技术领域technical field
本发明涉及一种AVS流媒体传输控制方法，具体是一种利用RTP/RTCP协议设计的实时视频传输速率控制机制。The invention relates to an AVS stream media transmission control method, in particular to a real-time video transmission rate control mechanism designed using the RTP/RTCP protocol.
背景技术Background technique
随着网络的迅速增长和普及，人们对多媒体信息的需求也越来越大，网络传输实时流媒体应运而生，传统的TCP/IP协议的传送机制和拥塞控制都不适合网络多媒体的实时传输，UDP协议本身又不提供任何Qos保证，因此，实时传输协议族RTP/RTCP[1]被广泛用于各种多媒体传输系统中来提供数据实时传输和Qos服务以满足音视频数据实时传输的要求。在多媒体领域中视频占有很大一部分内容，因此视频压缩成为多媒体领域很重要的一部分。With the rapid growth and popularity of the network, people's demand for multimedia information is also increasing, and real-time streaming media for network transmission emerges as the times require. The traditional TCP/IP protocol transmission mechanism and congestion control are not suitable for real-time transmission of network multimedia. , the UDP protocol itself does not provide any Qos guarantee, therefore, the real-time transport protocol family RTP/RTCP [1] is widely used in various multimedia transmission systems to provide real-time data transmission and Qos services to meet the requirements of real-time transmission of audio and video data . In the multimedia field, video occupies a large part of the content, so video compression has become a very important part of the multimedia field.
目前已出现新的视音频编码标准，在国际上有H.264，它具有较高的压缩效率，相比于其它编码标准可显著提高压缩效率，目前已经在各领域获得广泛应用。在国内正在制定先进音视频编解码标准AVS(Advanced audio video coding standard)，它采用了一系列技术来达到高效率的视频编码，包括帧内预测、帧间预测、变换、量化和熵编码等，是我国具有自主知识产权的先进音视频编解码标准，目前被广泛应用到数字电视领域。因此，随着目前网上流媒体服务的进一步发展，如何在IP网上实时传输AVS视频流对于IPTV，对于视频点播系统的应用有重要意义，正受到越来越多的重视。自适应方法的核心是网络状况标志量的计算，以丢包率或时间标志量作为网络状况的标志量，是目前多数算法实现此模块的方式。通过RTCP的SR和RR报文中有接收端向发送端报告最近一段时间包丢失率的字段或网络包忘返延迟时间，反映了目前的网络状况。但是这种计算都未能考虑到接收端缓存区的状况，在实际应用中由于解码器从缓冲区中解码的速率是不固定的，如果发送端发送速度过快的话有可能会导致接收端缓存区的上溢，反之，则会发生下溢。排队论(queueing theory)，或称随机服务系统理论，作为运筹学研究的一种有力手段，排队论理论在计算机网络中占有重要的地位，运用排队论对计算机网络进行性能预测和评价往往可以得到较为切合实际的指标，对网络设备的选型具有十分重要的理论指导意义。At present, new video and audio coding standards have emerged. There is H.264 in the world. It has high compression efficiency and can significantly improve compression efficiency compared with other coding standards. It has been widely used in various fields. The advanced audio and video coding standard AVS (Advanced audio video coding standard) is being formulated in China, which uses a series of technologies to achieve high-efficiency video coding, including intra prediction, inter prediction, transformation, quantization and entropy coding, etc. It is an advanced audio and video codec standard with independent intellectual property rights in my country, and is currently widely used in the field of digital TV. Therefore, with the further development of streaming media services on the Internet, how to transmit AVS video streams in real time on the IP network is of great significance to the application of IPTV and video-on-demand systems, and is receiving more and more attention. The core of the self-adaptive method is the calculation of the network status indicator. Taking the packet loss rate or the time indicator as the indicator of the network status is the way most algorithms implement this module at present. In the SR and RR messages of RTCP, there is a field in which the receiving end reports the packet loss rate in the latest period to the sending end or the network packet forget-to-return delay time, which reflects the current network status. However, this kind of calculation fails to take into account the status of the receiving end buffer area. In practical applications, since the rate at which the decoder decodes from the buffer area is not fixed, if the sending end sends too fast, it may cause the receiving end to buffer Area overflow, otherwise, underflow will occur. Queuing theory, or stochastic service system theory, as a powerful means of operations research research, queuing theory occupies an important position in computer networks, using queuing theory to predict and evaluate the performance of computer networks can often get More practical indicators have very important theoretical guiding significance for the selection of network equipment.
