CN102710374A - Speed control method in wireless streaming media transmission - Google Patents
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Abstract
本发明涉及无线多媒体通信。为减小流媒体在无线环境下传输时由于无线信道不可靠而引起的传输质量抖动,进而提高传输质量以及无线信道的利用率,本发明采取的技术方案是,无线流媒体传输中的速率控制方法,比传统速率控制方法更短的时间间隔内通过平滑处理获取丢包率和业务数据通过率参数;基于上述参数及改进的加增乘减算法调整流媒体传输速率,在信道轻载时,由丢包率决定的变常数加性因子和常量加性因子决定速率增长幅度,在信道超载时,由丢包率决定的变常数乘性因子和常量减性因子决定速率降低幅度;设定速率切换门限,只有当本次预测速率与上次预测速率的差值超过设定门限时才认为速率切换有效。本发明主要应用于无线多媒体通信。
The present invention relates to wireless multimedia communication. In order to reduce the transmission quality jitter caused by the unreliable wireless channel when the streaming media is transmitted in a wireless environment, and then improve the transmission quality and the utilization rate of the wireless channel, the technical solution adopted by the present invention is that the rate control in the wireless streaming media transmission method, obtain packet loss rate and service data pass rate parameters through smooth processing in a shorter time interval than traditional rate control methods; adjust the streaming media transmission rate based on the above parameters and the improved increase, multiply, and subtract algorithm, and when the channel is lightly loaded, The variable constant additive factor and constant additive factor determined by the packet loss rate determine the rate increase. When the channel is overloaded, the variable constant multiplicative factor and constant subtractive factor determined by the packet loss rate determine the rate reduction; set the rate Switching threshold, only when the difference between the current predicted rate and the last predicted rate exceeds the set threshold, the rate switching is considered valid. The invention is mainly applied to wireless multimedia communication.
Description
技术领域 technical field
本发明属于无线多媒体通信技术领域,涉及一种基于丢包率、业务数据通过率以及改进的加增乘减(AIMD)算法保证无线流媒体传输服务质量的速率控制方法,具体讲,涉及基于约束满足框架的空间问题求解器。The invention belongs to the technical field of wireless multimedia communication, and relates to a rate control method for guaranteeing service quality of wireless streaming media transmission based on packet loss rate, service data pass rate and an improved AIMD algorithm. Specifically, it relates to a method based on constraint A solver for spatial problems that meets the framework.
背景技术 Background technique
近年来,随着通信技术、网络技术、视频编解码技术的快速发展,网络通信的服务内容变得更加丰富多彩。结合语音、视频、数据等信息传输的多媒体通信已经成为通信业务发展的必然趋势,其中多媒体视频通信由于具有连续性传输、数据量大、对实时性、可靠性要求高等特点而成为了研究热点。In recent years, with the rapid development of communication technology, network technology, and video codec technology, the service content of network communication has become more colorful. Multimedia communication combined with information transmission such as voice, video, and data has become an inevitable trend in the development of communication services. Among them, multimedia video communication has become a research hotspot due to its characteristics of continuous transmission, large data volume, and high requirements for real-time performance and reliability.
有线多媒体通信技术已趋成熟,如基于H.261/H.263标准的视频电话、视频点播(VoD:Video on Demand)、多媒体电子邮件、有线数字电视和视频监控等技术已经获得广泛应用。然而,虽然有线信道能够为多媒体通信提供比较稳定可靠的传输环境,但是有线传输的固有弊端,如对布线环境有着较高的要求、成本较高等,限制了多媒体通信的进一步发展。基于无线传输的多媒体通信业务受到人们的关注。Wired multimedia communication technology has matured, such as video telephony based on H.261/H.263 standard, video on demand (VoD: Video on Demand), multimedia email, cable digital TV and video surveillance and other technologies have been widely used. However, although wired channels can provide a relatively stable and reliable transmission environment for multimedia communication, the inherent disadvantages of wired transmission, such as high requirements for wiring environment and high cost, limit the further development of multimedia communication. Multimedia communication services based on wireless transmission have attracted people's attention.
近十年无线传输技术的发展突飞猛进,无线网络能够提供的传输带宽也得到了飞速提升。第三代移动通信网可以为固定位置服务提供2Mbps的速率,在人行走的状态下速率能达到384Kbps,在车速行进中也可达到128Kbps,已经达到了多媒体视频传输业务的带宽要求。在第四代(4G)无线通讯系统中,传输速率能达到80Mbps以上;IEEE 802.11系列标准可提供每秒数兆至数百兆的传输速率;中国近期提出的具有自主知识产权的超高速无线局域网标准更是将物理层传输速率提升到了Gbps。无线通信带宽的快速增长,为多媒体通信提供了更好的发展机遇。然而,无线信道具有错误率高以及时变的特性,无线信道的不可靠会导致多媒体视频质量发生严重抖动,从而使得保障无线流媒体传输服务质量面临着巨大挑战。In the past ten years, the development of wireless transmission technology has advanced by leaps and bounds, and the transmission bandwidth that wireless networks can provide has also been rapidly improved. The third-generation mobile communication network can provide a rate of 2Mbps for fixed location services. The rate can reach 384Kbps when people are walking, and 128Kbps when driving at a speed of a car, which has reached the bandwidth requirements of multimedia video transmission services. In the fourth generation (4G) wireless communication system, the transmission rate can reach more than 80Mbps; IEEE 802.11 series standards can provide a transmission rate of several megabytes to hundreds of megabytes per second; China recently proposed an ultra-high-speed wireless local area network with independent intellectual property rights The standard even increases the transmission rate of the physical layer to Gbps. The rapid growth of wireless communication bandwidth provides better development opportunities for multimedia communication. However, the wireless channel has a high error rate and time-varying characteristics. The unreliability of the wireless channel will cause severe jitter in the quality of multimedia video, which makes it a huge challenge to ensure the quality of wireless streaming media transmission services.
