WO2021164405A1

WO2021164405A1 - Data encoding and decoding methods, and related device and system

Info

Publication number: WO2021164405A1
Application number: PCT/CN2020/137448
Authority: WO
Inventors: 陈军; 周进华; 李建平; 郭建伟
Original assignee: 华为技术有限公司
Priority date: 2020-02-21
Filing date: 2020-12-18
Publication date: 2021-08-26
Also published as: CN113301387A; CN113301387B

Abstract

Disclosed by the present invention are data encoding and decoding methods, and a related device and system, which are applied to the transmission of low-latency sensitive service data. The methods comprise: at an encoding end, acquiring data to be encoded, and acquiring a delay T, and the maximum number of consecutive lost packets B and the maximum number of burst lost packets N within the delay T; according to the delay T, the maximum number of consecutive lost packets B and the maximum number of burst lost packets N, encoding the data to obtain a code stream; and at a decoding end, acquiring the code stream, and according to the delay T, the maximum number of consecutive lost packets B and the maximum number of burst lost packets N, decoding the code stream so as to obtain data to be decoded. The use of the embodiments of the present invention is advantageous in solving the problems of continuous packet loss and discrete burst packet loss during data transmission.

Description

Data coding and decoding method, related equipment and system

This application claims the priority of a Chinese patent application filed with the State Intellectual Property Office of China, the application number is 202010107486.4, and the title of the invention is "data coding and decoding methods, related equipment and systems" on February 21, 2020, the entire contents of which are incorporated by reference In this application.

Technical field

The present invention relates to the field of data transmission, in particular to a data coding and decoding method, related equipment and system.

Background technique

The future 5G and B5G (Beyond 5G) evolution networks need to meet the business needs of all walks of life, while providing lower latency. In order to deal with real-time interactive applications, the end-to-end latency is less than 150ms; by 2020, wireless communication The data plane delay control needs to be controlled within 4ms; more stringent requirements come from ultra-reliable low-latency communication (uRLLC), one of the three main scenarios of 5G, and its delay requirements on the data plane At the ms level or even lower than 1ms, it is required to ensure that the data transmission has good robustness. For example, the bit error rate (BER) is less than 10-6 or lower, which requires the system not only to be low At the same time, it needs to have better error correction capabilities.

The actual communication systems are also very diverse. Wired systems include optical fibers and copper wires, and wireless systems include cellular and WiFi. Even if the wired network, its bandwidth will be jittered, and the wireless system will have bandwidth fluctuations due to channel fading and co-channel interference. In order to combat the jitter of the wired channel bandwidth and the interference or fading in the wireless channel, various video fault-tolerant technologies are widely used in modern communication systems, such as automatic repeat reQuest (HARQ)/hybrid automatic retransmission based on retransmission Request (hybrid automatic repeat request, HARQ), feedback-based reference frame selection (reference picture selection, RPS) and previous error correction (forward error correction, FEC) solutions, etc. Due to the round-trip transmission delay of the network, the ARQ/RPS solution introduces a longer delay and is not suitable for real-time communication scenarios that have strong requirements on the delay, while FEC has become a relatively effective solution.

For FEC, traditional FEC solutions such as the secure and reliable transport (SRT) system's built-in error correction code based on the "exclusive OR" operation can only correct specific error packets in the actual system, but the system does not meet the requirements of 5G The system has the ability to adapt to the different needs of massive applications and different channels. The design of the Reed-Solomon (RS) code can only guarantee the normal recovery when the packet loss is corrected, and consider the delay requirements of the business, that is, how to ensure that under the premise of the application requirements, such as the delay-sensitive business requires 10ms Or within 5ms, recover the lost data packets within the time window. Therefore, the RS code cannot meet the delay requirements of the services in the 5G and B5G systems. In the existing 5G network, delay-sensitive services (such as AR/VR, video live broadcast services) require the system to correct continuous packet loss and burst discrete packet loss within a specific time window (such as 10ms), traditional system coding Designs such as FEC solutions cannot meet the needs of delay-sensitive services.

Summary of the invention

The embodiments of the present invention provide a data encoding and decoding method, related equipment, and system. The embodiments of the present invention are beneficial to solve the problems of continuous packet loss and discrete burst packet loss in data transmission.

In the first aspect, an embodiment of the present invention provides a data encoding method, including:

Obtain the data to be encoded, and obtain the target delay T and the number of consecutive packet losses B and the number of burst packet losses N within the target delay T. The target delay T is the delay required in the data transmission process; The target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N are encoded to obtain the code stream.

Among them, the number of burst packet loss N is caused by the channel.

Preferably, the target delay T is the maximum delay required in the data transmission process, for example, an application needs to use the data to be encoded, and the target delay T is the maximum delay required by the application. The number of consecutive packet losses B is the maximum number of consecutive packet losses within the target delay T, and the number of burst losses N is the maximum number of burst packet losses within the target delay T.

By encoding the to-be-coded data according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, to obtain the code stream, it is helpful to solve the problems of continuous packet loss and discrete burst packet loss in data transmission. It meets China's demand for time delay during data transmission.

In a feasible embodiment, encoding the to-be-coded data according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a code stream includes:

Determine k and n according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N, where k is the number of codewords before encoding, and n is the number of codewords after encoding; encode the data to be encoded according to k and n to Get the code stream.

In a feasible embodiment, determining k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N includes:

The value range of the channel coding efficiency R is determined according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N; the capacity C is determined according to the channel coding efficiency R, and k and n are determined according to the capacity C. Among them, k/n≤C, C is the maximum value of channel coding efficiency R.

In a feasible embodiment, obtaining the target delay T includes:

Obtain the service type corresponding to the data to be coded, and determine the target delay T according to the service type corresponding to the data to be coded. Different delays are used when transmitting data of different types of services, which meets the delay requirements of different types of services.

Further, after obtaining the target time delay T, the target time delay T is sent to multiple user terminals through broadcast, or is sent to multiple user terminals through unicast.

The target time delay T is sent to the user side, so that the server and the user side use the same time delay T when encoding and decoding, so as to ensure that the user side can decode correctly and overcome the problem of time delay.

In a feasible embodiment, obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target delay T includes:

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the user terminal U, and determine the consecutive packet loss number B according to the multiple historical consecutive packet losses, and according to the multiple The historical burst loss number determines the burst loss number N, and the user terminal U is any one of the user terminals that receive the data to be encoded; or;

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with multiple clients, determine the consecutive packet loss number B according to multiple historical consecutive packet losses, and The number of historical bursts of packet loss determines the number of bursts of packet loss N; multiple user terminals are all user terminals that receive the data to be encoded.

By separately determining the number of consecutive packet losses B and the number of burst packet losses N for each user end, the optimal codec adaptation for each user end is realized during data encoding and decoding, thereby ensuring that each User experience; by simultaneously determining the number of consecutive packet losses B and the number of burst packet losses N for multiple users, the efficiency of optimization statistics is improved, and due to the transmission needs of the number of consecutive packet losses B and the number of burst losses N Occupy bandwidth, thereby improving channel utilization.

Further, after obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the data encoding method further includes:

Unicast the number of consecutive packet losses B and the number of burst packet losses N to the user end U, or; send the number of consecutive packet losses B and the number of burst packet losses N to multiple users via broadcast, or; send the continuous packet loss The number B and the number of burst packet loss N are sent to multiple users through unicast.

The number of consecutive packet losses B and the number of burst packet losses N are sent to the user end, so that the server and the user end use the same B and N when encoding and decoding, so as to ensure that the user end can decode correctly and overcome the continuous packet loss and The problem of sudden packet loss.

In a feasible embodiment, obtaining the target delay T includes:

Receive the reference delay sent by multiple client terminals, the reference delay of the client U is the delay required by the client U; the client U is any one of the multiple client terminals; according to the reference time sent by the multiple client terminals Delay determines the target time delay T.

It should be noted here that the reference delay of the user end U is the maximum time difference between the service transmission time frame and the frame when the application of the user end U is not stuck or screened during the data transmission process.

In a feasible embodiment, after obtaining the target time delay T, the data encoding method further includes:

The target delay T is sent to multiple client terminals through broadcast, or; the target delay T is sent to multiple client terminals through unicast.

The target time delay T is sent to the user side, so that the server and the user side use T during encoding and decoding to be consistent, so as to ensure that the user side can decode correctly and overcome the problem of time delay.

Obtain the reference continuous packet loss number and the reference burst packet loss number sent by multiple client terminals; determine the continuous packet loss number B according to the reference continuous packet loss number sent by multiple client terminals, and determine the continuous packet loss number B according to the reference burst packet loss of multiple client terminals The number determines the burst loss number N;

Among them, the reference number of consecutive packet losses and the reference number of burst packet losses are the maximum number of consecutive packet losses and the number of burst packet losses within the aforementioned reference time delay, respectively.

In a feasible embodiment, after obtaining the maximum continuous packet loss number B and the maximum burst packet loss number N within the target time delay T, the data encoding method further includes:

The number of consecutive packet losses B and the number of burst packet losses N are sent to multiple clients via broadcast, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple clients via unicast.

