTP head and RTP carry only, and wherein RTP included RTP in clean year and carries header and media data only.

14, video code flow block transmission system according to claim 10 is characterized in that, includes the parameter of sign RTP grouping important level in the described RTP grouping, the high RTP grouping of important level of preferentially decoding of described encoder.

15, video code flow block transmission system according to claim 10 is characterized in that, described code stream is the AVS-P2 code stream, includes the grammer layer element and the media data of RTP header file, AVS-P2 code stream in the described RTP grouping.

16, according to each described video code flow block transmission system among the claim 10-15, it is characterized in that, comprise the parameter that the sign growth data occurs in the described RTP grouping, described network condition comprises one or more in the network bandwidth, code check and the packet loss, and the size of the RTP grouping that service end generated increases along with the increase of the network bandwidth, increases along with the increase of code check, reduces along with the increase of packet loss.

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