基于上述原因,人们提出了多种速率控制方法来提高无线流媒体传输的服务质量、改善其传输效率。这些流媒体速率控制方法被广泛用于基于实时传输协议(RTP)的流媒体传输系统中。实时传输协议(RTP)是用于互联网上针对多媒体数据流的一种传输协议。RTP协议由两部分组成:RTP和RTCP(实时传输控制协议)。RTP协议位于传输层协议TCP和UDP协议之上,为了保证传输的实时性,RTP协议在负载前加上RTP包头之后,就直接将RTP包交付给UDP进行传输。RTCP主要对传输过程中的各种传输参数(丢包率、时延等)进行统计。传统的速率控制算法均是利用RTCP提供的传输参数进行速率控制。根据目前流媒体速率控制算法的速率控制原理,可以将其分为两类:Based on the above reasons, various rate control methods have been proposed to improve the service quality and transmission efficiency of wireless streaming media transmission. These streaming media rate control methods are widely used in real-time transport protocol (RTP) based streaming media transmission systems. The Real-time Transport Protocol (RTP) is a transport protocol for multimedia data streams on the Internet. The RTP protocol consists of two parts: RTP and RTCP (Real-time Transport Control Protocol). The RTP protocol is located on top of the transport layer protocols TCP and UDP. In order to ensure the real-time transmission, the RTP protocol directly delivers the RTP packet to UDP for transmission after adding the RTP packet header before the payload. RTCP mainly counts various transmission parameters (packet loss rate, delay, etc.) in the transmission process. Traditional rate control algorithms use the transmission parameters provided by RTCP for rate control. According to the rate control principle of the current streaming media rate control algorithm, it can be divided into two categories:
■加增乘减(AIMD)算法。AIMD是在TCP/IP传输模型中,为网络层提供拥塞控制的算法。该算法能为TCP/IP网络提供可靠的数据传输服务。它是通过比较实测丢包率与预设的丢包率门限值来判断网络的状态。如果丢包率大于丢包率门限,说明发生拥塞,则按照预设的常数乘性降低因子降低发送速率;反之,按照预设的常数加性增长因子增加发送速率。AIMD算法进行速率调整时并不考虑当前信道拥塞的程度与速率调整幅度的关系,是一种线性算法,计算很简单。然而,使用AIMD算法容易使传输速率发生锯齿状波动,不利于提高传输服务质量。■ Add-increase-multiply-subtract (AIMD) algorithm. AIMD is an algorithm that provides congestion control for the network layer in the TCP/IP transmission model. This algorithm can provide reliable data transmission service for TCP/IP network. It judges the state of the network by comparing the measured packet loss rate with the preset packet loss rate threshold. If the packet loss rate is greater than the packet loss rate threshold, it means that congestion occurs, and the sending rate is reduced according to the preset constant multiplicative reduction factor; otherwise, the sending rate is increased according to the preset constant additive growth factor. The AIMD algorithm does not consider the relationship between the current channel congestion level and the rate adjustment range when performing rate adjustment. It is a linear algorithm and the calculation is very simple. However, using the AIMD algorithm tends to cause saw-tooth fluctuations in the transmission rate, which is not conducive to improving the quality of transmission services.
■TCP友好速率控制(TFRC,TCP Friendly Rate Control)算法。TFRC算法基于TCP吞吐量模型来进行传输速率的调整,如公式(1)所示■TCP Friendly Rate Control (TFRC, TCP Friendly Rate Control) algorithm. The TFRC algorithm adjusts the transmission rate based on the TCP throughput model, as shown in formula (1)
其中MTU是最大传输单元;λ为丢包率,RTT是环回时间,SendRate是网络吞吐量;该算法能得到比较稳定和平滑的传输速率,然而该算法是根据网络资源使用最大化来进行速率调整,忽略了视频流的感知质量。Among them, MTU is the maximum transmission unit; λ is the packet loss rate, RTT is the loopback time, and SendRate is the network throughput; this algorithm can obtain a relatively stable and smooth transmission rate, but the algorithm is based on maximizing the use of network resources. adjustment, ignoring the perceived quality of the video stream.
上述两类算法的具有共同的特点:两类算法均是利用RTCP协议中RR(接收者报告)包所提供的传输参数统计信息来进行流媒体传输速率控制。由于RTCP协议中RR包是间隔5s进行发送,这就意味着流媒体传输系统只能每5s进行一次传输速率控制,这对于信道相对稳定的有线传输而言是足够的;然而对于带宽具有时变特性的无线传输,传输速率的控制无法适应信道的变化。The above two types of algorithms have a common feature: both types of algorithms use the transmission parameter statistics information provided by the RR (Receiver Report) package in the RTCP protocol to control the streaming media transmission rate. Since the RR packets in the RTCP protocol are sent at intervals of 5s, this means that the streaming media transmission system can only perform transmission rate control every 5s, which is sufficient for wired transmission with relatively stable channels; however, the bandwidth has time-varying Due to the characteristics of wireless transmission, the control of the transmission rate cannot adapt to the change of the channel.
发明内容 Contents of the invention
本发明旨在克服现有技术的不足,提供一种无线环境下流媒体传输的速率控制方法,以便减小流媒体在无线环境下传输时由于无线信道不可靠而引起的传输质量抖动,进而提高流媒体传输服务质量以及无线信道的利用率。为达到上述目的,本发明采取的技术方案是,无线流媒体传输中的速率控制方法,比传统速率控制方法更短的时间间隔内通过平滑处理获取丢包率和业务数据通过率参数;基于上述参数及改进的加增乘减算法调整流媒体传输速率,在信道轻载时,由丢包率决定的变常数加性因子和常量加性因子决定速率增长幅度,在信道超载时,由丢包率决定的变常数乘性因子和常量减性因子决定速率降低幅度;设定速率切换门限,只有当本次预测速率与上次预测速率的差值超过设定门限时才认为速率切换有效。The present invention aims to overcome the deficiencies of the prior art, and provides a rate control method for streaming media transmission in a wireless environment, so as to reduce the transmission quality jitter caused by unreliable wireless channels when streaming media is transmitted in a wireless environment, thereby improving the quality of streaming. Quality of service for media transmission and utilization of wireless channels. In order to achieve the above object, the technical solution adopted by the present invention is that the rate control method in the wireless streaming media transmission obtains the packet loss rate and service data pass rate parameters through smooth processing in a shorter time interval than the traditional rate control method; based on the above Parameters and an improved algorithm of adding, multiplying, and subtracting to adjust the streaming media transmission rate. When the channel is lightly loaded, the rate increase is determined by the variable constant additive factor and the constant additive factor determined by the packet loss rate. When the channel is overloaded, the rate increase is determined by the packet loss rate. The variable constant multiplicative factor and constant subtractive factor determined by the rate determine the rate reduction rate; set the rate switch threshold, and only when the difference between the current predicted rate and the last predicted rate exceeds the set threshold, the rate switch is considered valid.
包括以下步骤:Include the following steps:
(1)获取丢包率以及业务数据通过率:在流媒体传输系统的接收端,通过对接收的流媒体数据包进行统计分析来获取丢包率以及业务数据通过率参数;(1) Obtain packet loss rate and service data pass rate: at the receiving end of the streaming media transmission system, obtain the packet loss rate and service data pass rate parameters by statistically analyzing the received streaming media data packets;
(2)预测新的传输速率:流媒体传输系统的接收端获取丢包率以及业务数据通过率之后,基于改进的AIMD算法,分析无线信道在此时的状态并且预测新的适合此时无线信道带宽的传输速率;(2) Predict the new transmission rate: After the receiving end of the streaming media transmission system obtains the packet loss rate and business data pass rate, based on the improved AIMD algorithm, it analyzes the state of the wireless channel at this time and predicts a new suitable wireless channel at this time bandwidth transmission rate;
(3)判断是否进行传输速率调整:流媒体传输系统的接收端预测出新的传输速率后,需要比较本次预测新速率与上次预测的新速率之间的差值是否超过设定的阈值,如果超过该阈值,则本次速率调整有效,否则,本次速率调整无效。(3) Determine whether to adjust the transmission rate: After the receiving end of the streaming media transmission system predicts the new transmission rate, it needs to compare whether the difference between the new rate predicted this time and the new rate predicted last time exceeds the set threshold , if the threshold is exceeded, the current rate adjustment is valid; otherwise, the current rate adjustment is invalid.
