CN101123606A - AVS transmission control method based on real time transmission protocol or real time control protocol - Google Patents

AVS transmission control method based on real time transmission protocol or real time control protocol Download PDF

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CN101123606A
CN101123606A CNA2007100437530A CN200710043753A CN101123606A CN 101123606 A CN101123606 A CN 101123606A CN A2007100437530 A CNA2007100437530 A CN A2007100437530A CN 200710043753 A CN200710043753 A CN 200710043753A CN 101123606 A CN101123606 A CN 101123606A
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滕国伟
张倩
方勇
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Central Academy of SVA Group Co Ltd
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Abstract

The invention provides an AVS transmission control method, which is based on a real time transmission protocol or a real time control protocol. The invention utilizes an auto-adaptive method to feed back the network status, thereby adjusting the transmission rate, preventing the network congestion, inhibiting the buffer overflowing, ensuring the video transmission service quality and realizing the end to end flow capacity control.

Description

AVS transmission control method based on real-time transmission protocol or real-time control protocol
Technical Field
The invention relates to an AVS (Advanced audio video coding standard) transmission Control method based on a Real-time transmission Protocol or a Real-time Control Protocol, in particular to an SR (sender message) and RR (receiver message) message designed by using an RTP (Real-time Transport Protocol) or RTCP (Real-time Control Protocol) Protocol to provide transmission Control.
Background
In modern society, the amount of information that people need to transmit and process is increasing, and the variety of information is also increasing, wherein the demand for broadband new services such as video conferencing, high-speed data transmission, distance teaching, video on demand, etc. is rapidly increasing. It is desirable to be able to deliver a variety of services using an efficient video compression and transmission protocol.
The AVS in our country has been receiving more and more attention due to a series of technologies to achieve high efficiency video coding, including intra prediction, inter prediction, transformation, quantization, entropy coding, and so on. AVS becomes an important technology for new generation video compression, which can provide unprecedented scalability and cost performance, as well as support for future real-time services, multimedia services, and the like. In future information systems, AVS will play an important role.
Disclosure of Invention
The AVS transmission control method based on the real-time transmission protocol or the real-time control protocol utilizes the self-adaptive method to feed back the network condition, thereby adjusting the transmission rate, preventing the network congestion, inhibiting the buffer overflow, ensuring the service quality of video transmission and realizing the end-to-end flow control.
In order to achieve the above object, the present invention provides an AVS transmission control method based on real-time transmission protocol or real-time control protocol, comprising the following steps:
step 1, a sending end sends data;
step 1.1, a sending end encodes a code stream by utilizing an AVS encoder;
step 1.2, packaging the coded data according to a packaging rule formulated by an AVS standard;
(1) When the Video _ Sequence _ Header of the AVS appears, it always appears at the beginning of an RTP payload (payload refers to the image integrity data);
(2) When the Picture _ Header (i _ Picture _ Header or pb _ Picture _ Header) Picture Header of AVS appears, it always appears at the beginning of the RTP payload, or follows the Sequence Header (Video Sequence Header);
(3) Scalable RTP payload format type. Providing priority level reference according to the importance degree of media content, dynamically adjusting the sending content and the receiving strategy to adapt to the dynamic change of network resources (such as insufficient bandwidth), and when the network resources are in shortage, preferentially transmitting key media data as much as possible, such as RTP payload containing a sequence header and a picture header;
(4) Each media stream data unit header (e.g., sequence _ header and picture _ header) must be completely contained within one RTP packet. Otherwise, there is no other limit to the position of the data unit header in the RTP packet payload;
(5) A plurality of video packets can be loaded in one RTP packet;