Optionally, when the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N are sent by broadcast or unicast, they can be sent through a newly established signaling channel with the user end. The signaling channel It is different from the data channel that transmits data; or it is sent through the data channel that transmits data.

In a feasible embodiment, the multiple client terminals are: multiple client terminals using the same transmission method for receiving the data to be encoded, or; multiple client terminals processing the same type of service, or; multiple required quality of service QoS Clients with the same level or QoE experience.

By classifying the clients, different types of clients and servers can obtain the corresponding T, B, and N during data transmission, so that the problems of delay and packet loss can be better overcome.

In the second aspect, an embodiment of the present invention provides a data decoding method, including:

Obtain the code stream, and obtain the target delay T, and obtain the number of consecutive packet loss B and the number of burst packet loss N within the target delay T; the target delay T is the delay required in the data transmission process; according to the target The time delay T, the number of consecutive packet losses B and the number of burst packet losses N are used to decode the code stream to obtain the decoded data.

Preferably, the target delay T is the maximum delay required in the data transmission process, for example, an application needs to use decoded data, and the target delay T is the maximum delay required by the application. The number of consecutive packet losses B is the maximum number of consecutive packet losses within the target delay T, and the number of burst losses N is the maximum number of burst packet losses within the target delay T.

By decoding the code stream according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N, to obtain the decoded data, it is helpful to solve the problems of continuous packet loss and discrete burst packet loss in data transmission. At the same time, it satisfies the demand for delay in the data transmission process.

In a feasible embodiment, the code stream is decoded according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain decoded data, including:

Determine k and n according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N, where k is the number of codewords after decoding, and n is the number of codewords before decoding;

Perform decoding according to the k and n to-be-decoded data to obtain the decoded data.

Determine the value range of the channel coding efficiency R according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N; determine the value range of the channel coding efficiency R according to the channel coding efficiency R, and determine k and n according to the capacity C, where k /n≤C, C is the maximum value of channel coding efficiency R.

In a feasible embodiment, obtaining the target delay T includes:

The delay received from the server is determined as the target delay T, or; the delay required by the user terminal U is determined as the target delay T.

The target time delay T sent by the server is received, so that the server and the user side use the same time delay T when encoding and decoding, so as to ensure that the user side can decode correctly and overcome the problem of time delay.

In a feasible embodiment, obtaining the target delay T includes:

The reference delay is sent to the server, and the reference delay is the delay required by the client; the target delay T sent by the server is received, and the target delay T is obtained by the server according to the reference delays sent by multiple clients.

It should be noted here that the delay required by the user end U can be understood as the time difference between the service transmission time frame and the frame when the application of the user end is not stuck or blurred during the data transmission process.

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the server; determine the consecutive packet loss number B according to multiple historical consecutive packet losses, and according to multiple historical bursts The number of lost packets determines the burst number N of lost packets.

In a feasible embodiment, after determining the number of consecutive packet losses B according to multiple historical consecutive packet losses, and determining the number of burst packet losses N according to multiple historical burst packet losses, the data decoding method further includes:

Send the number of consecutive packet loss B and the number of burst packet loss N to the server.

The number of consecutive packet losses B and the number of burst packet losses N are sent to the server, so that the server and the user end use the same B and N when encoding and decoding, so as to ensure that the user end can decode correctly and overcome the continuous packet loss and sudden loss. The problem of packet loss

Send the reference number of consecutive packet losses and the reference number of burst packet losses to the server. The reference number of consecutive packet losses is the number obtained by the user terminal U based on multiple historical consecutive packet losses, and the reference number of burst losses is the number of the user terminal U based on multiple histories. Obtained by burst packet loss;

Receive the number of consecutive packet losses B and the number of burst packet losses N sent by the server. The number of consecutive packet losses B is obtained by the server based on the number of consecutive packet losses from multiple clients. The number of burst packets Based on the number of lost packets, multiple client terminals include client U.

In a feasible embodiment, the multiple client terminals are:

Multiple user terminals using the same transmission method to obtain the code stream, or; multiple user terminals processing the same type of service, or; multiple user terminals with the same required quality of service QoS level or the same quality of experience QoE.

In a third aspect, an embodiment of the present invention provides a server, including:

The acquisition unit is used to acquire the data to be encoded, and acquire the target delay T and the number of consecutive packet losses B and the number of burst packet losses N within the target delay T. The target delay T is required in the data transmission process Time delay

The encoding unit is used to encode the to-be-coded data according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain a code stream.

In a feasible embodiment, the coding unit is specifically used for:

In a feasible embodiment, in terms of determining k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, the coding unit is used to:

In a feasible embodiment, in terms of acquiring the target delay T, the acquiring unit is specifically configured to:

Obtain the service type corresponding to the data to be coded, and determine the target delay T according to the service type corresponding to the data to be coded.

In a feasible embodiment, in terms of obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the obtaining unit is specifically configured to:

Further, the server also includes:

The sending unit is configured to send the continuous packet loss number B and the burst packet loss number N to the user terminal U through unicast after the acquiring unit acquires the number of consecutive packet losses B and the number of burst packet losses N within the target delay T , Or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple client terminals through broadcast, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple client terminals through unicast.

In a feasible embodiment, the server further includes:

The sending unit is configured to, after obtaining the target delay T, send the target delay T to multiple user terminals by broadcasting, or; unicast the target delay T to multiple user terminals.

Obtain the reference continuous packet loss number and the reference burst packet loss number sent by multiple clients; determine the continuous packet loss number B according to the reference continuous packet loss number sent by multiple clients, and the reference burst sent by multiple clients The number of lost packets determines the burst number of lost packets N;

Among them, the reference number of consecutive packet losses and the reference number of burst packet losses are the maximum number of consecutive packet losses and the maximum number of burst packet losses within the aforementioned reference time delay, respectively.

In a feasible embodiment, the server further includes:

The sending unit is used to send the continuous packet loss number B and the burst packet loss number N to multiple client terminals through broadcast after obtaining the maximum continuous packet loss number B and the maximum burst packet loss number N within the target delay T, Or; both the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple users via unicast.

In a fourth aspect, an embodiment of the present invention provides a user terminal, including:

The acquisition unit is used to acquire the code stream and acquire the target delay T, and acquire the number of consecutive packet losses B and the number of burst packet losses N within the target delay T; the target delay T is required in the data transmission process Time delay

The decoding unit is used to decode the code stream according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain decoded data.

In a feasible embodiment, the decoding unit is specifically configured to:

Determine k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords after decoding, and n is the number of codewords before decoding; decode according to the data to be decoded based on k and n. Get the decoded data.

In a feasible embodiment, in terms of determining k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, the decoding unit is specifically configured to:

The reference delay is sent to the server, and the reference delay is the delay required by the client; the target delay T sent by the server is received, and the target delay T is obtained by the server according to the reference delays of multiple clients. The multiple clients include user terminal.

In a feasible embodiment, the user terminal further includes:

The sending unit is used to send the continuous packet loss number B and the sudden packet loss number B and the sudden packet loss number to the server after determining the continuous packet loss number B according to multiple historical continuous packet loss numbers and determining the burst packet loss number N according to multiple historical burst packet losses The number of lost packets is N.

The reference number of consecutive packet losses and the reference number of burst packet losses are sent to the server. The reference number of consecutive packet losses is the number obtained by the user terminal based on multiple historical consecutive packet losses. The number of packages;

Receive the number of consecutive packet losses B and the number of burst packet losses N sent by the server. The number of consecutive packet losses B is obtained by the server based on the number of consecutive packet losses from multiple clients. The number of burst packet losses N is the server's Obtained with reference to the number of burst packet losses, multiple client terminals include client terminals.

In a feasible embodiment, the multiple client terminals are:

In a fifth aspect, an embodiment of the present invention provides a server, including:

A memory and a processor coupled with the memory, and the processor executes instructions stored in the memory to implement all or part of the content in the first aspect.

In a sixth aspect, an embodiment of the present invention provides a user terminal, including:

These and other aspects of the present invention will be more concise and understandable in the description of the following embodiments.

Description of the drawings

In order to explain the embodiments of the present invention or the technical solutions in the prior art more clearly, the following will briefly introduce the drawings that need to be used in the description of the embodiments or the prior art. Obviously, the drawings in the following description are only These are some embodiments of the present invention. For those of ordinary skill in the art, other drawings can be obtained based on these drawings without creative work.

Figure 1 is a schematic diagram of the principle of the SRT algorithm;

Figure 2 is a schematic diagram of packet loss caused by channel changes;

Figure 3 is a schematic diagram of a block code;

Figure 4a is a schematic diagram of data transmission in a 5G network scenario;

Figure 4b is a schematic diagram of data transmission in a WiFi scenario;

FIG. 5 is a schematic flowchart of a data encoding method provided by an embodiment of this application;

Figure 6a is a schematic diagram of independent signaling interaction provided by an embodiment of the application;

Figure 6b is a schematic diagram of channel associated signaling interaction provided by an embodiment of this application;

FIG. 7 is a schematic diagram of a data format for transmission provided by an embodiment of the application;

FIG. 8 is a schematic flowchart of a data decoding method provided by an embodiment of this application;

FIG. 9 is a schematic structural diagram of a data encoding and decoding system provided by an embodiment of the invention;

FIG. 10 is a schematic structural diagram of a server provided by an embodiment of the present invention;

FIG. 11 is a schematic structural diagram of a user terminal provided by an embodiment of the present invention;

FIG. 12 is a schematic structural diagram of another server provided by an embodiment of the present invention;

FIG. 13 is a schematic structural diagram of another user terminal provided by an embodiment of the present invention.