获取丢包率以及业务数据通过率步骤如下:The steps to obtain the packet loss rate and service data pass rate are as follows:
1)根据无线信道状态,确定丢包率以及业务数据通过率获取的时间间隔Δt;1) According to the state of the wireless channel, determine the time interval Δt for obtaining the packet loss rate and the service data pass rate;
2)确定获取第i个时间间隔参数时参与计算的时间间隔个数N,以及参与计算的第i-j时间间隔相应的权值;2) Determine the number N of time intervals participating in the calculation when obtaining the i-th time interval parameter, and the corresponding weights of the i-j time interval participating in the calculation;
3)统计流媒体传输参数,在接收端接收流媒体数据包的同时,统计第i个时间间隔内参数,包括丢包数LPNi,总的发送包数SPNi,业务数据量Di以及准确的间隔时间Ti;3) Statistical streaming media transmission parameters, when the receiving end receives the streaming media data packets, count the parameters in the i-th time interval, including the number of lost packets LPN i , the total number of sent packets SPN i , the amount of business data D i and the exact The interval time T i ;
4)计算第i个时间间隔内丢包率Pi与业务数据通过率Ri,通过对第i-N+1个时间间隔到第i个时间间隔所获得传输参数进行加权平滑计算丢包率与业务数据通过率。获取丢包率以及业务数据通过率步骤中时间间隔Δt取2s,参与计算的时间间隔个数N取值为2或者3。4) Calculate the packet loss rate P i and business data pass rate R i in the i-th time interval, and calculate the packet loss rate by weighting and smoothing the transmission parameters obtained from the i-N+1 time interval to the i-th time interval and business data pass rate. The time interval Δt in the step of obtaining the packet loss rate and business data pass rate is 2s, and the number N of time intervals involved in the calculation is 2 or 3.
改进的AIMD算法包含以下步骤:The improved AIMD algorithm contains the following steps:
1)根据测试统计视频传输系统接收端视频质量与丢包率的关系,设定算法中丢包率上下门限值分别为Ploss_max和Ploss_min,最大业务数据速率Rmax以及最小业务数据速率Rmin;1) According to the relationship between video quality and packet loss rate at the receiving end of the video transmission system, set the upper and lower thresholds of the packet loss rate in the algorithm as P loss_max and P loss_min , the maximum service data rate R max and the minimum service data rate R min ;
2)比较实测丢包率Pi与预设丢包率门限大小关系,确定信道状态是超载、轻载还是负载平衡;2) Compare the relationship between the measured packet loss rate Pi and the preset packet loss rate threshold, and determine whether the channel state is overloaded, lightly loaded or load balanced;
3)当信道超载时,即Pi≥Ploss_max,计算信道超载的差值因子m=Pi-Ploss_max以及超载速率降低因子βk=βk-1×(1-0.6×m)2,其中k表示连续超载的次数,根据公式Rnew=MAX{βK×Ri-50,Rmin}计算新的传输速率,其中Ri是第i个时间间隔内业务数据通过率,Rmin是系统设定的最小业务数据速率;3) When the channel is overloaded, that is, P i ≥ P loss_max , calculate the channel overload difference factor m=P i -P loss_max and the overload rate reduction factor β k = β k-1 ×(1-0.6×m) 2 , Where k represents the number of consecutive overloads, and the new transmission rate is calculated according to the formula R new = MAX{β K × R i -50, R min }, where R i is the service data throughput rate in the i-th time interval, and R min is The minimum service data rate set by the system;
4)当信道轻载时,即Pi≤Ploss_min,计算信道轻载的差值因子m=λ×(Ploss_min-Pi)以及速率增加量α=0.2em×R0+60,其中R0为预设的速率增长因子;根据公式Rnew=MIN{α+Ri,Rmax}计算新的传输速率,其中Ri是第i个时间间隔内业务数据通过率,Rmax是系统允许的最大业务数据速率;4) When the channel is lightly loaded, that is, P i ≤ P loss_min , calculate the channel light load difference factor m=λ×(P loss_min -P i ) and rate increase α=0.2e m ×R 0 +60, where R 0 is the preset rate growth factor; calculate the new transmission rate according to the formula R new =MIN{α+R i , R max }, where R i is the service data throughput rate in the i-th time interval, and R max is the system The maximum service data rate allowed;
5)当信道负载平衡时,无需对速率进行调整,退出速率控制方法。5) When the channel load is balanced, there is no need to adjust the rate, and the rate control method is exited.
改进的AIMD算法实现步骤3)中首次超载速率降低因子β0的取值范围是[0.85,0.95];改进的AIMD算法实现步骤4)中速率增加因子R0取值范围为[80,150]。The value range of the first overload rate reduction factor β0 in step 3) of the improved AIMD algorithm is [0.85, 0.95]; the value range of the rate increase factor R0 in step 4) of the improved AIMD algorithm is [80, 150] .
速率控制方法中设定了速率切换门限,门限值Rth一般设定为40~80Kb/s。The rate switching threshold is set in the rate control method, and the threshold value R th is generally set to 40-80 Kb/s.
本发明的技术特点及效果:Technical characteristics and effects of the present invention:
本发明采用较小的信道状态获取时间间隔来对信道状态进行分析,使得该方法能较快的反映无线信道的变化。本发明基于改进的AIMD算法预测合适的传输速率,相对于传统AIMD算法,改进的算法加大了丢包率对传输速率调整幅度的影响,能够使速率的预测更加接近无线信道带宽。本发明充分考虑了无线信道的特性,可以作为在无线环境下提高流媒体传输质量的一种有效方法。The present invention uses a smaller channel state acquisition time interval to analyze the channel state, so that the method can quickly reflect the change of the wireless channel. The invention predicts a suitable transmission rate based on the improved AIMD algorithm. Compared with the traditional AIMD algorithm, the improved algorithm increases the impact of the packet loss rate on the adjustment range of the transmission rate, and can make the rate prediction closer to the wireless channel bandwidth. The invention fully considers the characteristics of the wireless channel, and can be used as an effective method for improving the quality of streaming media transmission in a wireless environment.
