TWI475842B - Real-time control method of servo-to-client data stream transfer rate - Google Patents

Description

Instant control method for server-to-client data stream transmission rate

The invention relates to the transmission of multimedia materials, in particular to a method and system for real-time transmission of continuous data in a regional network/wide area network environment.

This application is a continuation application of U.S. Application No. 11/172,489, entitled "Adaptive Media Transmission Management for Continuous Media Streaming in a Network/WAN Environment", which was filed on June 29, 2005. An application was filed by Huang, which requested the right of US Provisional Application No. 60/583,360, filed on June 29, 2004, and the case was incorporated into the case.

With the emergence of new application methods and new usage models, video streaming in the regional/wide area network environment has gained significant importance. Consumers can enjoy watching video streams and/or listening to audio streams from desktop or laptop computers, creating a new paradigm from the consumption of traditional media. Not only in the personal digital video entertainment environment, video streaming has been accepted in the corporate environment with the increasing popularity of video conferencing and distance learning.

Unfortunately, the popularity of video streaming has placed a significant burden on the available bandwidth of existing network infrastructure. In various web-based applications, meaningful data can be transmitted or received in a burst mode. For example, a user using a web browser can invoke a URL on a URL (single resource locator) link to invoking a web address to update the user's web page. Network communication between the web browser and the desired web address may continue until the web page update is complete or the user selects another URL to refresh their browser. However, unlike previous web-based applications, streaming video and audio data must be delivered to a client (viewer) in a continuous stream after the connection between the client and the server is established. If the number of streams is not maintained above a certain level (or not equal to a minimum amount of transmission), the video may be stopped because the decoder does not have enough data to process (decode). However, the video will respond immediately when the data is available again.

In the past, video broadcasting was to transmit video information in the analog domain. The streaming video information was different from the previous video broadcasting. It had to be converted into a digital format, so it became an IP packet. A communication protocol (IP) network [such as a home network (regional network) or an internet (wide area network)]. However, the simple conversion of video from an analog domain to a digital domain does not immediately allow users to transmit video over an IP network because the required video bandwidth is greater than the bandwidth allowed by most existing networks. For example, (according to the National Television Standards Committee) at a standard rate of 30 frames per second, and a standard home TV with 720x480 pixels, each pixel requires 2 bytes for color projection, to transmit a full screen. For video without any data compression, the minimum throughput will be around 19.78MBps or 158Mbps. This condition obviously exceeds the range supported by the capacity of a typical local area network (10/100 Mbps). Therefore, it is not practical to transmit the original video length through the conventional network architecture; or, at least, it is difficult to use the existing technology for live video transmission on the network. Therefore, compressing video data before transmitting data on an IP network becomes a necessary condition.

A variety of video compression technologies have been introduced and are constantly being improved. The most notable compression techniques are MPEG, H.261, H.263, and H.264. However, even with these compression engines, a constant amount of throughput is required to transfer mobile video from the server to a display client. Unfortunately, the WAN/WAN is designed as a “best effort” medium, meaning that although it attempts to get data from the source (server) to the receiver (client) in the lowest possible time range, it does not. This must be guaranteed.

If the minimum amount of transmission is not met, video jitter may occur during playback. In order to find out the existing video jitter in mobile video, different methods are widely used in various online video (VOD) applications, such as configuring buffers on both the client and the server. The content provider (equivalent to the server) in a typical online video application can retransmit the video data when the display client requests that the packet is lost or transferred during transmission. This technology removes video jitter and enhances viewing experience. However, while this approach achieves the goal of online video, it can cause significant delays in applications that require instant viewing. In some instant applications, such as live video broadcast, security monitoring, and video conferencing, it is time sensitive. In this case, the server can only transmit video data once.

In order to overcome the "best effort" nature of the network while managing traffic on the network in a more efficient and predictable manner, a number of different quality of service (QoS) processing methods have been developed. For example, one method is to place a priority flag in the header of the information stream to allow a mobile application to receive a higher level of service from the router and other applications. This method is represented by DiffServ. Another approach is to allow the mobile application to broadcast its service requirements to the network via a reservation request. This method is represented by IntServ. In general, according to the mobile application of a QoS-awared router, the Resource Reservation Setting Protocol (RSVP) can be used as its end-to-end signal protocol. Unfortunately, none of these methods guarantees that each transfer point between the server and the client has a device that supports or understands these protocols, so it may not be possible to achieve the continuous stream elements required for mission-critical applications.