Detailed ways

The embodiments of the present application will be described below in conjunction with the drawings.

The existing SRT error correction algorithm is based on a simple "exclusive OR" operation. As shown in Figure 1, an "exclusive OR" operation is performed on the original data packet of each row or each column to obtain a check packet. At the receiving end, if a packet is found to be lost, the lost packet can be recovered by checking the packet and performing an "exclusive OR" operation.

However, the drawbacks of this algorithm are obvious, and the robustness is not high, that is, it can only correct one packet lost in each row or column, and cannot cope with complex scenarios. When the existing wireless or wired system transmits packets of different sizes, within a certain time interval, it can cause the loss of multiple consecutive packets or the loss of multiple scattered packets. As shown in Figure 1, the actual system packet loss may be the 0th, 1st, 2nd or the 2nd, 5th packet loss. For such a packet loss mode, SRT's own FEC algorithm cannot correct the lost packets, and the SRT's own FEC algorithm cannot adjust FEC parameters by itself. When dealing with the diversity of mass applications and channels, SRT's own FEC algorithm Unable to deal with it. Finally, the FEC algorithm that comes with SRT does not consider the actual application delay constraints. For some delay-sensitive services, it is required to ensure correct decoding within 5 packet intervals. The encoding method shown in Figure 1 is just The requirement cannot be met.

In the process of data transmission, there are requirements for data stability, which not only requires data to be transmitted correctly, but also requires delay. As shown in Figure 2, for each received data window length T, the window length T can be obtained according to actual applications. The server obtains it according to the feedback of the user's experience, or obtains it through a certain statistical method, and the unit of measurement can be milliseconds (ms) or the duration of the packet length. When transmitting data based on FEC, it is hoped that the data can be restored at a specific time according to the needs of the actual application. At the application layer, the system's packet loss modes are mainly divided into the following two modes:

● There are a maximum of B consecutive lost packets in each receiving time window T.

● N separate packet loss within each delay interval T (also called burst packet loss).

As required by the above scenario, it can be proved that the channel coding rate R that satisfies the time delay T and the above-mentioned required error correction capability satisfies the following conditions:

The upper bound of the above rate R is defined as capacity, expressed as: C _{(T, B, N)} . For C _(T,B,N) , the code that meets the requirements is called the optimal streaming code, namely

Through the above analysis, the block code (n, k, T) that can correct the maximum B consecutive packet loss or N burst packet loss exists when a specific time delay T is met. Now we can meet this requirement by designing this block code Actual demand. The block code is shown in Figure 3.

As shown in Figure 3, assuming that the input information is a code word of k m bits (where the common value of m is 8, 16, etc.), expressed as [s(0),s(1),...,s(k )] ^T , the n codewords after encoding are expressed as: [x(0),x(1),...,x(n-1)] ^T. The generator matrix that defines the size of the block code as n×k is

Among them, I _k is a check matrix with a scale of k×k, and P is a check matrix with a scale of (n×k)×k. The encoded information and the encoded information can be expressed as:

How to obtain the coding matrix G through coding, can adopt the following two schemes:

Option One:

Step 1: Through the obtained values of T, B, and N, generate an n×k Vandermonde matrix G1 as shown below:

Step 2: Perform column transformation on G1, and change the upper k×k matrix of G1 into the identity matrix to obtain G.

Option II:

Step 1: Through the obtained values of T, B, and N, generate an n×k Cauchy matrix G2 as shown below:

Among them, the value of the element in the i-th row and j-th column in G2 is

Step 2: Perform column transformation on G2, and change the upper k×k matrix of G2 to the identity matrix to obtain G.

The following describes the application scenarios of the present invention.

Figure 4a is a schematic diagram of data transmission in a 5G network scenario. As shown in Figure 4a, the high-definition video source in the server is encoded by the FEC encoder to obtain a video stream. For example, the server obtains the delay T, the number of consecutive packet losses B and the number of burst packet losses N within the delay T, and encodes the high-definition video source according to the delay T, the number of consecutive packet losses B and the number of burst packet losses N , In order to obtain the video code stream, and transmit the video code stream to the user terminal through the 5G network. The 5G network includes the core network, base stations, and Customer Premise Equipment (CPE). The user end can be an AR/VR helmet that supports connection with CPE or a smart phone or handheld game device, smart TV, smart box, etc.

Similarly, Figure 4b is a schematic diagram of data transmission in a WiFi scenario. As shown in Figure 4b, the high-definition video source in the server is encoded by the FEC encoder to obtain a video stream, which is transmitted to the user end through the backbone network and WiFi equipment.

After the user terminal obtains the video code stream, the FEC decoder in the user terminal decodes the video code stream to obtain a high-definition video source. Specifically, the time delay T, the number of consecutive packet losses B and the number of burst packet losses N within the time delay T are obtained, and the video stream is decoded according to the time delay T, the number of consecutive packet losses B and the number of burst packet losses N, In order to get high-definition video source.

Referring to FIG. 5, FIG. 5 is a schematic flowchart of a data encoding method according to an embodiment of the present invention. As shown in Figure 5, the method includes:

S501: The server obtains the data to be encoded, and obtains the target delay T and the number of consecutive packet losses B and the number of burst packet losses N within the target delay T.

Among them, the target delay T is the delay required in the data transmission process. The data to be encoded can be data that requires a high delay, such as live video data, VR/AR video data, and so on.

Preferably, the target delay T is the maximum delay required in the data transmission process, for example, the maximum delay required by the application on the user side, the number of consecutive packet loss B and the number of burst packet loss N are the target delay in the cedar tree The maximum number of consecutive packet losses and the maximum number of burst packet losses within T. Among them, burst packet loss is caused by the channel, for example, burst packet loss is caused by fluctuations in channel bandwidth or interference from other users.

In a feasible embodiment, the server obtaining the target delay T includes:

The server obtains the service type corresponding to the data to be coded, and determines the target delay T according to the service type of the data to be coded.

Specifically, on the server side, different types of services are processed by different APPs, and different APPs require different QOS. Therefore, the target delay T can be determined based on the type of service corresponding to the data to be encoded.

Optionally, the target delay T may be a duration of 6 frames or a duration of 10 frames.

By determining the target delay T according to the service type corresponding to the data to be encoded, the delay requirements of different types of services can be met.

In a feasible embodiment, the server acquiring the target delay T includes:

The server receives the reference delay sent by multiple clients, and the reference delay of the client U is the delay required by the client U. The client U is any one of the multiple clients, according to the reference sent by the multiple clients The delay determines the target delay.

Wherein, the target delay T is a maximum value, a minimum value, an average value of the reference delays of a plurality of user terminals or is obtained by prediction based on the reference delays of the plurality of user terminals.

It should be noted here that the reference delay of the user end U is the maximum time difference between the service transmission time frame and the frame under the premise that the application of the user end U is not stuck or blurred during the data transmission process.

Further, after obtaining the target time delay T, the server sends the target time delay T to the above-mentioned multiple client terminals through broadcast or unicast, so that the delays used by the server and the client terminals are consistent.

In a feasible embodiment, the server obtains the number of consecutive packet losses B and the number of burst packets N within the target delay T, including:

The server obtains multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the user terminal U, and determines the consecutive packet loss number B according to the multiple historical consecutive packet losses, and according to the multiple A historical burst loss number determines the burst loss number N, and the user terminal U is any one of the user terminals that receive the data to be encoded; or;

The server obtains multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with multiple clients, determines the consecutive packet loss number B according to the multiple historical consecutive packet losses, and The number of historical bursts of packet loss determines the number of bursts of packet loss N; multiple user terminals are all user terminals that receive the data to be encoded.

Specifically, after the server obtains the multiple historical continuous packet loss numbers and multiple historical burst packet loss numbers used in the process of data transmission with the user terminal U, it determines the continuous packet loss number B according to the multiple historical continuous packet loss numbers. Determine the burst packet loss number N according to multiple historical burst packet loss numbers. Among them, the number of consecutive packet losses B is the maximum, minimum, or average value of multiple historical consecutive packet losses, or the number of consecutive packet losses B is obtained by predicting multiple historical consecutive packet losses. The burst loss number N is the maximum, minimum, or average value among multiple historical burst loss numbers, or the burst loss number N is obtained by predicting multiple historical burst loss numbers.

More specifically, after the server obtains the multiple historical continuous packet loss numbers and multiple historical burst packet loss numbers used in the process of data transmission with multiple clients, it determines the continuous packet loss based on the historical continuous packet loss numbers of the multiple clients. For packet loss number B, the burst packet loss number N is determined according to the historical burst packet loss numbers of multiple users. Among them, the number of consecutive packet losses B is the maximum, minimum, or average value of the historical consecutive packet losses of multiple clients, or the number of consecutive packet losses B is obtained by predicting the historical consecutive packet losses of multiple clients . The burst loss number N is the maximum, minimum, or average value of the historical burst loss numbers of multiple clients, or the burst loss number N is to predict the historical burst loss numbers of multiple clients owned.