本发明的有益效果还在于:The beneficial effect of the present invention also lies in:
■快速反映无线信道变化。无线信道具有衰落、时变等特性,信道状态的变化相对较快。本发明基于自定义较小的时间间隔获取丢包率以及业务数据通过率,同时对重要参数进行了平滑处理,符合无线信道的时变特性,使得速率的调整更加及时;■ Quickly reflect changes in wireless channels. The wireless channel has characteristics such as fading and time-varying, and the channel state changes relatively quickly. The present invention obtains the packet loss rate and the service data pass rate based on a self-defined smaller time interval, and at the same time smooths important parameters, which conforms to the time-varying characteristics of the wireless channel, making the adjustment of the rate more timely;
■降低了传输速率抖动。本发明通过改进的AIMD速率自适应调整算法,充分利用了丢包率和业务数据通过率这两个重要的参数对传输速率进行调整,使得速率调整更加准确,减少了锯齿状波动;■ Reduced transmission rate jitter. Through the improved AIMD rate self-adaptive adjustment algorithm, the present invention makes full use of the two important parameters of packet loss rate and service data pass rate to adjust the transmission rate, making the rate adjustment more accurate and reducing sawtooth fluctuations;
■减小系统开销。本发明通过在传输速率调整反馈之前加入判断机制,能够降低系统不必要的开销;■ Reduce system overhead. The present invention can reduce unnecessary overhead of the system by adding a judgment mechanism before the transmission rate adjustment feedback;
■提高无线流媒体传输服务质量和无线信道利用率。本发明通过对流媒体传输过程中速率进行控制,降低了丢包率以及传输速率的抖动,从而提高了传输服务质量和信道利用率。■Improve the service quality of wireless streaming media transmission and the utilization rate of wireless channel. The invention reduces the packet loss rate and the jitter of the transmission rate by controlling the rate in the streaming media transmission process, thereby improving the transmission service quality and channel utilization rate.
附图说明 Description of drawings
图1速率控制方法总体流程图。Figure 1 shows the overall flow chart of the rate control method.
图2丢包率以及业务数据通过率获取方法流程图。Fig. 2 is a flowchart of a method for obtaining packet loss rate and service data pass rate.
图3改进的AIMD算法总流程图。Figure 3 shows the general flow chart of the improved AIMD algorithm.
图4改进的AIMD算法中信道超载情况下速率调整流程图。Figure 4 is a flow chart of rate adjustment in the case of channel overload in the improved AIMD algorithm.
图5改进的AIMD算法中信道轻载情况下速率调整流程图。Figure 5 is a flow chart of rate adjustment in the case of channel light load in the improved AIMD algorithm.
图6无线实时视频传输系统的结构组成示意图。Fig. 6 is a schematic diagram of the structural composition of the wireless real-time video transmission system.
图7实施例系统性能测试结果峰值信噪比图。Fig. 7 is a peak signal-to-noise ratio diagram of the system performance test results of the embodiment.
图8实施例系统性能测试结果丢包率与业务数据通过率图。Figure 8 is a diagram of the packet loss rate and service data pass rate of the system performance test results of the embodiment.
具体实施方式 Detailed ways
基于以上考虑,本发明针对提高无线流媒体传输服务质量与信道利用率这一问题,提出了一种无线流媒体传输的速率控制方法。该方法基于流媒体传输系统,在系统接收端以较传统速率控制方法更短的时间间隔对丢包率和业务数据通过率等参数进行统计,基于上述两参数,通过改进的AIMD算法预测流媒体业务数据传输速率,在判定速率切换有效后由TCP协议反馈给发送端。和传统的流媒体速率控制方法相比,本发明能使得流媒体传输速率更准确的适配无线信道的带宽,从而保障流媒体传输服务质量、提高无线信道利用率。Based on the above considerations, the present invention proposes a rate control method for wireless streaming media transmission aiming at improving the service quality and channel utilization rate of wireless streaming media transmission. This method is based on the streaming media transmission system, and at the receiving end of the system, the parameters such as the packet loss rate and the business data passing rate are counted at a shorter time interval than the traditional rate control method. Based on the above two parameters, the streaming media is predicted by the improved AIMD algorithm. The service data transmission rate is fed back to the sender by the TCP protocol after the rate switching is determined to be valid. Compared with the traditional streaming media rate control method, the invention can make the streaming media transmission rate adapt to the bandwidth of the wireless channel more accurately, thereby ensuring the service quality of the streaming media transmission and improving the utilization rate of the wireless channel.
为了达到上述目的,本发明提供了一种无线流媒体传输中的速率控制方法,其特征在于:该方法在比传统速率控制方法更短的时间间隔内通过平滑处理获取丢包率和业务数据通过率等参数;基于上述参数及改进的加增乘减算法调整流媒体传输速率,在信道轻载时,由丢包率决定的变常数加性因子和常量加性因子决定速率增长幅度,在信道超载时,由丢包率决定的变常数乘性因子和常量减性因子决定速率降低幅度;最后本发明设定了速率切换门限,只有当本次预测速率与上次预测速率的差值超过设定门限时才认为速率切换有效。In order to achieve the above object, the present invention provides a rate control method in wireless streaming media transmission, which is characterized in that: the method obtains the packet loss rate and service data through smooth processing in a shorter time interval than the traditional rate control method Rate and other parameters; adjust the streaming media transmission rate based on the above parameters and the improved increase, increase, multiply and subtract algorithm. When the channel is lightly loaded, the variable constant additive factor and the constant additive factor determined by the packet loss rate determine the rate increase. When overloaded, the variable constant multiplicative factor and the constant subtractive factor determined by the packet loss rate determine the rate reduction rate; finally, the present invention sets the rate switching threshold, and only when the difference between the current predicted rate and the last predicted rate exceeds the set The rate switching is considered valid only when the threshold is set.
本发明提供了一种无线流媒体传输中的速率控制方法,其特征在于:所述方法包括以下步骤:The present invention provides a rate control method in wireless streaming media transmission, characterized in that: the method includes the following steps:
(1)获取丢包率以及业务数据通过率。在流媒体传输系统的接收端,通过对接收的流媒体数据包进行统计分析来获取丢包率以及业务数据通过率等参数;(1) Obtain the packet loss rate and service data pass rate. At the receiving end of the streaming media transmission system, parameters such as packet loss rate and service data passing rate are obtained by statistically analyzing the received streaming media data packets;
(2)预测新的传输速率。流媒体传输系统的接收端获取丢包率以及业务数据通过率之后,基于改进的AIMD算法,分析无线信道在此时的状态并且预测新的适合此时无线信道带宽的传输速率;(2) Predict the new transmission rate. After the receiving end of the streaming media transmission system obtains the packet loss rate and the service data pass rate, based on the improved AIMD algorithm, it analyzes the state of the wireless channel at this time and predicts a new transmission rate suitable for the wireless channel bandwidth at this time;
(3)判断是否进行传输速率调整。流媒体传输系统的接收端预测出新的传输速率后,需要比较本次预测新速率与上次预测的新速率之间的差值是否超过设定的阈值,如果超过该阈值,则本次速率调整有效,否则,本次速率调整无效。(3) Determine whether to adjust the transmission rate. After the receiving end of the streaming media transmission system predicts the new transmission rate, it needs to compare whether the difference between the new rate predicted this time and the new rate predicted last time exceeds the set threshold. If it exceeds the threshold, the current rate The adjustment is valid, otherwise, the current rate adjustment is invalid.