Therefore, we need a system that can transmit data in real time across traditional networks, and at the same time meet minimum thresholds during data transmission to maintain human visually acceptable video quality.

The present invention provides a variety of methods and systems for instantaneously transmitting compressed video data streams across conventional networks or channels. Each system uses Adaptive Rate Control (ARC) technology. This technology is based on the detection of 瑕疵 packets and other influencing factors such as overall system performance, usage patterns, and channel characteristics. In order to control the flow choking in each channel and maintain the visual display quality above a certain level, the present invention further utilizes human vision and human adaptability to changing visual conditions.

The aforementioned adaptive rate control (ARC) technology relies on the client's ability to calculate the video packet quality transmitted by the server and the ability to transmit the required transmission rate information to the server. This method simplifies the hardware and software implementation complexity of the previous server or content provider, and shifts the monitoring bandwidth and transmission control burden from the server to the client, thereby reducing the overall design cost.

One of the objects of the present invention is to provide an instant control method for a server-to-client data stream transmission rate, wherein the client has a comparison check table generated using a set of characteristics thereof. The method comprises the steps of: receiving a first portion of a data stream from a server at a transmission rate, wherein the first portion comprises a majority data packet; checking the integrity of the data packet; determining a packet based on its integrity a leakage ratio; using the packet leakage ratio and the comparison checklist to generate a request for adjusting the transmission rate; transmitting the request to the server; and receiving the data stream from the server at the adjusted transmission rate Two parts.

Another object of the present invention is to provide an instant control method for a server-to-client data stream transmission rate, wherein the client has a comparison check table generated using a set of features thereof. The method includes the steps of: receiving a request from a client to adjust a transmission rate, wherein the client generates the request by using a packet loss ratio and the comparison checklist; receiving at least one audio input and at least one video input; Requires updating a data rate and the transmission rate; encoding the audio input and the video input at an updated data rate; composing the encoded audio and video input into the data stream; and transmitting the data stream to the client end.

Still another object of the present invention is to provide an instant control method for a server-to-client data stream transmission rate, wherein the client has a comparison check table generated using a set of features thereof. The method includes the steps of: causing the server to transmit a first portion of a data stream to the client at a transmission rate; causing the client to receive the first portion, wherein the first portion includes a majority data packet Causing the client to check the integrity of the data packet and determine a packet loss ratio based on its integrity; causing the client to use the packet loss ratio and the comparison checklist to generate a request to adjust the transmission rate, and Transmitting the request to the server; causing the server to receive the request and adjusting the transmission rate according to the requirement; and causing the server to transmit the second portion of the data stream at the adjusted transmission rate.

The above and other features, objects and advantages of the present invention will become more apparent from the <RTIgt;

The present invention is described in the following detailed description of the preferred embodiments of the invention.

It must be noted that the singular forms "a", "and" and "the" are used in the <RTIgt; Thus, for example, reference to "a client" includes the equivalent meaning of one or more clients and those skilled in the art.

Broadly speaking, the present invention provides a variety of methods and systems for transmitting compressed video data streams across most conventional networks or channels in an instant environment. Adaptive Rate Control (ARC) technology can be used for each system. This technology is based on the detection of 瑕疵 packets and other influencing factors such as overall system performance, usage patterns, and channel characteristics. The present invention is different from the current occlusion control mechanism based only on the channel characteristics or the multi-layer flow research method, and can effectively utilize the human visual and human adaptability to the changing visual state to maintain the visual display quality at an acceptable level. .

The aforementioned adaptive rate control (ARC) technology can rely on the computing power of the client to transmit the video packet quality of the server, and the ability to transmit the required transmission rate information to the server. This method simplifies the hardware/software implementation tool complexity that has been undertaken by the previous server or content provider, and shifts the monitoring bandwidth and transmission control burden from the server to the client, thereby reducing the overall design cost.