Optionally, the number of consecutive packet losses B is obtained by smoothing/averaging multiple historical consecutive packet losses. Similarly, the number of burst losses N is smoothing/averaging multiple historical packet losses. owned. Smoothing/averaging processing includes: arithmetic average method, exponential average method, or geometric average method, although the present invention is not limited to this.

Taking the arithmetic average method as an example, the burst packet loss number N can be expressed as:

Optionally, the number of consecutive packet losses B is obtained by prediction based on multiple historical consecutive packet losses. Similarly, the number of burst packet losses N is obtained by prediction based on multiple historical burst packet losses.

For example, the number of burst packet losses N=α ₁ N ₁ +α ₂ N ₂ +…+α _M N _M , where the prediction parameters α ₁ ,α ₂ ,…,α _M are estimated using the parameters in the existing linear prediction scheme Algorithm is obtained.

_{_{N 1, N 2, ...,}} N M M number of bursts for the first historical loss. The time required to count M times of data is the preset time. Within the preset time, the change of the maximum burst loss to the smallest burst loss is not greater than a certain threshold, such as 10%, the threshold It can be adjusted according to actual needs.

It should be noted here that the number of consecutive lost packets B can also be obtained by the above method, and will not be described here.

Further, after obtaining the number of consecutive packet losses B and the number of burst packet losses N, the server sends the number of consecutive packet losses B and the number of burst packet losses N to the user terminal U via unicast, or; the number of consecutive packet losses B And the number of burst packet losses N are sent to the aforementioned multiple client terminals through broadcast, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple client terminals through unicast.

The server obtains the reference continuous packet loss number and the reference burst packet loss number sent by multiple client terminals; determines the continuous packet loss number B according to the reference continuous packet loss number sent by multiple client terminals, and determines the continuous packet loss number B according to the reference sudden packet loss sent by multiple client terminals. The number of lost packets determines the number of burst packets N;

In a feasible embodiment, after the server obtains the maximum continuous packet loss number B and the maximum burst packet loss number N within the target delay T, the server broadcasts the continuous packet loss number B and the burst packet loss number N to the maximum A user end, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple user ends via unicast.

Specifically, the parameter interaction between the server and the user terminal can be through the independent signaling channel interaction mode or the channel associated signaling interaction mode.

Among them, the independent signaling channel interaction mode is important coding information for the server and the user side, such as T, B, and N information, which can be transmitted by independently initiating a communication link. As shown in Figure 6a, the format of the transmitted data is not limited. When sending signaling based on the independent signaling channel of the scheme, the sending of the signaling should match the current data, that is, the data and the signaling are sent through different channels, but they can be considered synchronized in time, which can be achieved by aligning the respective chains. Time stamp or unified frame number is used to ensure time synchronization.

The channel-associated signaling interaction mode means that the signaling and data between the server and the user are transmitted in the same communication link, as shown in Figure 6b. The format of the transmitted data is shown in Figure 7. It can be format I or format II.

In a feasible embodiment, the multiple client terminals are: multiple client terminals that use the same transmission mode to receive the data to be encoded, or; multiple client terminals that process the same type of service, or; multiple required QoS levels are the same Or the same client as QoE.

Specifically, the classification of the user side that performs data with the server can be classified based on the transmission mode, and can be classified according to the type of service processed by the user side, or the QoS or QoE required by the user side. Among them, the transmission methods include 5G network, WiFi and optical fiber; the types of services handled by the user end are classified based on the types of APPs in the user end, such as music, video, and information.

S502. The server encodes the to-be-coded data according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a code stream.

In a feasible embodiment, the server encodes the to-be-coded data according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a code stream, including:

The server determines k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords before encoding, and n is the number of codewords after encoding. The data to be encoded is encoded according to k and n, To get the code stream.

Further, the server determines k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses, including:

The server determines the value range of the channel coding efficiency R according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N; the capacity C is determined according to the channel coding efficiency R, and k and n are determined according to the capacity C.

Among them, the capacity C is the maximum value of the channel coding efficiency R, and the capacity C is greater than or equal to k/n.

It should be noted here that the to-be-coded data is encoded according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain the code stream. For the specific implementation process, please refer to the relevant description of the embodiment shown in FIG. 3 , No longer describe here.

It can be seen that in the solution of the present application, the data to be encoded is encoded according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a bit stream, which is beneficial to solve the problem of continuous loss in data transmission. The problem of packet loss and discrete bursts. In addition, different delays are used when transmitting data of different types of services, which meets the delay requirements of different types of services.

Referring to FIG. 8, FIG. 8 is a schematic flowchart of a data decoding method provided by an embodiment of the present invention. As shown in Figure 8, the method includes:

S801. The user terminal obtains the code stream, and obtains the target delay T, and obtains the number of consecutive packet losses B and the number of burst packet losses N within the target delay T.

Among them, the target delay T is the delay required in the data transmission process.

Optionally, the target delay T is the maximum delay required in the data transmission process, for example, an application needs to use decoded data, and the target delay T is the maximum delay required by the application. The number of consecutive packet losses B is the maximum number of consecutive packet losses within the target delay T, and the number of burst losses N is the maximum number of burst packet losses within the target delay T.

In a feasible embodiment, the user side acquiring the target delay T includes:

The user side determines the time delay received from the server as the target time delay T, or; the user side determines the time delay it needs as the target time delay T.

In a feasible embodiment, the user side acquiring the target delay T includes:

The user end sends a reference delay to the server, and the reference delay is the delay required by the user end. The client receives the target delay T sent by the server, and the target delay T is obtained by the server according to the reference delays sent by multiple clients.

It should be noted here that the delay required by the user end can be understood as the time difference between the service transmission time frame and the frame when the application of the user end is not stuck or blurred during the data transmission process.

In a feasible embodiment, the user terminal obtains the number of consecutive packet losses B and the number of burst packet losses N within the target delay T, including:

The user terminal obtains multiple historical continuous packet loss numbers and multiple historical burst packet loss numbers used in the process of data transmission with the server; the user terminal determines the continuous packet loss number B according to the multiple historical continuous packet losses, and according to the multiple A historical burst loss number determines the burst loss number N.

Among them, the number of consecutive packet losses B is obtained by smoothing/averaging multiple historical consecutive packet losses. Similarly, the number of burst losses N is obtained by smoothing/averaging multiple historical packet losses. . Smoothing/averaging processing includes: arithmetic average method, exponential average method, or geometric average method, although the present invention is not limited to this.

In a feasible embodiment, after the user terminal determines the number of consecutive packet losses B according to multiple historical consecutive packet losses, and after determining the number of burst packet losses N according to the multiple historical burst packet losses, the data decoding method further includes:

The user end sends the number of consecutive packet losses B and the number of burst packet losses N to the server.

The user side sends the reference number of consecutive packet losses and the reference number of burst packet losses to the server. The reference number of consecutive packet losses is obtained by the user end U according to multiple historical consecutive packet losses. The reference number of burst packet losses is the user end U according to Obtained from multiple historical burst packet losses;

The client receives the number of consecutive packet losses B and the number of burst packet losses N sent by the server. The number of consecutive packet losses B is obtained by the server based on the number of consecutive packet losses from multiple clients. The number of burst packet losses N is the number It is obtained by referring to the number of burst packet losses at the user end, and the multiple user ends include the above-mentioned user end.

In a feasible embodiment, the multiple client terminals are:

It should be noted here that the specific description of step S801 can refer to the related description of step S501 above, which will not be described here.

S802. The user terminal decodes the code stream according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain decoded data.

In a feasible embodiment, the user terminal decodes the code stream according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain decoded data, including:

The user terminal determines k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords after decoding, and n is the number of codewords before decoding;

The user terminal decodes the data to be decoded according to k and n to obtain the decoded data.

In a feasible embodiment, the user terminal determines k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, including:

The user terminal determines the value range of the channel coding efficiency R according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N; the channel coding efficiency R is determined as the capacity C, and k and n are determined according to the capacity C, where , K/n≤C, C is the maximum value of channel coding efficiency R.

It should be noted here that the code stream is encoded according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain decoded data. The specific implementation process can refer to the embodiment shown in FIG. 3 Relevant descriptions will not be described here.

It can be seen that in the solution of the embodiment of the present invention, the code stream is decoded according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain the decoded data, which is beneficial to solve the problem of data loss. The problem of continuous packet loss and discrete burst packet loss in transmission. In addition, different delays are used when transmitting data of different types of services, which meets the delay requirements of different types of services.

Refer to FIG. 9, which is a schematic structural diagram of a data encoding and decoding system according to an embodiment of the present invention. As shown in Figure 9, the data encoding and decoding system includes a server and a client. Among them, the server includes APP interface module, FEC encoder, application layer and physical layer interface, physical layer and server parameter prediction management module; among them, the server parameter prediction management module includes APP classification/parser and parameter prediction module. The user side includes APP interface module, FEC decoder, application layer and physical layer interface, physical layer and user side parameter statistics management module; among them, the user side parameter statistics management module includes APP classification/parser and parameter estimation module.