本发明提供了一种无线流媒体传输中的速率控制方法,其特征在于:上述获取丢包率以及业务数据通过率步骤如下:The present invention provides a rate control method in wireless streaming media transmission, which is characterized in that: the steps of obtaining the packet loss rate and service data pass rate are as follows:
1)根据无线信道状态,确定丢包率以及业务数据通过率获取的时间间隔Δt;1) According to the state of the wireless channel, determine the time interval Δt for obtaining the packet loss rate and the service data pass rate;
2)确定获取第i个时间间隔丢包率以及业务数据通过率时参与计算的时间间隔个数N,以及参与计算的第i-j时间间隔相应的权值αj和μj,其中0≤j≤N;2) Determine the number N of time intervals participating in the calculation when obtaining the packet loss rate and business data pass rate of the i-th time interval, and the corresponding weights α j and μ j of the ij-th time interval participating in the calculation, where 0≤j≤ N;
3)统计流媒体传输参数。在接收端接收流媒体数据包的同时,统计第i个时间间隔内参数,包括丢包数LPNi,总的发送包数SPNi,业务数据量Di以及准确的间隔时间Ti;3) Statistical streaming media transmission parameters. While receiving streaming media data packets at the receiving end, count the parameters in the i-th time interval, including the number of lost packets LPN i , the total number of sent packets SPN i , the amount of business data D i and the accurate interval time T i ;
4)计算第i个时间间隔的丢包率与业务数据通过率。通过对第i-N+1个时间间隔到第i个时间间隔所获得传输参数进行加权平滑计算丢包率与业务数据通过率。4) Calculate the packet loss rate and business data pass rate of the i-th time interval. The packet loss rate and the service data pass rate are calculated by performing weighted smoothing on the transmission parameters obtained from the i-N+1 time interval to the i-th time interval.
本发明所提出的无线流媒体传输中的速率控制方法,其特征在于:获取丢包率以及业务数据通过率步骤中时间间隔Δt一般取2s左右,参与计算的时间间隔个数N取值为2或者3。The rate control method in the wireless streaming media transmission proposed by the present invention is characterized in that: the time interval Δt in the step of obtaining the packet loss rate and the service data pass rate is generally about 2s, and the number N of time intervals participating in the calculation is 2 or 3.
本发明所提出的无线流媒体传输中的速率控制方法,其特征在于:改进的AIMD算法包含以下步骤:The rate control method in the wireless streaming media transmission proposed by the present invention is characterized in that: the improved AIMD algorithm comprises the following steps:
1)根据测试统计视频传输系统接收端视频质量与丢包率的关系,设定算法中丢包率上下门限值分别为Ploss_max和Ploss_min,最大业务数据速率Rmax以及最小业务数据速率Rmin;1) According to the relationship between video quality and packet loss rate at the receiving end of the video transmission system, set the upper and lower thresholds of the packet loss rate in the algorithm as P loss_max and P loss_min , the maximum service data rate R max and the minimum service data rate R min ;
2)比较实测丢包率Pi与预设丢包率门限大小关系,确定信道状态是超载、轻载还是负载平衡;2) Compare the relationship between the measured packet loss rate Pi and the preset packet loss rate threshold, and determine whether the channel state is overloaded, lightly loaded or load balanced;
3)当信道超载时,即Pi≥Ploss_max,计算信道超载的差值因子m=Pi-Ploss_max以及超载速率降低因子βk=βk-1×(1-0.6×m)2,其中k表示连续超载的次数,根据公式Rnew=MAX{βK×Ri-50,Rmin}计算新的传输速率,其中Ri是第i个时间间隔内业务数据通过率,Rmin是系统设定的最小业务数据速率;3) When the channel is overloaded, that is, P i ≥ P loss_max , calculate the channel overload difference factor m=P i -P loss_max and the overload rate reduction factor β k = β k-1 ×(1-0.6×m) 2 , Where k represents the number of consecutive overloads, and the new transmission rate is calculated according to the formula R new = MAX{β K × R i -50, R min }, where R i is the service data throughput rate in the i-th time interval, and R min is The minimum service data rate set by the system;
4)当信道轻载时,即Pi≤Ploss_min,计算信道轻载的差值因子m=λ×(Poss_min-Pi)以及速率增加量α=0.2em×R0+60,其中R0为预设的速率增长因子;根据公式Rnew=MIN{α+Ri,Rmax}计算新的传输速率,其中Ri是第i个时间间隔内业务数据通过率,Rmax是系统允许的最大业务数据速率;4) When the channel is lightly loaded, that is, P i ≤ P loss_min , calculate the channel light load difference factor m=λ×(P oss_min -P i ) and rate increase α=0.2e m ×R 0 +60, where R 0 is the preset rate growth factor; calculate the new transmission rate according to the formula R new =MIN{α+R i , R max }, where R i is the service data throughput rate in the i-th time interval, and R max is the system The maximum service data rate allowed;
5)当信道负载平衡时,无需对速率进行调整,退出速率控制方法。5) When the channel load is balanced, there is no need to adjust the rate, and the rate control method is exited.
本发明所提出的无线流媒体传输中的速率控制方法,其特征在于:改进的AIMD算法实现步骤3)中首次超载速率降低因子β0的取值范围是[0.85,0.95];改进的AIMD算法实现步骤4)中速率增加因子R0取值范围为[80,150]。The rate control method in the wireless streaming media transmission proposed by the present invention is characterized in that: the improved AIMD algorithm realizes step 3) in the value range of first overload rate reduction factor β 0 is [0.85,0.95]; Improved AIMD algorithm The value range of the rate increase factor R0 in step 4) is [80, 150].
本发明所提出的无线流媒体传输中的速率控制方法,其特征在于:速率控制方法中设定了速率切换门限,门限值Rth一般设定为40~80Kb/s。The rate control method in the wireless streaming media transmission proposed by the present invention is characterized in that a rate switching threshold is set in the rate control method, and the threshold value R th is generally set at 40-80 Kb/s.
为了使本发明目的、技术方案特征以及优点更加清楚,下面结合附图详细描述本发明。In order to make the purpose, technical solution features and advantages of the present invention clearer, the present invention will be described in detail below in conjunction with the accompanying drawings.
本发明是一种用于无线信道环境下流媒体传输的速率控制方法。流媒体传输,特别是实时视频传输,具有连续性传输、数据量大、对实时性、可靠性要求高等特点,如果信道不可靠或者出现拥塞的情况,会严重影响流媒体传输的服务质量。在有线网络的情况下,影响流媒体传输质量的主要原因是网络的拥塞。然而在无线环境下,影响流媒体传输质量的主要因素是无线信道的带宽时变性和不可靠性。因此本发明主要针对无线环境下的流媒体传输,通过分析一段时间的传输情况从而判断无线信道的状态,最后预测出合适的传输速率。本发明能够降低流媒体传输过程中的丢包率,从而提高无线流媒体传输的服务质量以及无线信道利用率。The invention is a rate control method for streaming media transmission in a wireless channel environment. Streaming media transmission, especially real-time video transmission, has the characteristics of continuous transmission, large amount of data, and high requirements for real-time performance and reliability. If the channel is unreliable or congested, the service quality of streaming media transmission will be seriously affected. In the case of a wired network, the main reason affecting the quality of streaming media transmission is the congestion of the network. However, in a wireless environment, the main factor affecting the quality of streaming media transmission is the time-varying and unreliable bandwidth of the wireless channel. Therefore, the present invention mainly aims at the streaming media transmission in the wireless environment, and judges the state of the wireless channel by analyzing the transmission situation for a period of time, and finally predicts an appropriate transmission rate. The invention can reduce the packet loss rate in the streaming media transmission process, thereby improving the service quality and wireless channel utilization rate of the wireless streaming media transmission.