Please refer to Figure 1. 1 is a schematic diagram of a real-time video and audio data transmission system 100 in accordance with the present invention, the system including a local area network (LAN) system 101 and a wide area network (WAN) system 121. As shown, the local area network 101 refers to the network system on the left side of the air 120, which may include a network cable or WiFi channel 130 (hereinafter referred to as a cable), and most of the personal computers 114, 116 connected to the cable 130. A server 106 coupled to the cable 130, a router 112 coupled to the cable 130, and a laptop 110 configured to communicate with the router 112. The local area network system 101 may also include most video display devices such as a webcam 108, a player 104, and a television set 102. The audio-visual display device can be coupled to the server 106 and exchange audio and video data with the server (wired or wireless) 106 in real time. The regional network system 101 can communicate with other regional or wide area network systems over the air 120. The so-called air, preferably refers to the Internet. A firewall 118 can be included in the local area network system 101 to prevent malicious data/programs such as computer viruses and spyware from entering the system.

Wide area network refers to the system on the right side of the air 120, which may include a set of client or client devices and an access (access point) 122, commonly referred to as a hot spot, for providing wireless communication to the client. Among them, the client includes a personal computer 128 and other handheld devices, such as a personal digital assistant (PDA) 124 and a mobile phone 126. Clients 124, 126, 128 can communicate with the server over the air 120. The server can be a media server or a content provider for providing mobile data to the client immediately. References below to the term client or client device refer to a device that is configured to communicate with the server 106 and to exchange mobile data with the server 106 in real time. A client may have one or more audiovisual display systems for displaying streaming data. The details of the data flow between the server and one of the clients are further described below with reference to Figures 2-4. In FIG. 1, only nine clients 102, 104, 108, 110, 114, 116, 124, 126, and 128 are shown for convenience. However, those with general skills should be able to understand that other suitable types of client devices can be used in system 100.

In block diagram 200 of FIG. 2, an instant mode data stream between the server and one of the plurality of client devices is shown in FIG. As shown, a server 202 can transmit composite audio and video output data to a client 212 via a network (area or wide area network) 210. The following will use the words "multiplex" and "interleave" interactively. The composite video output data may include video data compressed at a predetermined data rate (compression rate) to reduce flow congestion in the network 210 and formatted according to a user datagram protocol (UDP). The method of determining the data rate will be described in detail later.

When the client 212 receives the composite video material, the data can be temporarily stored in the recipient buffer 214. The recipient buffer 214 can then forward the data to a video composite decomposition system 218. The audio-visual composite decomposition system 218 can decompose the composite data into two sets of individual data, wherein each set of data includes a video signal or an audio signal; check the integrity of the two sets of data, and decode the two sets of data. As will be explained later, the server 202 will multiply the audio stream and the video stream into a single data stream before transmitting the data to the client 212. As mentioned above, each video frame may require a considerable amount of data. Therefore, each frame of data may be divided into a plurality of data packets before being transmitted through the network 210. Each packet can have a header, which records information such as the total number of packets in the frame, the number of intra-packets, the sum of the frames (the parity), and the length of the packet.

The video composite decomposition system 218 has a video ring buffer 244 and an audio ring buffer 242 (see FIG. 4) for verifying the integrity of the video data packet and the audio data packet, respectively. The video composite decomposition system 218 will be described in further detail below with reference to FIG. When the video ring buffer 244 encounters a switched or lost packet (collectively referred to as a packet), it discards the frame containing the packet and transmits a notification to the packet error accumulator 222. In another embodiment, if there is a packet that is switched, the error correction method such as Forward Error Corrections can be used to reconstruct the frame. In another embodiment, if there is missing data (missing packets), the frame can be processed using a restoration method such as a Reversible Variable Length code. The audio/video composite decomposition system 218 can transmit the decoded audio and video data to a speaker/speaker 220a and a video display 220b, respectively.