step 1.3, sending data, wherein the sent data comprises an RTP data packet and a data control packet part of RTCP;
step 2, the receiving end receives data, and the received data comprises an RTP data packet and a data control packet part of RTCP;
step 2.1, the data packet reaches a buffer of a receiving end, and the buffer caches the data;
the processing service time of the data packet is independent and equally distributed random variable, and the distribution is usually set as exponential distribution;
step 2.1.1, when the first data packet completely reaches the buffer, initializing parameters, and setting the idle size cv of the buffer as the buffer size buffer: cv = buffer;
step 2.1.2, the buffer judges whether the serial number of the received data packet is the serial number of the previous data packet plus 1, if yes, the step 2.1.3 is continued, otherwise, the data packet is discarded, and the next data packet is continuously waited to be received;
step 2.1.3, judging whether the size of the buffer area meets the requirement, namely whether the sum of the idle size of the buffer area and the output data value is more than or equal to the input data value, if so, performing step 2.1.4, and if not, performing step 2.1.5;
step 2.1.4, judging whether the data overflow or not when the buffer area has the idle size equal to the idle size of the original buffer area plus the output-input difference value, if so, performing step 2.1.6, and if not, finishing the caching operation of a data packet;
step 2.1.5, making a mark that the buffer area works abnormally to ensure that a subsequent event can not be triggered normally, reducing the input data volume and avoiding continuously losing data packets;
step 2.1.6, enabling cv = buffer, and marking the buffer zone to work abnormally so that the subsequent events cannot be triggered normally;
step 2.2, unpacking the coded data according to an unpacking rule formulated by an AVS standard, namely taking out the data in the data packet and an image header part to form a code stream again;
step 2.3, decoding and playing the data;
step 3, a parameter counting module at a receiving end counts related parameters at any time and calculates the network bandwidth in real time;
step 3.1, the size of the data packet, bao _ size, the time T _ S when the sending end sends the message, the time T _ R when the receiving end receives the message,
step 3.2, calculating the round trip time RTT of the network:
RTT=T_R-T_S (1)
step 3.3, calculating packet loss rate Ploss:
Ploss=Pn/256 (2)
pn is the packet loss rate field in the message (this value can be directly derived from RTCP (transport control message) messages);
step 3.4, predicting the network bandwidth r:
Figure A20071004375300071
bao _ size is the size of the packet;
RTO is the retransmission interval, RTO =4 RTT;
step 4, feeding back the predicted network bandwidth obtained in the step 3 to a sending end at the same time interval;
the algorithm for the time interval t is as follows:
n is the number of the client, A is the data flow (the value can be directly obtained by an RTCP (transport control message), L is the size of an RTCP packet, alpha is a bandwidth allocation factor, the flow of the control data packet does not exceed five percent of the data flow (the value is a value specified in an RTCP protocol), the client is allocated to the bandwidth of alpha/1 + alpha, and the server is allocated to the bandwidth of 1/1+ alpha (the adjustment range of the bandwidth allocation factor alpha is 0 to 1);
step 5, the self-adaptive adjusting module of the sending end adjusts the sending rate according to the fed back network bandwidth r so as to adapt to the current network condition:
rate = (1-beta) × last predicted network bandwidth + beta × current predicted network bandwidth (5)
Beta is a smoothing factor, and the self-adjustable range is 0 to 1 in the experiment;
the AVS transmission control method based on the real-time transmission protocol or the real-time control protocol utilizes the self-adaptive method to feed back the network condition, thereby adjusting the transmission rate, preventing network congestion, inhibiting the overflow of a buffer area, ensuring the service quality of video transmission and realizing the end-to-end flow control.
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FIG. 1 is a flow chart of an AVS transmission control method based on a real-time transport protocol or a real-time control protocol according to the present invention;
fig. 2 is a flowchart of step 2.1 of an AVS transmission control method based on a real-time transport protocol or a real-time control protocol according to the present invention.
Detailed Description