本发明采用比传统流媒体速率控制方法更小的时间间隔获取丢包率以及业务数据通过率,以改进的AIMD算法算法预测新的传输速率。这使得该方法能够快速反映无线信道的变化,准确预测新传输速率,从而提高无线流媒体传输的服务质量以及无线信道的利用率。The invention adopts a shorter time interval than the traditional stream media rate control method to obtain the packet loss rate and the service data pass rate, and uses the improved AIMD algorithm to predict the new transmission rate. This enables the method to quickly reflect the change of the wireless channel and accurately predict the new transmission rate, thereby improving the service quality of wireless streaming media transmission and the utilization rate of the wireless channel.
在使用本发明之前,需要根据实际的流媒体应用情景以及无线环境来确定本发明的预设参数,这些参数值对本发明提供的速率控制方法的性能有较大的影响。Before using the present invention, it is necessary to determine the preset parameters of the present invention according to the actual streaming media application scenario and the wireless environment, and these parameter values have a great influence on the performance of the rate control method provided by the present invention.
■最大业务数据速率Rmax以及最小业务数据速率Rmim。这两个参数分别是业务数据速率调整的最大值以及最小值。它们的值是根据流媒体业务对业务数据吞吐量的实际要求来确定。■ Maximum service data rate R max and minimum service data rate R mim . These two parameters are the maximum value and the minimum value of the service data rate adjustment respectively. Their values are determined according to the actual requirements of the streaming media service on the service data throughput.
■最大丢包率门限Ploss_max以及最小丢包率门限Ploss_min。最大丢包率门限表示流媒体传输系统对丢包率的容忍程度,最小丢包率门限表示流媒体传输系统对信道轻载的容忍程度。■The maximum packet loss rate threshold P loss_max and the minimum packet loss rate threshold P loss_min . The maximum packet loss rate threshold indicates the tolerance of the streaming media transmission system to the packet loss rate, and the minimum packet loss rate threshold indicates the tolerance of the streaming media transmission system to the channel light load.
■速率控制的时间间隔Δt。该值表示在流媒体传输的过程中,每隔Δt时间,根据本发明的速率控制算法对流媒体传输速率进行调整。如果该值取值太大,则无法反映无线信道的时变性;如果该值取值太小,则会导速率抖动较大。■ Time interval Δt for rate control. This value indicates that in the process of streaming media transmission, the streaming media transmission rate is adjusted according to the rate control algorithm of the present invention at intervals of Δt. If the value is too large, it cannot reflect the time-varying nature of the wireless channel; if the value is too small, the rate jitter will be large.
下面详细说明本发明方法的各个步骤。Each step of the method of the present invention will be described in detail below.
图1为本发明方法总体流程图。从图中可以看出,本发明方法主要包括丢包率以及业务数据通过率获取、改进的AIMD算法速率调整以及系统速率切换判定三个部分。基于设定好速率控制的时间间隔Δt,本发明方法在Δt时间段内获取流媒体传输参数,在Δt最后时刻使用改进的AIMD算法进行新传输速率的预测,同时对系统速率切换的有效性进行判定。本发明的速率控制方法一直以Δt为时间间隔循环运行。为了使下面介绍的详细流程更加清楚,均以第i个时间间隔为例进行说明。Fig. 1 is the general flowchart of the method of the present invention. It can be seen from the figure that the method of the present invention mainly includes three parts: acquisition of packet loss rate and service data pass rate, rate adjustment of the improved AIMD algorithm, and system rate switching judgment. Based on the time interval Δt of the rate control being set, the method of the present invention obtains the streaming media transmission parameters within the time period Δt, and uses the improved AIMD algorithm to predict the new transmission rate at the last moment of Δt, and at the same time checks the effectiveness of the system rate switching determination. The rate control method of the present invention always runs cyclically with Δt as the time interval. In order to make the detailed process described below clearer, the ith time interval is taken as an example for illustration.
本发明详细流程如下:The detailed flow process of the present invention is as follows:
(1)丢包率以及业务数据通过率。接收端在接收流媒体数据包的同时,对第i个时间间隔内流媒体数据包的传输信息(包括发送端总共发送的数据包个数,接收端接收的数据包个数,接收端接收总的业务数据量以及准确的时间间隔值)进行统计,从而计算第i个时间间隔内丢包率Pi和业务数据通过率Ri;(1) Packet loss rate and service data pass rate. While the receiving end is receiving the streaming media data packet, the transmission information of the streaming media data packet in the i-th time interval (including the total number of data packets sent by the sending end, the number of data packets received by the receiving end, and the total number of data packets received by the receiving end business data volume and accurate time interval value) to calculate the packet loss rate P i and business data pass rate R i in the i-th time interval;
(2)预测新的传输速率。接收端在获取丢包率以及业务数据通过率后,将丢包率Pi与预设的丢包率门限进行比较,从而获知当前的信道状态:超载、轻载以及平衡。根据三种信道状态以及丢包率、业务数据通过率,改进的AIMD算法能准确的预测新速率;(2) Predict the new transmission rate. After obtaining the packet loss rate and service data pass rate, the receiving end compares the packet loss rate P i with the preset packet loss rate threshold to know the current channel status: overload, light load and balance. According to the three channel states, packet loss rate, and service data pass rate, the improved AIMD algorithm can accurately predict the new rate;
(3)判断是否进行传输速率调整。当得到新的传输速率Rnew后,将该值与上次速率预测值Rnew_last进行比较,只有当|Rnew-Rnew_last|>Rth时,才通知流媒体系统发送端改变传输速率,其中Rth为预设的速率调整门限值。(3) Determine whether to adjust the transmission rate. When the new transmission rate R new is obtained, compare this value with the last rate prediction value R new_last , and only when |R new - R new_last | > R th , the streaming media system is notified to change the transmission rate, where R th is a preset rate adjustment threshold.
图1中获取丢包率以及业务数据通过率的处理流程如图2所示,接收端通过接收到的流媒体数据包获取当前数据包的序号、当前数据包所属的视频帧以及包负载数据长度等信息,从而计算出速率自适应调整所需要的参量。The processing flow of obtaining the packet loss rate and service data pass rate in Figure 1 is shown in Figure 2. The receiving end obtains the sequence number of the current data packet, the video frame to which the current data packet belongs, and the length of the packet payload data through the received streaming media data packet and other information to calculate the parameters required for rate adaptive adjustment.