The packet error accumulator 222 can periodically determine and transmit a packet miss ratio to a transmission LUT 224. The packet leakage ratio is defined as: the ratio of the packet count to the total number of packets received in a time interval. At the beginning of the time period, the packet error accumulator 222 can reset the packet count. The packet error accumulator 222 may increment the packet count by one when it receives the packet notification from the video composite decomposition system 218. At the same time, it counts the total number of packets received during that time period. At the end of the time period, the packet error accumulator 222 can calculate the packet leakage ratio and transmit the ratio to the transmission comparison check table 224.

The client 212 can also transmit its feature information to the smart video transmission manager 208 when communication between the server 202 and the client 212 begins. Client feature information may include display size, highest resolution, number of supportable color bits (or color depth), display type (such as LCD, CRT, or plasma type), CPU type, indication supportable format type (such as MPEG1) , MPEG2, MPEG4, H.263, H.264, etc.) decoder capabilities, available memory, connection type (such as WiFi, LAN, Bluetooth, etc.), description of previously connected information (such as frames) Rate, data rate, quantization level, resolution, etc.), and the line structure of the video display 220b (continuous/interleaved). Then, based on these feature information, the server 202 can determine the initial transmission rate, which minimizes latency in data transmission and maximizes the quality of received data.

The transmission alignment check table 224 can use the packet leakage ratio to read a comparison checklist to determine whether the current transmission rate needs to be adjusted. The alignment checklist can be customer specific and generated using the characteristics of the client 212. Then, the transmission comparison check table 224 can transmit the current transmission rate requirement (such as the "control command" indicated in FIG. 2) to the smart video transmission manager 208 via the network 210. Details of the adjustment will be further explained below with reference to FIG.

Using the requirements and feature information transmitted by the client 212, the smart video transmission manager 208 can adjust a set of parameters including target frame rate, target display bit rate, target resolution, target audio sampling rate, quantized value, and adjust the brightness color. Degree (YUV) conversion method (4:2:2 to 4:2:0), scale size. The manager 208 can also adjust the data compression rate (referred to as the data rate) and transmit the adjusted data rate information to an audio/video composite system 204. The audio/video composite system 204 can receive one or more audio and video input streams 205, encode the received input stream, compress the input stream using the data rate information received from the smart video transmission manager 208, and divide each frame of data into a plurality of packets, and These packets are passed to a Media Access Control (MAC) buffer 206. As previously mentioned, each video frame can include a substantial amount of data and can be divided into a plurality of data packets for easy transmission over the network 210. More detailed information about the audio/video composite system 204 will be provided below with reference to FIG. The Media Access Control (MAC) buffer 206 may temporarily contain packets to be transmitted over the network 210, depending on the number of clients, in mono unicast, stereo two-channel FM, and Or broadcast (broadcast) mode to transmit these packets.

FIG. 3 is a block diagram of the AV composite system 204 implemented in the servo end 202 of FIG. As shown, the audio/video composite system 204 can include an audio encoder 230 for encoding an audio input stream, and an audio ring buffer 232 for relaying the encoded audio input stream to an audio/video combiner 238; The encoder 234 can encode (or compress) a video input stream according to the data rate received from the smart video transmission manager 208; a video ring buffer 236 can relay the encoded video stream to the audio/video combiner 238; The video and audio synthesizer 238 can combine the encoded audio stream and the video stream into a single composite data stream. To maintain audio/video synchronization and sample clock integrity, the video and audio synthesizer 238 can interleave the encoded audio stream with the encoded video stream.

As mentioned earlier, a composite stream packet transmitted by a server may be lost or transferred during transmission through the network due to network congestion, decoder load saturation, or system load saturation. Therefore, the integrity of the composite stream packet received through the network must be checked after the composite stream packet is decomposed. For example, FIG. 4 shows a block diagram of the video composite decomposition system 218 of FIG. 2, wherein the video composite decomposition system 218 can decompose the received composite stream into an audio and a video stream; check the integrity of the video stream; These audio streams and video streams are decoded for playback by the speaker/speaker 220a and the video display 220b. As shown, the video composite resolver 240 can decouple the received data stream into an audio stream and a video stream. It then places the video stream into video ring buffer 244 and the audio stream into an audio ring buffer 242. The audio ring buffer 242 and the video ring buffer 244 can check the video sync and frame sync, respectively, and perform the necessary corrections before decoding. The video ring buffer 244 can also check for any transitions and/or missed packets to verify the integrity of the video stream (or packet). When the video ring buffer 244 encounters a packet, it can transmit a notification to the packet error accumulator 222.