发明内容Contents of the invention
本发明的目的在于针对已有技术存在的缺陷，提供一种AVS流媒体传输控制方法，它是应用排队论对接受端缓冲区进行设计并利用RTCP传输协议的SR和RR报文的一种传输控制方法，抑制了缓冲区溢出，保证视频传输的服务质量。The purpose of the present invention is to provide a kind of AVS stream media transmission control method for the defect that prior art exists, and it is to apply queuing theory to design the receiver buffer zone and utilize a kind of transmission of SR and RR message of RTCP transmission protocol The control method suppresses buffer overflow and ensures the service quality of video transmission.
为达到上述目的，本发明采用下述技术方案：To achieve the above object, the present invention adopts the following technical solutions:
一种AVS流媒体传输控制方法，其特征在于通过在接受端对缓冲器输出速率进行统计，估计出发送端的发送速率，然后通过RTCP反馈到发送端；在发送端把预测的传输速率与所述的接受端反馈的发送速率进行比较来调整传输速率。A kind of AVS streaming media transmission control method, it is characterized in that by carrying out statistics to buffer output rate at receiving end, estimate the sending rate of sending end, then feed back to sending end by RTCP; Compared with the sending rate fed back by the receiving end to adjust the transmission rate.
本方法中的缓冲器排队设计用N策略的M/M/1休假模型，在缓冲器中实现数据的排队，所有信元都经过这一个缓冲器进行缓存。The buffer queuing design in this method uses the M/M/1 vacation model of the N strategy to implement data queuing in the buffer, and all cells are buffered through this buffer.
控制步骤如下：The control steps are as follows:
1)假设单个数据包的到达的平均时间间隔λ，及解码器和播放器的平均解码播放每帧的时间TS，当总的数据包有一帧时送入解码器解码再送入播放器进行播放，把接受端看作是N策略的M/M/1休假模型，计算出缓冲区的初始值；1) Assuming the average arrival time interval λ of a single data packet, and the average decoding and playback time TS of each frame of the decoder and player, when the total data packet has one frame, it is sent to the decoder for decoding and then sent to the player for playback , regard the receiving end as the M/M/1 vacation model of the N strategy, and calculate the initial value of the buffer;
数据包按照速率为λ的泊松过程到达，数据包的服务时间是独立同分布的随机变量，通常分布设为指数分布；假设顾客按照到达的顺序接受服务，即FCFS服务；The data packet arrives according to the Poisson process with a rate of λ, and the service time of the data packet is an independent and identically distributed random variable, usually distributed as an exponential distribution; assuming that customers receive services in the order of arrival, that is, FCFS service;
2)为了更好的跟踪当前客户端的服务情况，我们通过自适应反馈来跟踪客户端的服务速率以及网络状态，2) In order to better track the service status of the current client, we use adaptive feedback to track the service rate and network status of the client.