丢包率以及业务数据通过率获取的流程如下:The process of obtaining packet loss rate and business data pass rate is as follows:
(1)首先初始化计数器。包括初始化视频帧数计数器F_cnt、接收流媒体数据包计数器P_cnt以及业务数据量计数器Di。由于视频帧数据量比较大,所以发送端会将其切割为几个数据包进行传输。因此每接收一个流媒体数据包,首先需要根据发送端预设的标志位来判断该数据包是否是一个视频帧的起始数据包,如果是则将帧计数器F_cnt加1,否则保持不变;然后获取该数据包的负载业务数据长度,将该长度加到业务数据量计数器Di上;最后包计数器P_cnt相应加1。(1) Initialize the counter first. It includes initializing the video frame number counter F_cnt, the received stream media data packet counter P_cnt and the service data amount counter D i . Since the data volume of the video frame is relatively large, the sender will cut it into several data packets for transmission. Therefore, every time a streaming media data packet is received, it is first necessary to judge whether the data packet is the initial data packet of a video frame according to the flag bit preset by the sending end, if so, add 1 to the frame counter F_cnt, otherwise it remains unchanged; Then the payload service data length of the data packet is obtained, and the length is added to the service data amount counter D i ; finally, the packet counter P_cnt is incremented by 1 accordingly.
(2)确定计数起始参数。接收流媒体数据包计数器P_cnt为1时,新一时间段的信道预测量获取开始,获取该数据包的序号psn1(为了查看流媒体传输是否丢包,需要为每个数据包加上一个16比特序号)以及当前时间time1;(2) Determine the counting start parameters. When the counter P_cnt of the received streaming media data packet is 1, the channel prediction acquisition of a new time period starts, and the sequence number psn1 of the data packet is obtained (in order to check whether the streaming media transmission is lost, a 16-bit value needs to be added to each data packet serial number) and the current time time1;
(3)确定计数终止参数。为了适应无线信道的快速变化,该系统中设定当接收帧数达到30帧时,该时间段结束,获取此时数据包的序号psn2以及当前时间time2。系统的视频帧率为15帧/s,因此每一时间间隔大约为Δt=2s;(3) Determine the counting termination parameter. In order to adapt to the rapid change of the wireless channel, the system is set that when the number of received frames reaches 30 frames, the time period ends, and the sequence number psn2 of the data packet and the current time time2 are obtained at this time. The video frame rate of the system is 15 frames/s, so each time interval is about Δt=2s;
(4)计算该时间间隔的准确值Ti=time2-time1,以及在Ti时间内发送端发送的总包数SPNi=psn2-psn1+1,丢包数LPNi=SPNi-P_cnt。由于数据包头中包序号占16比特,因此最大序号为65535,在系统运行时会出现序号溢出,即出现发送包数SPNi<0情况,此时SPNi=psn2-psn1+1+65536。(4) Calculate the exact value of the time interval T i =time2-time1, and the total number of packets SPN i =psn2-
(5)计算丢包率Pi。由于预测周期Δt较小,而丢包率反映的是信道的拥塞状态,与之前时刻信道拥塞状态有关,因此取前N个时间间隔内各种数据包统计结果的加权值进行计算,N值取决于信道环境。根据公式(2)计算丢包率,式中当N=3时,加权值α取值为:α0=0.5,α1=0.3,α2=0.2;当N=2时,加权值α取值为:α0=0.7,α1=0.3;LPNi-j表示第i-j个时间间隔内统计得到的丢包总数,SPNi-j表示第i-j个时间间隔内发送端发送数据包总数。(5) Calculate the packet loss rate P i . Since the prediction period Δt is small, and the packet loss rate reflects the congestion state of the channel, which is related to the channel congestion state at the previous moment, the weighted value of the statistical results of various data packets in the previous N time intervals is used for calculation, and the N value depends on in the channel environment. Calculate the packet loss rate according to the formula (2), where when N=3, the weighted value α is: α 0 =0.5, α 1 =0.3, α 2 =0.2; when N=2, the weighted value α is The values are: α 0 =0.7, α 1 =0.3; LPN ij represents the total number of lost packets collected in the ij-th time interval, and SPN ij represents the total number of data packets sent by the sender in the ij-th time interval.
(6)计算业务数据通过率Ri。根据公式(3)计算业务数据通过率,式中当N=3时,加权值μ0=0.7,μ1=0.2,μ2=0.1;当N=2时,加权值μ0=0.8,μ1=0.2;Di-j表示第i-j个时间间隔内接收到的业务数据量,Ti-j为第i-j个时间间隔的准确时间值;(6) Calculate the business data pass rate R i . Calculate the service data pass rate according to formula (3), where when N=3, the weighted value μ 0 =0.7, μ 1 =0.2, μ 2 =0.1; when N=2, the weighted value μ 0 =0.8, μ 1 = 0.2; D ij represents the amount of business data received in the ij time interval, and T ij is the exact time value of the ij time interval;
(7)对上述使用的各种计数器进行复位,开始下一时间间隔的信道预测量计算。(7) Reset the various counters used above, and start the channel prediction calculation for the next time interval.
图1中改进的AIMD算法总流程图如图3所示。传统的AIMD算法不考虑实时丢包率参数的影响,当信道超载时以常数因子乘以当前速率来降低传输速率,信道轻载时以常数加上当前传输速率来增加传输速率,这样会导致传输速率的锯齿状波动。本发明在调整速率时考虑丢包率这一重要参数,丢包率与门限值差值越大,速率调整的幅度就越大。The overall flow chart of the improved AIMD algorithm in Figure 1 is shown in Figure 3. The traditional AIMD algorithm does not consider the influence of real-time packet loss rate parameters. When the channel is overloaded, the transmission rate is reduced by multiplying the current rate by a constant factor. When the channel is lightly loaded, the transmission rate is increased by adding a constant factor to the current transmission rate. The sawtooth fluctuation of the rate. The present invention considers the packet loss rate as an important parameter when adjusting the rate, and the greater the difference between the packet loss rate and the threshold value, the greater the range of rate adjustment.
改进的AIMD算法步骤如下:The steps of the improved AIMD algorithm are as follows:
(1)设定流媒体传输系统所允许的最大丢包率门限Ploss_max以及最小丢包率门限Ploss_min;(1) Setting the maximum packet loss rate threshold P loss_max and the minimum packet loss rate threshold P loss_min allowed by the streaming media transmission system;
(2)将丢包率Pi与预设的丢包率门限进行比较。若丢包率Pi大于最大丢包率门限Ploss_max,则说明当前信道超载,需要降低速率以提高视频传输服务质量;(2) Comparing the packet loss rate Pi with a preset packet loss rate threshold. If the packet loss rate P i is greater than the maximum packet loss rate threshold P loss_max , it means that the current channel is overloaded, and the rate needs to be reduced to improve the video transmission service quality;
(3)若丢包率Pi小于最小丢包率门限Ploss_min,则说明当前信道轻载,需要增加速率以提高无线信道利用率;(3) If the packet loss rate P i is less than the minimum packet loss rate threshold P loss_min , it means that the current channel is lightly loaded, and the rate needs to be increased to improve the utilization rate of the wireless channel;
(4)若丢包率Pi大于最小丢包率门限Ploss_min且小于最大丢包率门限Ploss_max,则说明信道负载平衡,无需进行速率调整,退出速率控制算法。(4) If the packet loss rate P i is greater than the minimum packet loss rate threshold P loss_min and smaller than the maximum packet loss rate threshold P loss_max , it means that the channel load is balanced, no rate adjustment is required, and the rate control algorithm is exited.