First, introduce the functions of each functional module of the server.

The APP interface module is used to process the data of each application and convert the data of the application into data that can be processed by the FEC encoder, that is, split or combine the video of the APP or the original data to be encoded for FEC encoding and encoding.

The FEC encoder is used to encode the data of each application to enhance the robustness of the device against channel transmission. For the specific encoding process, refer to the related description of FIG. 3. Among them, the parameters required by the FEC encoder, such as T, B, and N, are obtained through the server parameter prediction management module.

Application layer and physical layer interface: Since the data encoded by the FEC encoder cannot be directly transmitted to the physical layer for transmission, it must be processed and converted. If it is application data, it needs to add the corresponding header. If it is signaling, Need to do protection processing, specific protection processing can refer to the general signaling transmission scheme.

The physical layer corresponds to the actual 5G system or optical fiber network or WiFi network to transmit APP data.

The server parameter prediction management module is used to obtain the target delay T, the number of consecutive packet losses B and the number of burst packets N. Among them, the target delay T is strongly correlated with the application, and the number of consecutive packet losses B and the number of burst packet losses N are strongly correlated with the channel.

The target delay T, the number of consecutive packet losses B, and the number of burst packet losses N can be obtained by the server or the client, and then shared with the other party.

The target delay T is set according to the delay requirements of the actual application. For the server, it can be obtained according to the QoS of the APP in the server. In specific implementation, the APP can be classified by the classification parser in the server parameter prediction management module. According to the classification results, different T values are given. The T value can be set to 6 frames long time or 10 frames long time, or for the user side, the target delay can be maintained through the user side parameter statistics management module T, the value of T at this time can be counted by the application, based on the following methods: For example, when recording and watching the video, there is no blurring, freezing or between the adjacent data packets/data frames required when the video and audio are not synchronized. The maximum delay.

For the number of consecutive packet loss B and the number of burst packet loss N, since the channel is time-varying in practical applications, the packet loss situation caused by it is sometimes constant, and sometimes there may be large differences. Therefore, the real-time statistics of the packet loss length and The number of packet losses, that is, the number of consecutive packet losses B and the number of burst packet losses N, etc. need to be counted according to actual needs. The statistical module can be placed in the server or on the user side. The following is described separately from the server side and the client side:

On the server side, the server parameter prediction management module obtains the number of consecutive packet losses B and the number of burst packet losses N. The following methods can be specifically adopted:

1) Separate statistics of B and N for each client:

For the server, it is considered that the information used by each client when transmitting data is different, so the B, N and other decoding corresponding parameters used by each client are separately counted. For example, for a user terminal that plays a video, regardless of whether it uses the same APP for playback, it is necessary to separately count the B and N when transmitting data. The advantage of this statistical method is that the optimal coding and decoding parameters can be adapted to each user end, and the experience of each user can be guaranteed to the greatest extent. Since the B and N values of the actual system will change over time, the statistical scheme includes:

Statistical scheme 1: The value of the number of consecutive packet losses B can be the value obtained by smoothing/averaging the number of historical consecutive packet losses in a period of time; in the same way, the value of the number of burst packet losses N can be the value of a period of time The value obtained by smoothing/averaging historical continuous packet loss. The smoothing/average processing scheme includes, but is not limited to, arithmetic average method, exponential average method, or geometric average method. For example, the number of burst packet loss N can be expressed as:

N ₁ ,N ₂ ,...,N _M is the number of historical consecutive packet losses calculated for the previous M times. The duration required for counting M times is the preset duration. Within the preset duration, the largest packet loss is relative to the smallest packet loss change. Not greater than a certain threshold, such as 10%, the threshold can be adjusted according to actual needs.

Statistical scheme 2: The number of consecutive packet losses B can be predicted based on the historical number of consecutive packet losses in a period of time. Similarly, the number of burst packet losses N can be based on the number of historical packet losses in a period of time. The prediction method can be based on linear prediction model or autoregressive prediction model. Among them, the linear prediction scheme is as follows:

N=α ₁ N ₁ +α ₂ N ₂ +…+α _M N _M

α ₁ ,α ₂ ,...,α _M are prediction parameters, which are obtained by using the parameter estimation algorithm in the existing linear prediction scheme. N ₁ ,N ₂ ,...,N _M is the number of historical consecutive packet losses calculated for the previous M times. The duration required for counting M times is the preset duration. Within the preset duration, the largest packet loss is relative to the smallest packet loss change. Not greater than a certain threshold, such as 10%, the threshold can be adjusted according to actual needs.

Statistical scheme 3: No processing is performed on the statistical data, and the effective parameters are the values obtained from the current statistics. In other words, the number of consecutive packet losses B and the number of burst packet losses N are the number of consecutive packet losses and the number of burst packet losses historically used by the server in the process of data transmission with the client.

2) Classify the user end to determine the number of consecutive packet losses B and the number of burst packet losses N

The methods for classifying the user end include: classification can be based on the data transmission mode of the current APP; classification can be based on the APP of the user end; and classification can also be carried out according to the QoS or QoE level required by the application in the user end.

For example, the user terminals for data transmission via 5G are divided into one type, the user terminals for data transmission via WiFi are divided into one type, and the user terminals for data transmission via optical fiber are divided into one type. When determining the number of consecutive packet losses B and the number of burst packet losses N, statistics are calculated according to the transmission mode (which can be regarded as a channel). The advantage of this is to optimize the efficiency of obtaining B and N under the premise of ensuring a reduced APP experience, and indirectly improve the channel utilization rate (because the transmission of B and N both require bandwidth).

After classifying the user terminal, B and N can be obtained by taking the maximum value and the average value. An example of the scheme using the maximum value is as follows:

N=max(N ₁ ,N ₂ ,…,N _M )

N ₁ , N ₂ ,..., N _M are the previous M historical consecutive packet loss counts for a certain client in the first category. The time required to count M times is the preset duration. Within the preset duration, the largest loss is The relative minimum packet loss change is not greater than a certain threshold, such as 10%, and the threshold can be adjusted according to actual needs.

On the user side, the user-side parameter statistics management module obtains the number of consecutive packet losses B and the number of burst packet losses N. The following methods can be specifically adopted:

The specific method for the user side to obtain the number of consecutive packet losses B and the number of burst packet losses N is the same as that of the server for the number of consecutive packet losses B and the number of burst packet losses. The number of lost packets is averaged or smoothed, or predicted. Different from the server side, for B and N, the user side can only obtain its own historical burst packet loss number and historical continuous packet loss number, and cannot obtain other devices, nor can it be distinguished by channel type. The same as the server is that its statistical B and N can be classified by application types. For example, Tencent Music and NetEase Cloud Music are classified in the same category, and the audio category is used for statistics. The statistical application scale is much smaller than that of the server, and the calculation is complicated. Degree is low.

Whether it is the server or the client, after obtaining the relevant parameters (such as T, B, N), the server generates the required code stream according to the required code stream, and the client decodes according to the latest parameters (such as T, B, N) Recover the original data with the received data and encoded information.

In the above process, because the corresponding important parameters such as T, B, and N are necessary parameters for encoding and decoding, the server and the client need to be shared and consistent. Therefore, the important parameters need to be interacted by the server and the user, and the parameter interaction method can be There are two types:

Interactive plan 1: Server-based plan:

The server-based solution means that the T, B, and N used in data transmission between the server and the client are based on what the server obtains, and the server broadcasts or unicasts the corresponding T, B, and N to the client;

The broadcast scheme corresponds to the scenario of classification statistics. The user terminal is classified according to the category of its APP, and classified as the same type of user. The server broadcasts the parameters of the same type of user terminal, and the user terminal receives the broadcast T, B, N. That is to update.

Unicast is mainly for the scenario where each user counts the parameters separately. The server unicasts T, B, and N for each user, and the user updates the T, B, and N after receiving it.

Interactive plan 2: A plan based on user feedback:

After the user terminal feeds back the acquired parameters (such as T, B, N) to the server, the server acquires and updates the required parameters according to the parameters fed back by the user.

There are two scenarios. If the parameters of each user end are counted separately, the feedback parameters of the user end are only for the channel condition between it and the server. At this time, the parameters used by the encoder of the server are the parameters feedback from the user or the parameters for the user. The parameters fed back by the client are averaged to obtain the parameters; if the parameters used by the encoder of the server are obtained after averaging the parameters fed back by the client, the server needs to feed back the parameters to the client so that the server and the client can use the parameters. The parameters are the same.

If it is for a classified scenario, the server needs to judge and update the encoder parameters based on the parameters fed back by multiple clients. The cloud parameter processing scheme can use the aforementioned parameter feedback from multiple clients to average or take the maximum value. Etc., for the specific process, please refer to the aforementioned related description. The updated parameters need to be notified to the user. At this time, the user can obtain it by way of channel-associated signaling, or obtain its latest parameters through broadcast signals.