(1)接受端生成发送端的预测发送速率u，并反馈到发送端(1) The receiving end generates the predicted sending rate u of the sending end, and feeds back to the sending end
(1.1)计算当前接收帧L的RTP数据包数量N(L)；(1.1) Calculate the RTP packet quantity N (L) of the current received frame L;
(1.2)统计解码此帧需要的时间T(L)；(1.2) Statistically decode the time T(L) required for this frame;
(1.3)计算第L帧平均每个数据包的实际发送速率V*(L)；(1.3) calculate the average actual sending rate V* (L) of each data packet in the L frame;
V*(L)＝N(L)/T(L)V* (L)＝N(L)/T(L)
为了防止输出速率的随机抖动，所以用当前帧预测发送速率和当前帧实际发送速率来共同预测下一帧的预测发送速率(取两者间的平滑因子α为0.75，L≥1，V(1)＝V*(1)：In order to prevent the random jitter of the output rate, the predicted sending rate of the current frame and the actual sending rate of the current frame are used to jointly predict the predicted sending rate of the next frame (take the smoothing factor α between the two as 0.75, L≥1, V(1 )=V* (1):
V(L+1)＝(1-α)V(L)+αV*(L)V(L+1)＝(1-α)V(L)+αV* (L)
(1.4)将V(L+1)作为发送端的预测发送速率u，通过RTCP报文反馈到发送端，这样就可以随时利用RTCP反馈信息来快速追踪网络变化；(1.4) Use V(L+1) as the predicted transmission rate u of the sender, and feed it back to the sender through the RTCP message, so that the RTCP feedback information can be used at any time to quickly track network changes;
(2)发送端预测网络带宽(2) The sender predicts the network bandwidth
接收客户端反馈来的RR和RTCP报文并解析，获取往返时间、网络延时、丢包率和网络带宽，分析得出当前网络状况Receive and analyze the RR and RTCP messages fed back by the client, obtain the round-trip time, network delay, packet loss rate and network bandwidth, and analyze the current network status
设SSRC_R为产生该接收报告的数据接收端，SSRC_S为数据发送端，时间TSSRC_S为发送端发送报文的时刻，TSSRC_R为接收端接收报文的时刻，可以计算出网络往返时间RTT：Let SSRC_R be the data receiving end that generates the receiving report, SSRC_S is the data sending end, the time TSSRC_S is the time when the sending end sends the message, and TSSRC_R is the time when the receiving end receives the message, the network round-trip time RTT can be calculated:
RTT＝TSSRC_R-TSSRC_SRTT＝TSSRC_R -TSSRC_S
利用RR报文中的分组丢失率字段Pn(fraction lost)，可计算出丢包率PLOSS：Using the packet loss rate field Pn(fraction lost) in the RR message, the packet loss rate PLOSS can be calculated:
PLOSS＝Pn/256PLOSS = Pn /256
适用于RTP环境中的传输速率计算公式预测出网络带宽r：The transmission rate calculation formula applicable to the RTP environment predicts the network bandwidth r:
其中MTU为最大传输单元的大小，为1500字节；Among them, MTU is the size of the maximum transmission unit, which is 1500 bytes;
(3)发送端根据发馈回来的输出速率及预测出的网络带宽的传输速率来调整发送速率rate：(3) The sending end adjusts the sending rate rate according to the output rate sent back and the predicted transmission rate of the network bandwidth:
rate＝min(r，u*ρ)rate=min(r,u*ρ)
u初始使值为0，ρ为正常数在实验中数据到达速率与播放速率的比值，目的是使数据包的发送速率小于播放器的速率这样使播放端缓冲区的等待队列长度趋于稳定；The initial value of u is 0, and ρ is a normal number. In the experiment, the ratio of the data arrival rate to the playback rate, the purpose is to make the sending rate of the data packet smaller than the rate of the player, so that the waiting queue length of the buffer at the playback end tends to be stable;
由上述算法可以分析得到发送速率，发送端(Server)的参数调整模块就可以调整发送速率，以适应当前网络状况。The sending rate can be obtained by analyzing the above algorithm, and the parameter adjustment module of the sending end (Server) can adjust the sending rate to adapt to the current network condition.
本发明与现有技术相比较，具有如下显而易见的突出实质性特点和显著优点：本发明是应用排队论对接受端缓冲区进行设计，并利用RTCP传输协议的SR和RR报文的一种传输控制方法，抑制了缓冲区溢出，保证视频传输的服务质量。Compared with the prior art, the present invention has the following obvious outstanding substantive features and significant advantages: the present invention uses queuing theory to design the receiving end buffer, and utilizes a kind of transmission of SR and RR messages of RTCP transmission protocol The control method suppresses buffer overflow and ensures the service quality of video transmission.