As previously explained, the packet error accumulator 222 can determine and transmit a packet miss ratio to the transmission alignment check table 224. The transmission comparison check table 224 can generate a transmission rate requirement according to the packet loss ratio, and transmits the request (or control command) to the server 202 through a User Datagram Protocol (UDP) channel. The UDP command port/channel is an arbitrary port that is specifically designated and reserved by the server 202 for communication with the client 212. Subsequently, the server 202 can adjust the transmission rate by controlling the compression ratio, frame rate, group of pictures (GOP) size, quantization level, display resolution, or any combination of the above.

The packet loss ratio can be used as an indicator of the state of transmission in the network 210. The transmission comparison check table 224 can use the packet missing ratio to read a comparison check list. Wherein, the comparison checklist can be customer-specific and generated using a set of features. These features include display size, highest resolution, number of supportable color bits (or color depth), display type (such as LCD, CRT, or plasma type), CPU type, indication supportable format types (such as MPEG1, MPEG2) Decoder capabilities of MPEG4, H.263, H.264, etc., available memory, connection type (such as WiFi, LAN, Bluetooth, etc.), description of previously connected information (such as frame rate, data) Rate, quantization level, resolution, etc.), and the line structure of the video display 220b (continuous/interleaved).

The transmission state can be subdivided into three sub-states: (1) no load state (2) load state (3) congestion state. Each substate can have a predetermined range of packet leakage ratios. For example, suppose the current transmission state is in a no-load state where the packet leakage ratio is less than 5% (indicating that the maximum leakage loss at a transmission speed of 30 frames per second (30 fps) is 1.5 fps). When the packet leakage ratio is less than 5%, the transmission comparison check table 224 can determine that the current transmission state is in an unloaded state, and can achieve a higher data rate from the client feature side. The transmission alignment check table 224 may require the servo terminal 202 to incrementally increase the transmission rate until the packet leakage ratio exceeds 30%, that is, until the transmission state reaches the congestion state. Then, the transmission comparison check table 224 may require the servo terminal 202 to lower the transmission rate until the load state is reached (the packet leakage ratio reaches a steady state). If the packet loss ratio indicates that the transmission status has reached a steady state, the transmission comparison check table 224 can transmit a request to the smart video transmission manager 208, including a NOP signal (no operation signal or "no action" signal, That is, the "Do not change the data rate" signal). Since the traffic profile on the network can change over time, the client 212 will randomly transmit the request for increased data rate to the server 202.

The threshold level of the packet loss ratio can be carefully adjusted so that the display quality perceived by the user of the client 212 can be moderately reduced or improved as the network conditions change. Since the client 212 has no other available bandwidth traffic information except its current transmission rate, the client 212 may periodically request the server 202 to increase the transmission rate so that higher video quality can be achieved. In addition to monitoring the incoming video packet stream quality, the client 212 can also notify the server 202 whether the display size has been reduced by the customer, thereby reducing the transmission data rate to match the new display size and reducing the bandwidth requirements.

It is worth noting that the current method for controlling the transmission rate, called adaptive rate control (ARC) technology, can be based on lossy packet detection instead of buffering with the help of the Instant Transfer Protocol (RTP). The based approach. Because the adaptive rate control (ARC) technology can mainly rely on the client that has the ability to calculate the quality of the video stream packet and can transmit the required transmission rate information to the server, the burden of monitoring the bandwidth and data rate control can be shifted from the servo end. Go to the client. In addition, Adaptive Rate Control (ARC) technology allows the client to require the server to transmit data streams at an optimized transmission rate, reducing the possibility of traffic congestion in the network.

Figure 5 is a flow chart 500 of a typical procedure for data rate control of the present invention. It will be appreciated by those skilled in the art that the methods illustrated in the figures can be modified in various ways without departing from the spirit and scope of the invention. For example, in another embodiment of the present invention, various different portions may be combined, the different portions may be rearranged in another order, or the different portions may be removed. It should also be noted that this method can be implemented in a number of different ways, such as executing software in a general purpose computer, executing firmware and/or computer readable media using a microprocessor, using specialized hardware, and the like.