The parameter information interaction between the server and the user can be periodic, aperiodic, and periodic, and the update period does not exceed the coherence time of the current channel. The preferred update method is the periodic update method, which is more robust. This method can ensure that when the server cannot normally receive parameter information during a certain update, it can make appropriate remedies according to the parameters of the next cycle, avoiding the situation that the data of the server and the user cannot be synchronized for a long time, and it can adapt to the channel. The change.

The other sub-modules in the newly added module on the user side can be equivalent to the cloud function. For example, the parameter estimation module corresponds to the parameter prediction module of the server. The implementation method can refer to the server module, and the APP classification parser is similar to that of the server. I won't repeat them here.

The parameters T, B, and N are mainly provided to the corresponding decoding information of the FEC decoder. At this time, the decoder can decode the received data after obtaining the latest T, B, and N according to the parameter statistics module on the user side. There are many decoding schemes, which can be based on matrix inversion method or general RS code decoding method, such as: Berlekamp-Massey decoding method, Peterson-Gorenstein-Zierler decoding method, discrete Fourier transform method, etc.

Take matrix inversion decoding as an example:

First, according to the information of T, B, and N and the auxiliary information, the user end can generate a coding generation matrix G consistent with that of the server end. The specific generation method can refer to the relevant description above, which will not be described here. After the generator matrix is obtained, the original data can be recovered by the Gaussian elimination method according to the received data and the matrix G. The following formula:

Gs=r

Among them, G is the generator matrix, s is the data to be restored, and r is the received data. If you want to get s, it is equivalent to aligning and finding a system of linear equations, and s can be obtained based on the Gaussian elimination method. How to do the Gaussian elimination method can refer to the existing linear algebra textbooks, which will not be repeated here.

Refer to FIG. 10, which is a schematic structural diagram of a server according to an embodiment of the present invention. As shown in FIG. 10, the server 1000 includes:

The obtaining unit 1001 is used to obtain the data to be encoded, and obtain the target time delay T and obtain the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T. The target time delay T is determined during the data transmission process. Required delay;

The encoding unit 1002 is configured to encode the to-be-coded data according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain a code stream.

In a feasible embodiment, the encoding unit 1002 is specifically configured to:

In a feasible embodiment, in terms of determining k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, the encoding unit 1002 is configured to:

In a feasible embodiment, in terms of acquiring the target delay T, the acquiring unit 1001 is specifically configured to:

In a feasible embodiment, in terms of obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the obtaining unit 1002 is specifically configured to:

Further, the server 1000 further includes:

The sending unit 1003 is configured to, after the obtaining unit obtains the number of consecutive packet losses B and the number of burst packet losses N within the target delay T, send the number of consecutive packet losses B and the number of burst packet losses N to the user end via unicast U, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple client terminals through broadcast, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple client terminals through unicast.

In a feasible embodiment, the server 1000 further includes:

The sending unit 1003 is configured to, after obtaining the target delay T, send the target delay T to multiple client terminals through broadcast, or; send the target delay T to multiple client terminals through unicast.

In a feasible embodiment, in terms of acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the acquiring unit 1001 is specifically configured to:

In a feasible embodiment, the server 1000 further includes:

The sending unit 1003 is configured to send the number of consecutive packet losses B and the number of burst packet losses N through broadcast to the maximum number of consecutive packet losses B and the maximum number of burst packet losses N within the target time delay T by the obtaining unit 1001. A user end, or; the number of consecutive packet losses B and the number of burst packet losses N are sent to multiple user ends via unicast.

It should be noted that the aforementioned units (the acquiring unit 1001, the encoding unit 1002, and the sending unit 1003) are used to execute the relevant steps of the aforementioned method. For example, the acquiring unit 1001 and the sending unit 1003 are used to execute related content of S501, and the encoding unit 1002 is used to execute related content of S502.

In this embodiment, the server 1000 is presented in the form of a unit. The "unit" here can refer to an application-specific integrated circuit (ASIC), a processor and memory that executes one or more software or firmware programs, an integrated logic circuit, and/or other devices that can provide the above-mentioned functions . In addition, the above acquisition unit 1001 and encoding unit 1002 may be implemented by the processor 1201 of the server shown in FIG. 12.

Refer to FIG. 11, which is a schematic structural diagram of a user terminal according to an embodiment of the present invention. As shown in FIG. 11, the user terminal 1100 includes:

The acquiring unit 1101 is used to acquire the code stream and acquire the target delay T, and acquire the number of consecutive packet losses B and the number of burst packet losses N within the target delay T; the target delay T is required during data transmission Time delay

The decoding unit 1102 is configured to decode the code stream according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain decoded data.

In a feasible embodiment, the decoding unit 1102 is specifically configured to:

In a feasible embodiment, in terms of acquiring the target delay T, the acquiring unit 1101 is specifically configured to:

Send the reference delay to the server, the reference delay is the delay required by the client 1100; the target delay T sent by the server is received, and the target delay T is obtained by the server according to the reference delays of multiple clients. Including the client 1100.

In a feasible embodiment, in terms of obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the obtaining unit 1101 is specifically configured to:

In a feasible embodiment, the user terminal 1100 further includes:

The sending unit 1103 is configured to send the number of consecutive packet losses B and the number of consecutive packet losses to the server after determining the number of consecutive packet losses B according to multiple historical consecutive packet losses, and after determining the number of burst packet losses N according to the multiple historical burst packet losses The number of burst packet loss N.

Send the reference continuous packet loss number and the reference burst packet loss number to the server. The reference continuous packet loss number is the user end 1100 based on multiple historical consecutive packet losses. The reference burst loss number is the user end 1100 based on multiple Obtained from historical burst packet loss;

Receive the number of consecutive packet losses B and the number of burst packet losses N sent by the server. The number of consecutive packet losses B is obtained by the server based on the number of consecutive packet losses from multiple clients. The number of burst packet losses N is the server's Based on the number of burst packet losses, multiple client terminals include client terminal 1100.

In a feasible embodiment, the multiple client terminals are:

It should be noted that the aforementioned units (the acquiring unit 1101, the decoding unit 1102, and the sending unit 1103) are used to execute the relevant steps of the aforementioned method. For example, the acquiring unit 1101 and the sending unit 1103 are used to execute related content of S801, and the decoding unit 1102 is used to execute related content of S802.

In this embodiment, the user terminal 1100 is presented in the form of a unit. The "unit" here can refer to an application-specific integrated circuit (ASIC), a processor and memory that executes one or more software or firmware programs, an integrated logic circuit, and/or other devices that can provide the above-mentioned functions . In addition, the above obtaining unit 1101 and decoding unit 1102 may be implemented by the processor 1301 on the user side shown in FIG. 13.

The server 1200 shown in FIG. 12 may be implemented in the structure shown in FIG. 12. The server 1200 includes at least one processor 1201, at least one memory 1202 and at least one communication interface 1203. The processor 1201, the memory 1202, and the communication interface 1203 are connected through the communication bus and complete mutual communication.

The processor 1201 may be a general-purpose central processing unit (CPU), a microprocessor, an application-specific integrated circuit (ASIC), or one or more integrated circuits for controlling the execution of the above program programs.

The communication interface 1203 is used to communicate with other devices or communication networks, such as Ethernet, wireless access network (RAN), wireless local area network (Wireless Local Area Networks, WLAN), etc.

The memory 1202 may be a read-only memory (ROM) or other types of static storage devices that can store static information and instructions, random access memory (RAM), or other types that can store information and instructions The dynamic storage device can also be electrically erasable programmable read-only memory (Electrically Erasable Programmable Read-Only Memory, EEPROM), CD-ROM (Compact Disc Read-Only Memory, CD-ROM) or other optical disc storage, optical disc storage (Including compact discs, laser discs, optical discs, digital versatile discs, Blu-ray discs, etc.), magnetic disk storage media or other magnetic storage devices, or can be used to carry or store desired program codes in the form of instructions or data structures and can be used by a computer Any other media accessed, but not limited to this. The memory can exist independently and is connected to the processor through a bus. The memory can also be integrated with the processor.

Wherein, the memory 1202 is used to store application program codes for executing the above solutions, and the processor 1201 controls the execution. The processor 1201 is configured to execute application program codes stored in the memory 1202.

The code stored in the memory 1202 can perform any of the data encoding methods provided above, such as: acquiring the data to be encoded, and acquiring the target delay T and acquiring the number of consecutive packet losses B and the number of burst packet losses within the target delay T N, the target delay T is the delay required in the data transmission process; the data to be encoded is encoded according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain a code stream.

The user terminal 1300 shown in FIG. 13 may be implemented with the structure in FIG. 13, and the user terminal 1300 includes at least one processor 1301, at least one memory 1302, and at least one communication interface 1303. The processor 1301, the memory 1302, and the communication interface 1303 are connected through the communication bus and complete mutual communication.

The processor 1301 may be a general-purpose central processing unit (CPU), a microprocessor, an application-specific integrated circuit (ASIC), or one or more integrated circuits for controlling the execution of the above program programs.

The communication interface 1303 is used to communicate with other devices or communication networks, such as Ethernet, wireless access network (RAN), wireless local area network (Wireless Local Area Networks, WLAN), etc.