附图说明Description of drawings
图1是本发明的实时视频传输速率控制机制结构图。Fig. 1 is a structural diagram of the real-time video transmission rate control mechanism of the present invention.
具体实施方式Detailed ways
本发明的一个优选实施例结合附图详述如下：A preferred embodiment of the present invention is described in detail as follows in conjunction with accompanying drawing:
参见图1，本AVS流媒体传输控制方法是通过在接受端对缓冲器输出速率进行统计，估计出发送端的发送速率，然后通过RTCP反馈到发送端；在发送端，把预测的传输速率与所述的接受端反馈的发送率进行比较来调整传输速率。Referring to Fig. 1, this AVS streaming media transmission control method is to estimate the sending rate of the sending end by counting the output rate of the buffer at the receiving end, and then feed back to the sending end through RTCP; at the sending end, compare the predicted transmission rate with the set The transmission rate is adjusted by comparing the sending rate fed back by the receiving end.
其控制步骤如下：Its control steps are as follows:
1)假设单个数据包的到达的平均时间间隔λ，及解码器和播放器的平均解码播放每帧的时间TS，当总的数据包有一帧时送入解码器解码再送入播放器进行播放，把接受端看作是N策略的M/M/1休假模型，计算出缓冲区的初始值；1) Assuming the average arrival time interval λ of a single data packet, and the average decoding and playback time TS of each frame of the decoder and player, when the total data packet has one frame, it is sent to the decoder for decoding and then sent to the player for playback , regard the receiving end as the M/M/1 vacation model of the N strategy, and calculate the initial value of the buffer;
数据包按照速率为λ的泊松过程到达，数据包的服务时间是独立同分布的随机变量，通常分布设为指数分布；假设顾客按照到达的顺序接受服务，即FCFS服务；The data packet arrives according to the Poisson process with a rate of λ, and the service time of the data packet is an independent and identically distributed random variable, usually distributed as an exponential distribution; assuming that customers receive services in the order of arrival, that is, FCFS service;
2)为了更好的跟踪当前客户端的服务情况，我们通过自适应反馈来跟踪客户端的服务速率以及网络状态，2) In order to better track the service status of the current client, we use adaptive feedback to track the service rate and network status of the client.
(1)接受端生成发送端的预测发送速率u，并反馈到发送端(1) The receiving end generates the predicted sending rate u of the sending end, and feeds back to the sending end
(1.1)计算当前接收帧L的RTP数据包数量N(L)；(1.1) Calculate the RTP packet quantity N (L) of the current received frame L;
(1.2)统计解码此帧需要的时间T(L)；(1.2) Statistically decode the time T(L) required for this frame;
(1.3)计算第L帧平均每个数据包的实际发送速率V*(L)；(1.3) calculate the average actual sending rate V* (L) of each data packet in the L frame;
V*(L)＝N(L)/T(L)V* (L)＝N(L)/T(L)
为了防止输出速率的随机抖动，所以用当前帧预测发送速率和当前帧实际发送速率来共同预测下一帧的预测发送速率(取两者间的平滑因子α为0.75，L≥1，V(1)＝V*(1)：In order to prevent the random jitter of the output rate, the predicted sending rate of the current frame and the actual sending rate of the current frame are used to jointly predict the predicted sending rate of the next frame (take the smoothing factor α between the two as 0.75, L≥1, V(1 )=V* (1):
V(L+1)＝(1-α)V(L)+αV*(L)V(L+1)＝(1-α)V(L)+αV* (L)
(1.4)将V(L+1)作为发送端的预测发送速率u，通过RTCP报文反馈到发送端，这样就可以随时利用RTCP反馈信息来快速追踪网络变化；(1.4) Use V(L+1) as the predicted transmission rate u of the sender, and feed it back to the sender through the RTCP message, so that the RTCP feedback information can be used at any time to quickly track network changes;
(2)发送端预测网络带宽(2) The sender predicts the network bandwidth
接收客户端反馈来的RR和RTCP报文并解析，获取往返时间、网络延时、丢包率和网络带宽，分析得出当前网络状况Receive and analyze the RR and RTCP messages fed back by the client, obtain the round-trip time, network delay, packet loss rate and network bandwidth, and analyze the current network status