The method can begin with a state 502. In state 502, a client can establish a connection or channel with a server via a network (area/wide area network) for data exchange. Then, in state 504, the client can transmit a set of client feature information to the server at the beginning of channel establishment. Client feature information may include display size, highest resolution, number of supportable color bits (or color depth), display type (such as LCD, CRT, or plasma type), CPU type, indication supportable format type (such as MPEG1) , MPEG2, MPEG4, H.263, H.264, etc.) decoder capabilities, available memory, connection type (such as WiFi, LAN, Bluetooth, etc.), description of previously connected information (such as frames) Rate, data rate, quantization level, resolution, etc.), and the line structure of the video display 220b (continuous/interleaved). Then, in state 506, the server can determine a transmission rate based on the client characteristic information, and transmit the composite data stream to the client using the transmission rate in an immediate mode. The composite material may include audio and video data in the form of data packets. The program can then enter state 508.

In state 508, the client can receive the composite stream from the server. Second, in state 510, the client can decompose the received composite data stream into audio and video data packets. Then, in state 512, the client can determine the packet loss ratio of the video data packet according to the number of data packets. The program can then proceed to a decision diamond 514.

In state 514, the client may determine whether the packet miss rate ratio indicates a congestion status in the network. When the answer to state 514 is affirmative, in state 516, the client may generate a request to reduce the transmission rate. The program can then proceed to state 522. Otherwise, the program can proceed to another decision diamond 518.

In state 518, it may be determined whether the packet miss rate ratio indicates an unloaded state in the network. If the reply to state 518 is affirmative, the program can proceed to state 519. In state 519, it may be determined whether the transmission rate has reached the maximum transmission rate. If the reply to state 519 is negative, in state 520, the client may generate a request to increase the transmission rate. The program can then proceed to state 522.

When the reply to state 518 is negative, or if the reply to state 519 is positive, then the program proceeds to state 521. In state 521, the client can generate a signal containing a NOP (Do Not Change Data Rate). The program can then proceed to state 522. In state 522, the client can transmit the request to the server. Subsequently, in state 524, the server can transmit the composite data stream with the adjusted transmission rate; wherein the adjusted transmission rate is determined according to requirements. The program can then wait until the transfer is complete to proceed to state 508. It is worth noting that steps 508 through 524 should be implemented periodically to maintain the channel established between the server and the client.

It is to be understood that the foregoing is a description of the exemplary embodiments of the invention, and various modifications may be made without departing from the spirit and scope of the present invention.

100. . . Instant video data transmission system

101. . . Regional network system

102. . . TV set

104. . . Sounder

106. . . Servo end

108. . . Webcam

110. . . Laptop

112. . . router

114. . . personal computer

116. . . personal computer

118. . . Firewall

120. . . Air (internet)

121. . . WAN system

122. . . Access point (hot spot)

124. . . Personal digital assistant

126. . . Mobile phone

128. . . personal computer

130. . . Network cable or WiFi channel

200. . . Instant mode data flow block diagram between server and client

202. . . Servo end

204. . . Audio-visual composite system

205. . . Video input stream

206. . . Media access control buffer

208. . . Smart Video Transfer Manager

210. . . Network (region/wide area)

212. . . Client

214. . . Receiving buffer

218. . . Audio-visual composite decomposition system

220a. . . Speaker/speaker

220b. . . Video display

222. . . Packet error accumulator

224. . . Transmission comparison checklist

230. . . Audio encoder

232. . . Audio ring buffer

234. . . Video encoder

236. . . Video ring buffer

238. . . Audio and video composite

240. . . Video composite resolver

242. . . Audio ring buffer

244. . . Video ring buffer

500. . . Flow chart of typical steps of data transmission rate control

502-524. . . Data transfer rate control steps (status)

1 is a schematic diagram of a real-time video and audio data stream transmission system according to the present invention; FIG. 2 is a block diagram showing the data flow between the server and the client of FIG. 1; FIG. 3 is implemented in the server shown in FIG. Block diagram of video and audio composite system; FIG. 4 is a block diagram of a video/audio composite decomposition system implemented in the client shown in FIG. 2; and FIG. 5 is a flow chart of typical steps of data transmission rate control according to the present invention.