The memory 1302 can be a read-only memory (ROM) or other types of static storage devices that can store static information and instructions, random access memory (RAM), or other types that can store information and instructions The dynamic storage device can also be electrically erasable programmable read-only memory (Electrically Erasable Programmable Read-Only Memory, EEPROM), CD-ROM (Compact Disc Read-Only Memory, CD-ROM) or other optical disc storage, optical disc storage (Including compact discs, laser discs, optical discs, digital versatile discs, Blu-ray discs, etc.), magnetic disk storage media or other magnetic storage devices, or can be used to carry or store desired program codes in the form of instructions or data structures and can be used by a computer Any other media accessed, but not limited to this. The memory can exist independently and is connected to the processor through a bus. The memory can also be integrated with the processor.

Wherein, the memory 1302 is used to store application program codes for executing the above solutions, and the processor 1301 controls the execution. The processor 1301 is configured to execute application program codes stored in the memory 1302.

The code stored in the memory 1302 can perform any of the data encoding methods provided above, such as: obtaining the code stream, and obtaining the target delay T, and obtaining the number of consecutive packet loss B and the number of burst packet loss N within the target delay T ; The target delay T is the delay required in the data transmission process; the code stream is decoded according to the target delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain the decoded data.

It should be noted here that in practical applications, the above-mentioned server can be regarded as an encoding device, and correspondingly, the client can be regarded as a decoding device.

An embodiment of the present invention further provides a computer storage medium, wherein the computer storage medium may store a program, and the program includes part or all of the steps of any data encoding and decoding method recorded in the above method embodiment when the program is executed.

It should be noted that for the foregoing method embodiments, for the sake of simple description, they are all expressed as a series of action combinations, but those skilled in the art should know that the present invention is not limited by the described sequence of actions. Because according to the present invention, certain steps can be performed in other order or simultaneously. Secondly, those skilled in the art should also know that the embodiments described in the specification are all preferred embodiments, and the actions and modules involved are not necessarily required by the present invention.

In the above-mentioned embodiments, the description of each embodiment has its own emphasis. For parts that are not described in detail in an embodiment, reference may be made to related descriptions of other embodiments.

In the several embodiments provided in this application, it should be understood that the disclosed device may be implemented in other ways. For example, the device embodiments described above are merely illustrative. For example, the division of the units is only a logical function division, and there may be other divisions in actual implementation, for example, multiple units or components may be combined or may be Integrate into another system, or some features can be ignored or not implemented. In addition, the displayed or discussed mutual coupling or direct coupling or communication connection may be indirect coupling or communication connection through some interfaces, devices or units, and may be in electrical or other forms.

The units described as separate components may or may not be physically separated, and the components displayed as units may or may not be physical units, that is, they may be located in one place, or they may be distributed on multiple network units. Some or all of the units may be selected according to actual needs to achieve the objectives of the solutions of the embodiments.

In addition, the functional units in the various embodiments of the present invention may be integrated into one processing unit, or each unit may exist alone physically, or two or more units may be integrated into one unit. The above-mentioned integrated unit can be implemented in the form of hardware or software functional unit.

If the integrated unit is implemented in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer readable memory. Based on this understanding, the technical solution of the present invention essentially or the part that contributes to the existing technology or all or part of the technical solution can be embodied in the form of a software product, and the computer software product is stored in a memory, A number of instructions are included to make a computer device (which may be a personal computer, a server, or a network device, etc.) execute all or part of the steps of the method described in each embodiment of the present invention. The aforementioned memory includes: U disk, Read-Only Memory (ROM, Read-Only Memory), Random Access Memory (RAM, Random Access Memory), mobile hard disk, magnetic disk or optical disk and other media that can store program codes.

Those of ordinary skill in the art can understand that all or part of the steps in the various methods of the above-mentioned embodiments can be completed by a program instructing relevant hardware. The program can be stored in a computer-readable memory, and the memory can include: a flash disk , Read-only memory (English: Read-Only Memory, abbreviated as: ROM), random access device (English: Random Access Memory, abbreviated as: RAM), magnetic disk or optical disc, etc.

The embodiments of the present invention are described in detail above, and specific examples are used in this article to illustrate the principles and implementation of the present invention. The descriptions of the above embodiments are only used to help understand the methods and core ideas of the present invention; at the same time, for Those of ordinary skill in the art, based on the idea of the present invention, will have changes in the specific implementation and the scope of application. In summary, the content of this specification should not be construed as limiting the present invention.

Claims

A data encoding method, characterized in that it comprises:

Obtain the data to be encoded, and obtain the target delay T and the number of consecutive packet losses B and the number of burst packet losses N within the target delay T; the target delay T is the time required in the data transmission process Extend

The data to be encoded is encoded according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a code stream.
The method according to claim 1, wherein the data to be encoded is encoded according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain Code stream, including:

Determine k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords before encoding, and the n is the number of codewords after encoding;

The data to be encoded is encoded according to the k and n to obtain the code stream.
The method according to claim 2, wherein the determining k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N comprises:

Determine the value range of the channel coding efficiency R according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N;

The capacity C is determined according to the channel coding efficiency R, and the k and n are determined according to the capacity C.
The method according to any one of claims 1 to 3, wherein the obtaining the target time delay T comprises:

Obtain the service type corresponding to the data to be encoded, and determine the target delay T according to the service type corresponding to the data to be encoded.
The method according to any one of claims 1 to 3, wherein the acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T comprises:

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the decoding device U, and determine the consecutive packet losses B according to the multiple historical consecutive packet losses, And determining the burst packet loss number N according to the multiple historical burst packet loss numbers, and the decoding device U is any one of the decoding devices that receive the data to be encoded; or;

Acquiring multiple historical continuous packet loss numbers and multiple historical burst packet loss numbers used in the process of data transmission with multiple decoding devices, and determining the continuous packet loss number B according to the multiple historical continuous packet loss numbers, And determining the burst packet loss number N according to the multiple historical burst packet loss numbers; the multiple decoding devices are all decoding devices that receive the data to be encoded.
The method according to any one of claims 1 to 3, wherein the obtaining the target time delay T comprises:

Receiving reference delays sent by multiple decoding devices, where the reference delay of the decoding device U is the delay required by the decoding device U;

Determine the target delay T according to the reference delay sent by the multiple decoding apparatuses;

Wherein, the decoding device U is any one of the plurality of decoding devices.
The method according to claim 6, characterized in that, after the acquisition of the target delay T, the method further comprises:

Sending the target delay T to the multiple decoding devices via broadcast, or;

The target time delay T is unicast to the multiple decoding devices.
The method according to any one of claims 1 to 3, wherein the acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T comprises:

Acquiring the reference continuous packet loss number and the reference burst packet loss number sent by multiple decoding devices;

The number of consecutive packet losses B is determined according to the reference number of consecutive packet losses sent by the multiple decoding apparatuses, and the number of burst packet losses N is determined according to the reference burst packet losses sent by the multiple decoding apparatuses.
The method according to claim 8, wherein after the obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the method further comprises:

Sending the number of consecutive packet losses B and the number of burst packet losses N to the plurality of decoding apparatuses through broadcast, or;

The number of consecutive packet losses B and the number of burst packet losses N are sent to the plurality of decoding apparatuses through unicast.
The method according to any one of claims 5-9, wherein the multiple decoding devices are:

Multiple decoding devices that use the same transmission mode for receiving the data to be encoded, or;

Multiple decoding devices processing the same type of service, or;

Multiple decoding devices with the same required quality of service QoS level or the same quality of experience QoE.
A data decoding method, characterized in that it comprises:

Obtain the code stream, and obtain the target delay T, and obtain the number of consecutive packet losses B and the number of burst packet losses N within the target delay T; the target delay T is the time required in the data transmission process Extend

The code stream is decoded according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain decoded data.
The method according to claim 11, wherein the code stream is decoded according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain the decoded Data, including:

Determine k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords after decoding, and the n is the number of codewords before decoding;

Perform decoding according to the k and n to-be-decoded data to obtain the decoded data.
The method according to claim 12, wherein the determining k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N comprises:

Determine the value range of the channel coding efficiency R according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N;

The capacity C is determined according to the channel coding efficiency R, and the k and n are determined according to the capacity C.
The method according to any one of claims 11-13, wherein the obtaining the target delay T comprises:

The time delay received from the encoding device is determined as the target time delay T.
The method according to any one of claims 11-13, wherein the obtaining the target time delay T comprises:

The time delay required by the decoding device U is determined as the target time delay T.
The method according to claim 15, wherein the delay required by the decoding device U is determined as the target delay T, after which the method further comprises:

Send the target time delay T to the encoding device.
The method according to any one of claims 11-13, wherein the obtaining the target time delay T comprises:

Sending a reference delay to the encoding device, where the reference delay is the delay required by the decoding device U;

Receive a target delay T sent by the encoding device, where the target delay T is obtained by the encoding device according to the reference delays of multiple decoding devices, and the multiple decoding devices include the decoding device U.
The method according to any one of claims 11-17, wherein the obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T comprises:

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the encoding device;