The preferred embodiment of the present invention is described in detail below with reference to fig. 1 and 2:
as shown in fig. 1, the present invention provides an AVS transmission control method based on real-time transport protocol or real-time control protocol, comprising the following steps:
step 1, a sending end sends data, and the size of a data packet is (1000, 1500);
step 1.1, a sending end encodes a code stream by utilizing an AVS encoder;
step 1.2, packaging the encoded data according to a packaging rule formulated by an AVS standard;
step 1.3, sending data;
step 2, the receiving end receives data;
as shown in fig. 2, step 2.1, the data packet arrives at the buffer of the receiving end, and the buffer caches the data;
the service time for processing the data packets is independent and equally distributed random variables, the distribution is usually set as exponential distribution, and the data packets are supposed to receive services according to the arrival sequence, namely, FCFS (First com First reverse) services;
step 2.1.1, when the first data packet completely reaches the buffer, initializing parameters, and setting the idle size cv of the buffer as the buffer size buffer: cv = buffer;
step 2.1.2, the buffer judges whether the serial number of the received data packet is the serial number of the previous data packet plus 1, if yes, the step 2.1.3 is continued, otherwise, the data packet is discarded, and the next data packet is continuously waited to be received;
step 2.1.3, judging whether the size of the buffer area meets the requirement, namely whether the sum of the idle size of the buffer area and the output data value is more than or equal to the input data value, if so, performing step 2.1.4, and if not, performing step 2.1.5;
step 2.1.4, judging whether the data overflow or not when the buffer area has the idle size equal to the idle size of the original buffer area plus the output-input difference value, if so, performing step 2.1.6, and if not, finishing the caching operation of a data packet;
step 2.1.5, making abnormal marks of the buffer area to ensure that the subsequent events can not be triggered normally, reducing the input data volume and avoiding continuously losing data packets;
step 2.1.6, enabling cv = buffer, and marking the buffer zone to work abnormally so that the subsequent events cannot be triggered normally;
step 2.2, unpacking the coded data according to an unpacking rule formulated by an AVS standard;
step 2.3, playing the decoded data;
step 3, a parameter counting module of the receiving end counts related parameters at any time and calculates the network bandwidth in real time;
step 3.1, the size bao size of the data packet is counted, the time T _ S of sending the message by the sending end, the time T _ R of receiving the message by the receiving end,
step 3.2, calculating the round trip time RTT of the network:
RTT=T_R-T_S (1)
step 3.3, calculating packet loss rate Ploss:
Ploss=Pn/256 (2)
pn is the packet loss rate field in the message (this value can be directly derived from RTCP (transport control message) messages);
step 3.4, predicting the network bandwidth r:
bao _ size is the size of the packet;
RTO is the retransmission interval, RTO =4 RTT;
step 4, feeding back the predicted network bandwidth obtained in the step 3 to a sending end at the same time interval;
the algorithm for the time interval t is as follows:
Figure A20071004375300102
n is the number of the client, A is the data flow (the value can be directly obtained by an RTCP (transport control message), L is the size of an RTCP packet, alpha is a bandwidth allocation factor, the flow of the control data packet does not exceed five percent of the data flow (the value is a value specified in an RTCP protocol), the client is allocated to the bandwidth of alpha/1 + alpha, and the server is allocated to the bandwidth of 1/1+ alpha (the adjustment range of the bandwidth allocation factor alpha is 0 to 1);
step 5, the self-adaptive adjusting module of the sending end adjusts the sending rate according to the fed back network bandwidth r so as to adapt to the current network condition:
rate = (1-beta) × last predicted network bandwidth + beta × current predicted network bandwidth (5)
Beta is a smoothing factor, and the self-adjustable range is 0 to 1 in the experiment;
the AVS transmission control method based on the real-time transmission protocol or the real-time control protocol utilizes the self-adaptive method to feed back the network condition, thereby adjusting the transmission rate, preventing the network congestion, inhibiting the buffer overflow, ensuring the service quality of video transmission and realizing the end-to-end flow control.