100. . . Instant video data transmission system

101. . . Regional network system

102. . . TV set

104. . . Sounder

106. . . Servo end

108. . . Webcam

110. . . Laptop

112. . . router

114. . . personal computer

116. . . personal computer

118. . . Firewall

120. . . Air (internet)

121. . . WAN system

122. . . Access point (hot spot)

124. . . Personal digital assistant

126. . . Mobile phone

128. . . personal computer

130. . . Network cable or WiFi channel

Claims (20)

An instant control method for data rate of a server-to-client data stream, wherein the client generates a set of comparison checklists with dynamic client characteristic variables, and uses a packet loss ratio of a dynamically updated data as a comparison checklist. Soliciting access to the comparison checklist, the method comprising: the client receiving a first portion of a data stream from the server at a first transmission rate, the first portion comprising a majority data packet; the client Checking the integrity of the data packet; the client derives a packet loss ratio of the dynamic update data according to the integrity; the client generates a request to the server to adjust the first transmission rate, wherein the client Adjusting the first transmission rate is determined according to the packet loss ratio of the dynamic update data and one or more dynamic variables in the comparison check table of the client, which are matched with the dynamic client feature variables, and the dynamic client feature variables of the group are determined. Will change continuously while the data is transmitted by the server to the client; the client transmits the request to the server; the server is The capable video transmission manager processes the request, and the intelligent video transmission manager adjusts the first transmission rate or a luminance chrominance (YUV) conversion method according to the requirement to convert the higher chroma video content to the comparison. Low video chroma content, and vice versa; and the client receives the second portion of the data stream from the server at an adjusted transmission rate. For example, in the method of claim 1, the first data stream is received Before the step, the method further includes: establishing a connection between the server and the client; causing the client to transmit the value in the one or more client feature variables to the server; and causing the server The first transmission rate is determined using the value of the one or more client feature variables, and the first portion is transmitted to the client at the first transmission rate. The method of claim 2, wherein the set of client feature variables comprises the luminance chrominance (YUV) conversion method, and the higher chroma video content data is converted to lower according to the packet loss ratio of the dynamically updated data. The chroma video content and vice versa. The method of claim 1, wherein the data stream is a composite data stream comprising an audio stream and a video stream; the method, before the step of checking integrity, further comprises: first Partially decomposed into the audio stream and the video stream. For example, the method of claim 4, before the step of checking integrity, further comprises: decoding the audio stream and the video stream. For example, in the method of claim 1, wherein the step of determining the packet loss ratio includes the steps of: measuring the percentage of the data packet in the majority of the data packets; the packet is either a data packet, or The data packet lost in the step of receiving the first part of the data stream. For example, the method of claim 1 of the patent scope, wherein the adjustment request is generated The step includes the following steps: determining whether the packet loss ratio indicates a congestion state; if the packet loss ratio of the dynamic update data indicates a congestion state, generating a requirement for lowering the transmission rate; determining whether the packet loss ratio indicates a no load a state, a maximum transmission rate, and if the packet loss ratio of the dynamic update data does not indicate an unloaded state, generating a request including a NOP (no operation) signal; if the packet loss ratio of the dynamic update data indicates a no load a state, determining whether the first transmission rate has reached a highest transmission rate; if the first transmission rate has not reached a maximum transmission rate, generating a request for increasing the first transmission rate; and if the first transmission rate has been Achieving this highest transmission rate results in a requirement to include a NOP (no operation) signal. The method of claim 1, wherein the data stream is formatted according to a User Data Package Agreement (UDP). The method of claim 1, wherein the data stream is transmitted through a network, the network comprising at least a local area network (LAN) and a wide area network (WAN). The method of claim 1, wherein the client comprises a visual device. The method of claim 1, further comprising: repeating the step of the integrity check to the step of receiving the second portion until the client receives the entire portion of the data stream. For example, in the method of claim 1, the transmission rate has a set of parameters including at least a target frame rate, a target display bit rate, a target resolution, a target audio sampling rate, a quantized value, and a gray color. Degree (YUV) conversion