设SSRC_R为产生该接收报告的数据接收端，SSRC_S为数据发送端，时间TSSRC_S为发送端发送报文的时刻，TSSRC_R为接收端接收报文的时刻，可以计算出网络往返时间RTT：Let SSRC_R be the data receiving end that generates the receiving report, SSRC_S is the data sending end, the time TSSRC_S is the time when the sending end sends the message, and TSSRC_R is the time when the receiving end receives the message, the network round-trip time RTT can be calculated:
RTT＝TSSRC_R-TSSRC_SRTT＝TSSRC_R -TSSRC_S
利用RR报文中的分组丢失率字段Pn(fraction lost)，可计算出丢包率PLOSS：Using the packet loss rate field Pn(fraction lost) in the RR message, the packet loss rate PLOSS can be calculated:
PLOSS＝Pn/256PLOSS = Pn /256
适用于RTP环境中的传输速率计算公式预测出网络带宽r：The transmission rate calculation formula applicable to the RTP environment predicts the network bandwidth r:
其中MTU为最大传输单元的大小，为1500字节；Among them, MTU is the size of the maximum transmission unit, which is 1500 bytes;
(3)发送端根据发馈回来的输出速率及预测出的网络带宽的传输速率来调整发送速率rate：(3) The sending end adjusts the sending rate rate according to the output rate sent back and the predicted transmission rate of the network bandwidth:
rate＝min(r，u*ρ)rate=min(r,u*ρ)
u初始使值为0，ρ为正常数在实验中数据到达速率与播放速率的比值，目的是使数据包的发送速率小于播放器的速率这样使播放端缓冲区的等待队列长度趋于稳定；The initial value of u is 0, and ρ is a normal number. In the experiment, the ratio of the data arrival rate to the playback rate, the purpose is to make the sending rate of the data packet smaller than the rate of the player, so that the waiting queue length of the buffer at the playback end tends to be stable;
由上述算法可以分析得到发送速率，发送端(Server)的参数调整模块就可以调整发送速率，以适应当前网络状况。The sending rate can be obtained by analyzing the above algorithm, and the parameter adjustment module of the sending end (Server) can adjust the sending rate to adapt to the current network condition.
上述的服务器端可以完成视频流的RTP打包以及包分解等操作，建立和维护传输包发送和接收队列，以及记录客户端的连接情况，分析包丢失和当前网络带宽状态，QoS监视等。The above-mentioned server can complete operations such as RTP packaging and packet decomposition of video streams, establish and maintain transmission packet sending and receiving queues, record client connection status, analyze packet loss and current network bandwidth status, QoS monitoring, etc.
客户端可以完成视频流的RTP解包以及包组合等操作，建立和维护传输包接收队列，以及记录客户端与服务器的连接情况，分析缓冲区状态等。The client can complete operations such as RTP depacketization and packet combination of the video stream, establish and maintain the transmission packet receiving queue, record the connection between the client and the server, and analyze the buffer status, etc.
网络传输模块：建立和关闭网络连接socket，负责从发送包队列取数据发送和将接收数据包提交给接收队列，反映当前网络连接情况等。Network transmission module: establish and close the network connection socket, responsible for fetching data from the sending packet queue and submitting the received data packet to the receiving queue, reflecting the current network connection status, etc.
视频解码模块：采用AVS解码器，在客户端将视频流解码播放。Video decoding module: AVS decoder is used to decode and play the video stream on the client side.
参数统计模块：由于解码器和播放器从缓冲区中取出解码和播放的速率是不固定的为了更好的跟踪当前服务器的服务情况，我们通过反馈来跟踪服务器的服务速率。因此，通过在接受端的参数估计模块对缓冲器输出速率进行统计估计出发送端的发送速率，通过RTCP的扩展位反馈到发送端，在发送端用RTP自适应算法把预测的传输速率与接受端反馈的发送速率比较来调整发送速率。Parameter statistics module: Since the decoding and playback rate of the decoder and player from the buffer is not fixed, in order to better track the current server service situation, we use feedback to track the server service rate. Therefore, through the parameter estimation module at the receiving end, the output rate of the buffer is statistically estimated to estimate the sending rate at the sending end, and the extended bit of RTCP is fed back to the sending end, and the RTP adaptive algorithm is used at the sending end to compare the predicted transmission rate with the receiving end. to adjust the sending rate by comparing the sending rate.