The number of consecutive packet losses B is determined according to the multiple historical numbers of consecutive packet losses, and the number N of burst packet losses is determined according to the multiple historical numbers of burst packet losses.
The method according to claim 18, wherein the number of consecutive packet losses B is determined according to the multiple historical numbers of consecutive packet losses, and the sudden number of packet losses is determined according to the multiple historical burst packet losses. After sending the number of lost packets N, the method further includes:

Send the number of consecutive packet losses B and the number of burst packet losses N to the encoding device.
The method according to any one of claims 11-17, wherein the obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T comprises:

Send the reference continuous packet loss number and the reference burst packet loss number to the encoding device, where the reference continuous packet loss number is obtained by the decoding device U according to multiple historical continuous packet loss numbers, and the reference burst packet loss number is Obtained by the decoding device U according to multiple historical burst packet loss numbers;

Receive the number of consecutive packet losses B and the number of burst packet losses N sent by the encoding device, where the number of consecutive packet losses B is obtained by the encoding device according to the reference number of consecutive packet losses of multiple decoding devices, and The burst packet loss numbers N are respectively obtained by the encoding device according to the reference burst packet loss numbers of multiple decoding devices, and the multiple decoding devices include the decoding device U.
The method according to any one of claims 17-20, wherein the multiple decoding devices are:

Multiple decoding devices that use the same transmission mode to obtain the code stream, or;

Multiple decoding devices processing the same type of service, or;

Multiple decoding devices with the same required quality of service QoS level or the same quality of experience QoE.
An encoding device, characterized in that it comprises:

The acquiring unit is used to acquire the data to be encoded, and acquire the target time delay T and the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T; the target time delay T is in the data transmission process The delay required in the process;

The encoding unit is configured to encode the data to be encoded according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N to obtain a code stream.
The encoding device according to claim 21, wherein the encoding unit is specifically configured to:

Determine k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords before encoding, and the n is the number of codewords after encoding;

Perform encoding according to the k and n to-be-encoded data to obtain the code stream.
The encoding device according to claim 23, wherein in the aspect of determining k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, the The coding unit is specifically used for:

Determine the value range of the channel coding efficiency R according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N;

The capacity C is determined according to the channel coding efficiency R, and the k and n are determined according to the capacity C.
The encoding device according to any one of claims 22-24, wherein, in terms of acquiring the target time delay T, the acquiring unit is specifically configured to:

Obtain the service type corresponding to the data to be encoded, and determine the target delay T according to the service type corresponding to the data to be encoded.
The encoding device according to any one of claims 22-24, wherein in the aspect of acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the acquiring unit Specifically used for:

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the decoding device U, and determine the consecutive packet losses B according to the multiple historical consecutive packet losses, And determining the burst packet loss number N according to the multiple historical burst packet loss numbers, and the decoding device U is any one of the decoding devices that receive the data to be encoded; or;

Acquiring multiple historical continuous packet loss numbers and multiple historical burst packet loss numbers used in the process of data transmission with multiple decoding devices, and determining the continuous packet loss number B according to the multiple historical continuous packet loss numbers, And determining the burst packet loss number N according to a plurality of historical burst packet loss numbers; the plurality of decoding devices are all decoding devices that receive the data to be encoded.
The encoding device according to any one of claims 22-24, wherein, in terms of acquiring the target time delay T, the acquiring unit is specifically configured to:

Receiving reference delays sent by multiple decoding devices, where the reference delay of the decoding device U is the delay required by the decoding device U;

Determine the target delay T according to the reference delay sent by the multiple decoding apparatuses;

Wherein, the decoding device U is any one of the plurality of decoding devices.
The encoding device according to claim 27, wherein the encoding device further comprises:

The sending unit is configured to, after the acquiring unit acquires the target delay T, send the target delay T to the plurality of decoding devices by broadcasting, or; send the target delay T to all the decoding devices by unicast The multiple decoding devices.
The encoding device according to any one of claims 22-24, wherein in the aspect of acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the acquiring unit Specifically used for:

Acquiring the reference continuous packet loss number and the reference burst packet loss number sent by multiple decoding devices;

The number of consecutive packet losses B is determined according to the reference number of consecutive packet losses sent by the multiple decoding apparatuses, and the number of burst packet losses N is determined according to the reference burst packet losses sent by the multiple decoding apparatuses.
The encoding device according to claim 29, wherein the encoding device further comprises:

The sending unit is configured to, after obtaining the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, send the number of consecutive packet losses B and the number of burst packet losses N to The multiple decoding devices, or;

And configured to send the number of consecutive packet losses B and the number of burst packet losses N to the plurality of decoding apparatuses through unicast.
The encoding device according to any one of claims 26-30, wherein the plurality of decoding devices are:

Multiple decoding devices that use the same transmission mode for receiving the data to be encoded, or;

Multiple decoding devices processing the same type of service, or;

Multiple decoding devices with the same required quality of service QoS level or the same quality of experience QoE.
A decoding device, characterized in that it comprises:

The acquiring unit is used to acquire the code stream and acquire the target delay T, and acquire the number of consecutive packet losses B and the number of burst packet losses N within the target delay T; the target delay T is the data transmission The delay required in the process;

The decoding unit is configured to decode the code stream according to the target time delay T, the number of consecutive packet losses B and the number of burst packet losses N to obtain decoded data.
The decoding device according to claim 32, wherein the decoding unit is specifically configured to:

Determine k and n according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N, where k is the number of codewords after decoding, and the n is the number of codewords before decoding;

Perform decoding according to the k and n to-be-decoded data to obtain the decoded data.
The decoding device according to claim 33, wherein, in the aspect of determining k and n according to the target time delay T, the number of consecutive packet losses B, and the number of burst packet losses N, the The decoding unit is specifically used for:

Determine the value range of the channel coding efficiency R according to the target delay T, the number of consecutive packet losses B, and the number of burst packet losses N;

The capacity C is determined according to the channel coding efficiency R, and the k and n are determined according to the capacity C.
The decoding device according to any one of claims 32-34, wherein, in terms of obtaining the target time delay T, the obtaining unit is specifically configured to:

The time delay received from the encoding device is determined as the target time delay T.
The decoding device according to any one of claims 32-34, wherein, in terms of obtaining the target time delay T, the obtaining unit is specifically configured to:

The time delay required by the decoding device is determined as the target time delay T.
The decoding device according to claim 36, wherein the decoding device further comprises:

The first sending unit is configured to send the target delay T to the encoding device after the acquiring unit determines the delay required by the decoding device as the target delay T.
The decoding device according to any one of claims 32-34, wherein, in terms of obtaining the target time delay T, the obtaining unit is specifically configured to:

Sending a reference delay to the encoding device, where the reference delay is the delay required by the decoding device;

Receive a target delay T sent by the encoding device, where the target delay T is obtained by the encoding device according to the reference delays of multiple decoding devices, and the multiple decoding devices include the decoding device.
The decoding device according to any one of claims 32-38, wherein in the aspect of acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the acquiring unit Specifically used for:

Obtain multiple historical consecutive packet losses and multiple historical burst packet losses used in the process of data transmission with the encoding device;

The number of consecutive packet losses B is determined according to the multiple historical numbers of consecutive packet losses, and the number N of burst packet losses is determined according to the multiple historical numbers of burst packet losses.
The decoding device according to claim 39, wherein the decoding device further comprises:

The second sending unit is configured to determine, at the acquiring unit, the number of consecutive packet losses B according to the multiple historical consecutive packet losses, and determine the burst packet loss according to the multiple historical burst packet losses After the number N, send the number of consecutive packet losses B and the number of burst packet losses N to the encoding device.
The decoding device according to any one of claims 32-38, wherein in the aspect of acquiring the number of consecutive packet losses B and the number of burst packet losses N within the target time delay T, the acquiring unit Specifically used for:

Send the reference continuous packet loss number and the reference burst packet loss number to the encoding device, where the reference continuous packet loss number is obtained by the decoding device according to multiple historical continuous packet loss numbers, and the reference burst packet loss number is the Obtained by the decoding device based on multiple historical burst packet loss numbers;

Receive the number of consecutive packet losses B and the number of burst packet losses N sent by the encoding device, where the number of consecutive packet losses B is obtained by the encoding device according to the reference number of consecutive packet losses of multiple decoding devices, and The burst packet loss number N is obtained by the encoding device according to the reference burst packet loss numbers of multiple decoding devices, and the multiple decoding devices include the decoding device.
The decoding device according to any one of claims 39-41, wherein the plurality of decoding devices are:

Multiple decoding devices that use the same transmission mode to obtain the code stream, or;

Multiple decoding devices processing the same type of service, or;

Multiple decoding devices with the same required quality of service QoS level or the same quality of experience QoE.
An encoding device, characterized in that it comprises:

Memory

A processor coupled with the storage, and the processor executes instructions stored in the storage to implement the method according to any one of claims 1-10.
A decoding device, characterized in that it comprises:

Memory

A processor coupled to the storage, and the processor executes instructions stored in the storage to implement the method according to any one of claims 11-21.
A data encoding and decoding system. The data encoding and decoding system includes an encoding device and a plurality of decoding devices, and is characterized in that it includes:

The encoding device is configured to execute the method according to any one of claims 1-10;

Any one of the plurality of decoding devices is configured to execute the method according to any one of claims 11-21.