method, a scale size, and a description of the connection state of the previous connection state. A method for instantly controlling a data stream transmission rate, wherein the client generates a comparison check table in combination with a customer feature variable, and utilizes a packet loss ratio of the dynamic update data as a reference for the comparison checklist, the method comprising: receiving a client End request to adjust the transmission rate or adjust the luminance chrominance (YUV) conversion method to convert the higher color video content data in the client into lower color video content data, and vice versa, and wherein the client The packet loss ratio of the dynamic update data is accumulated to form a reference of the comparison check table to access values in one or more comparison check tables to generate a request of the client, and wherein the client adjusts the transmission rate Based on the packet loss ratio of the dynamic data and one or more dynamic variables in the comparison check table of the client, the dynamic client characteristic variables are determined by the server. Changing to the client; receiving at least one audio input and at least one by an audio/video multiplexer system group Inputting; and accepting the client's request from the client to adjust the transmission rate or adjust the luminance chrominance (YUV) conversion method to convert higher color video content to lower video chroma content, And vice versa, and the steps of receiving the client request are finally based on a smart video transmission. The change manager, rather than the client's request to make adjustments; handle the client's requirements in the smart video transfer manager, adjust the transfer rate or adjust the luminance chromaticity (YUV) conversion method to use the higher The color video content data is converted into lower color video content data, and vice versa, and the intelligent video transmission manager processes the client requirements: decoding the audio input and video input according to the client request; according to the client The multiplexed processing of the decoded audio and video input is required to form a data stream; the data stream is transmitted according to the requirements of the client. The method of claim 13, wherein the comparison checklist is generated by the client and includes at least a display size, a maximum resolution, a color depth, a display type, a CPU type, one or more decoder capabilities, and an available memory. One of the body, the type of connection, the previous connection description with the transmission information, and the line structure of the video display. A method for real-time adaptive media data packet continuous data stream transmission management, the method comprising: establishing a connection between the server and the client; the server transmitting the data stream of the multimedia data packet by using a LAN or a WAN The first part is given to the client, wherein the server includes an intelligent video transmission manager, which adjusts the data rate or adjusts the luminance and chrominance (YUV) conversion method according to the packet loss ratio of the data provided by the client, so as to be higher. Converting color video content data into lower color video content data; The client receives the first part of the data stream of the multimedia data packet by the server by using the LAN or the WAN, and wherein the client has a comparison check table, where the comparison check table includes detecting the second part of the multimedia data packet. Adjusted to the requirement of the luminance chrominance (YUV) conversion method, the second part is to be transmitted to the client, and the client is based on the packet loss ratio of a dynamically updated data and the dynamic variable value in one or more comparison checklists Combining a set of dynamic client feature variables that are constantly changing during the continuous transmission of the data packet, the server is independently required to adjust the speed and luminance chrominance (YUV) conversion method; and the LAN or WAN is operatively connected to the server and Client, where the LAN or WAN provides a path for data packet transmission. For example, in the method of claim 15, before the step of transmitting the second part, the method further comprises: an audio/video multiplexer system for receiving and decoding an audio/video receiving data stream, and a MAC buffer To temporarily store the decoded data stream of a multimedia data packet. The method of claim 15, further comprising: a receiving buffer to accumulate the multimedia data packet decoded data stream transmitted by the client, and an audio/video demultiplexer system to decode at least a portion of the multimedia Data decoding data stream. The method of claim 17, wherein the client further comprises a packet error accumulator for calculating the packet loss ratio of the data in real time, and the packet loss ratio of the data is used as a client comparison checklist. The comparison checklist in the reference. The method of claim 15, wherein the client sends a request to the server to adjust a luminance chrominance (YUV) conversion method of the second portion of the multimedia data packet, the second portion being transmitted by the server to Client's. The method of claim 19, wherein the server adjusts a luminance chrominance (YUV) conversion method of the second portion of the multimedia data packet.