系统首先根据解码器和播放器的共同服务速率1/TS和数据包的预期到达速率(1/TS)*ρ计算出缓冲区的初始值L。把接受端看作是N策略的M/M/1休假模型。即当N个数据包全部到达时播放器才开始工作其余时间处于休假状态。数据包按照速率为λ的泊松过程到达，数据包的服务时间是独立同分布的随机变量，通常分布设为指数分布。The system first calculates the initial value L of the buffer zone according to the common service rate 1/TS of the decoder and the player and the expected arrival rate (1/TS)*ρ of the data packet. Think of the receiving end as the M/M/1 vacation model of the N strategy. That is, the player starts to work when all the N data packets arrive, and the rest of the time is in the vacation state. The data packet arrives according to the Poisson process with rate λ, and the service time of the data packet is an independent and identically distributed random variable, usually distributed as an exponential distribution.
N策略的M/M/1休假系统中稳态队长可分解成独立随即变量之和Lv＝L+Ld，其中L是经典无休假M/M/1的稳态队长，其中：The steady-state team leader in the N-strategy M/M/1 vacation system can be decomposed into the sum of independent random variables Lv =L+Ld , where L is the steady-state team leader of the classic M/M/1 without vacation, where:
附加Ld有母函数
虽然休假长度与到达过程不独立，但分解定理结论仍然成立，直接计算给出Although the vacation length is not independent of the arrival process, the conclusion of the decomposition theorem still holds, and the direct calculation gives
其中ρ为数据到达速率与播放速率的比值，b为播放平均速率。Among them, ρ is the ratio of the data arrival rate to the playback rate, and b is the average playback rate.
当第一个帧的N个数据包全部到达，播放器开始工作，进入播放器，缓冲区空闲大小初始为缓冲区大小buffer。判断信元是否出错，若出错，则不继续进行；若不出错，则继续进行。如果缓冲区大小满足要求，即缓冲区空闲大小与输出数据值之和大于等于输入数据值，缓冲区空闲大小等于原缓冲区空闲大小加上输出输入差值。当缓冲区数据溢出时，作缓冲区工作不正常的标记使得后续事件不能正常触发。如果缓冲区大小不满足要求，即缓冲区空闲大小与输出数据值之和小于输入数据值，就会出现数据丢失，作缓冲区工作不正常的标记使得后续事件不能正常触发。When all the N data packets of the first frame arrive, the player starts to work and enters the player, and the free size of the buffer is initially the buffer size buffer. Judging whether the cell is wrong, if wrong, do not continue; if not wrong, continue. If the buffer size meets the requirements, that is, the sum of the free size of the buffer and the output data value is greater than or equal to the input data value, the free size of the buffer is equal to the free size of the original buffer plus the difference between output and input. When the buffer data overflows, mark the buffer as abnormal so that subsequent events cannot be triggered normally. If the buffer size does not meet the requirements, that is, the sum of the free size of the buffer and the output data value is less than the input data value, data loss will occur, and marking that the buffer is not working properly will prevent subsequent events from being triggered normally.
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNA2007100427505ACN101075957A (en) | 2007-06-26 | 2007-06-26 | Method for controlling AVS fluid-medium transmission |
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNA2007100427505ACN101075957A (en) | 2007-06-26 | 2007-06-26 | Method for controlling AVS fluid-medium transmission |
| Publication Number | Publication Date |
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| CN101075957Atrue CN101075957A (en) | 2007-11-21 |
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNA2007100427505APendingCN101075957A (en) | 2007-06-26 | 2007-06-26 | Method for controlling AVS fluid-medium transmission |
| Country | Link |
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| CN (1) | CN101075957A (en) |
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