Claims (10)

1. An AVS transmission control method based on a real-time transmission protocol or a real-time control protocol is characterized by comprising the following steps:
step 1, a sending end sends data;
step 2, the receiving end receives data;
step 3, a parameter counting module at a receiving end counts related parameters at any time and calculates the network bandwidth in real time;
step 4, feeding back the predicted network bandwidth obtained in the step 3 to a sending end at the same time interval;
step 5, the self-adaptive adjusting module of the sending end adjusts the sending rate according to the fed back network bandwidth r to adapt to the current network condition:
rate = (1- β) × last predicted network bandwidth + β × current predicted network bandwidth (5) β is a smoothing factor.
2. The AVS transmission control method based on real-time transport protocol or real-time control protocol of claim 1, wherein the step 1 comprises the steps of:
step 1.1, a sending end encodes a code stream by utilizing an AVS encoder;
step 1.2, packaging the encoded data according to a packaging rule formulated by an AVS standard;
and 1.3, sending data.
3. The AVS transmission control method based on real-time transport protocol or real-time control protocol according to claim 2, wherein the step 1.2 comprises:
(1) When the Video _ Sequence _ Header of the AVS appears, it always appears at the beginning of an RTP payload;
(2) When the Picture _ Header Picture Header of the AVS appears, it always appears at the beginning of the RTP payload, or follows the sequence Header;
(3) Scalable RTP payload format type. Providing priority level reference according to the importance degree of media content, dynamically adjusting the sending content and the receiving strategy to adapt to the dynamic change of network resources, such as insufficient bandwidth, and preferentially transmitting key media data when the network resources are in shortage, such as RTP payload containing a sequence header and a picture header;
(4) Each media stream data unit header, such as sequence _ header and picture _ header, must be completely contained in one RTP packet, otherwise, there is no other limitation on the position of the data unit header in the RTP packet payload;
(5) A plurality of video packets can be loaded in one RTP packet.
4. The AVS transmission control method based on real-time transport protocol or real-time control protocol of claim 1, wherein the step 2 comprises the steps of:
step 2.1, the data packet reaches a buffer of a receiving end, and the buffer caches the data;
step 2.2, unpacking the coded data according to unpacking rules formulated by AVS standards;
and 2.3, decoding the data and then playing the data.
5. The AVS transmission control method based on real-time transport protocol or real-time control protocol of claim 4, characterized in that the step 2.1 comprises the following steps:
step 2.1.1, when the first data packet completely reaches the buffer, initializing parameters, and setting the idle size cv of the buffer as the buffer size buffer: cv = buffer;
step 2.1.2, the buffer judges whether the serial number of the received data packet is the serial number of the previous data packet plus 1, if yes, the step 2.1.3 is continued, otherwise, the data packet is discarded, and the next data packet is continuously waited to be received;
step 2.1.3, judging whether the size of the buffer area meets the requirement, namely whether the sum of the idle size of the buffer area and the output data value is more than or equal to the input data value, if so, performing step 2.1.4, and if not, performing step 2.1.5;
step 2.1.4, the idle size of the buffer area is equal to the idle size of the original buffer area plus the output-input difference value, whether the data overflow is caused or not is judged due to the reasons of network jitter and the like, if so, the step 2.1.6 is carried out, and if not, the caching operation of a data packet is finished;
step 2.1.5, making a mark that the buffer area works abnormally to ensure that a subsequent event can not be triggered normally, reducing the input data volume and avoiding continuously losing data packets;
step 2.1.6, let cv = buffer, mark buffer as not working properly so that subsequent events cannot be triggered properly.
6. The method of claim 4, wherein the service time of processing the packet in step 2 is an exponential distribution, and is an independent and equally distributed random variable.
7. The AVS transmission control method based on the real-time transport protocol or real-time control protocol as claimed in claim 4, wherein in step 2, the data packets are served in the order of arrival, i.e. first come first served.
8. The AVS transmission control method based on real-time transport protocol or real-time control protocol of claim 1, wherein the step 3 comprises the steps of:
step 3.1, the size bao size of the data packet is counted, the time T _ S of sending the message by the sending end, the time T _ R of receiving the message by the receiving end,
step 3.2, calculating the round trip time RTT of the network:
RTT=T_R-T_S (1)
step 3.3, calculating packet loss rate Ploss:
Ploss=Pn/256 (2)
pn is the packet loss rate field in the message, which value is directly obtained from RTCP, the message);
step 3.4, predicting the network bandwidth r:
Figure A2007100437530004C1
bao _ size is the size of the packet;
RTO is the retransmission interval, RTO =4RTT.
9. The AVS transmission control method based on real-time transport protocol or real-time control protocol according to claim 1, wherein the algorithm of the time interval t in step 4 is as follows:
Figure A2007100437530004C2
n is the number of the client sides, A is the data flow, L is the size of the RTCP packet, alpha is the bandwidth allocation factor, the flow of the control data packet is not more than five percent of the data flow, the client side allocates the bandwidth of alpha/1 + alpha, the server allocates the bandwidth of 1/1+ alpha, and the adjustment range of the bandwidth allocation factor alpha is 0 to 1.
10. The AVS transmission control method based on real-time transport protocol or real-time control protocol according to claim 1, wherein the value range of the smoothing factor β in the step 5 is 0 to 1.
CNA2007100437530A 2007-07-13 2007-07-13 AVS transmission control method based on real time transmission protocol or real time control protocol Pending CN101123606A (en)

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