图2中信道超载时的处理流程如图4所示,改进的AIMD算法通过综合考虑信道超载的程度以及信道当前业务数据通过率来确定新的传输速率。超载时新速率确定流程如下:The processing flow when the channel is overloaded in Figure 2 is shown in Figure 4. The improved AIMD algorithm determines the new transmission rate by comprehensively considering the degree of channel overload and the current service data throughput rate of the channel. The new rate determination process when overloaded is as follows:
(1)首先初始化首次信道超载速率降低因子β0,该值影响速率降低的幅度,在本发明中取值为0.93;(1) Initialize the first time channel overload rate reduction factor β 0 , which affects the rate reduction range, which is 0.93 in the present invention;
(2)判断信道连续超载的次数k。当信道连续超载时,说明信道进入了比较大的衰落,需要相对增加速率降低的幅度。(2) Judge the times k of continuous channel overloading. When the channel is continuously overloaded, it means that the channel has entered a relatively large fading, and it is necessary to increase the rate reduction range relatively.
(3)计算本次信道超载的差值因子m=Pi-Ploss_max,其中Pi为实测丢包率,Ploss_max为系统预设最大丢包率;(3) Calculate the difference factor m=P i -P loss_max for this channel overload, where P i is the measured packet loss rate, and P loss_max is the maximum packet loss rate preset by the system;
(4)计算本次信道超载速率降低因子βk=βk-1×(1-0.6×m)2;(4) Calculating the channel overload rate reduction factor β k = β k-1 × (1-0.6 × m) 2 ;
(5)计算本次速率自适应调整产生的新速率Rnew=MAX{βK×Ri-50,Rmin},其中Ri为业务数据通过率,Rmin为系统预设最小业务数据传输速率;(5) Calculate the new rate R new =MAX{β K ×R i -50, R min } generated by this rate adaptive adjustment, where R i is the service data throughput rate, and R min is the system preset minimum service data transmission rate;
图2中当信道轻载时的处理流程如图5所示,速率自适应控制模块通过综合考虑信道轻载的程度以及信道当前业务数据通过率来确定新的传输速率。轻载时新速率确定流程如下:The processing flow when the channel is lightly loaded in Figure 2 is shown in Figure 5. The rate adaptive control module determines the new transmission rate by comprehensively considering the degree of channel light load and the current service data throughput rate of the channel. The new rate determination process at light load is as follows:
(1)初始化速率增加因子R0,该值影响速率增加的幅度,在系统中取值为100;(1) Initialize the rate increase factor R 0 , which affects the range of rate increase, and the value is 100 in the system;
(2)恢复速率降低因子β0,为下次信道超载做准备;(2) The recovery rate is reduced by a factor β 0 to prepare for the next channel overload;
(3)计算本次信道轻载的丢包率差值因子m=λ×(Ploss_min-Pi),其中λ是丢包率差值放大倍数;(3) Calculate the packet loss rate difference factor m=λ×(P loss_min -P i ) under light load of the channel this time, where λ is the magnification factor of the packet loss rate difference;
(4)计算本次信道轻载下速率增加量α=0.2em×R0+60,其中R0速率增加因子;(4) Calculate the rate increase under light channel load this time α = 0.2e m × R 0 +60, where R 0 is the rate increase factor;
(5)计算本次速率自适应调整产生的新速率Rnew=MIN{α+Ri,Rmax},其中Ri为第i个时间间隔内业务数据通过率,Rmax为最大业务数据速率;(5) Calculate the new rate R new = MIN{α+R i , R max } generated by the rate adaptive adjustment this time, where R i is the service data throughput rate in the i-th time interval, and R max is the maximum service data rate ;
本发明提供的无线流媒体速率控制方法可用于多种场景,包括无线视频监控、无线视频会议、无线视频电话以及流媒体下载播放等。本发明提供的方法能够有效的提高这些应用的服务质量。The wireless streaming media rate control method provided by the present invention can be used in various scenarios, including wireless video monitoring, wireless video conferencing, wireless video telephony, streaming media downloading and playing, and the like. The method provided by the invention can effectively improve the service quality of these applications.
下面以对服务质量要求较高的实时视频应用作为本发明的实施方案。将流媒体速率控制方法用于端到端无线实时视频传输系统中,从而保障实时视频在传输过程中的服务质量。In the following, real-time video applications with higher requirements on service quality are taken as the embodiment of the present invention. The streaming media rate control method is used in an end-to-end wireless real-time video transmission system, thereby ensuring the service quality of real-time video during transmission.
图6为本发明具体实时方案的系统框图,该系统是基于嵌入式Linux平台实现的端到端无线视频实时传输系统。该系统以802.11b/g标准的Ad-Hoc(无线自组织网络)模式作为实时视频传输的无线环境。由于802.11b/g标准物理层速率最高能够到达54Mb/s,能够达到实时视频传输的带宽要求。该系统由视频发送端和视频接收端组成,视频发送端主要包括视频采集编码模块和视频数据发送模块,视频接收端主要包括数据接收模块、视频解码显示模块以及速率控制模块。下面对该系统各个模块及其特点进行详细描述。FIG. 6 is a system block diagram of a specific real-time solution of the present invention, which is an end-to-end wireless video real-time transmission system based on an embedded Linux platform. The system uses the 802.11b/g standard Ad-Hoc (wireless self-organizing network) mode as the wireless environment for real-time video transmission. Because the 802.11b/g standard physical layer rate can reach up to 54Mb/s, it can meet the bandwidth requirements of real-time video transmission. The system consists of a video sending end and a video receiving end. The video sending end mainly includes a video capture encoding module and a video data sending module. The video receiving end mainly includes a data receiving module, a video decoding display module and a rate control module. Each module of the system and its characteristics are described in detail below.
视频发送端:Video sender:
(1)视频采集编码模块。该模块通过配置摄像头视频采集的帧率、分辨率等参数来获取满足要求的实时视频,为了降低传输所需的带宽,由编码模块对其进行H264视频格式编码。系统运行时,采集编码模块的主要参数如下:(1) Video capture and encoding module. This module acquires real-time video that meets the requirements by configuring parameters such as the frame rate and resolution of camera video collection. In order to reduce the bandwidth required for transmission, the encoding module encodes it in H264 video format. When the system is running, the main parameters of the acquisition and encoding module are as follows:
■视频分辨率:640x400,传输过程中保持不变;■Video resolution: 640x400, remains unchanged during transmission;
■视频帧率:15帧/s,传输过程中保持不变;■Video frame rate: 15 frames/s, which remains unchanged during transmission;
■编码器输出业务数据速率:初始化为2048Kb/s,之后由速率控制模块进行调整;■Encoder output service data rate: initialized to 2048Kb/s, then adjusted by the rate control module;
(2)视频数据发送模块。为了保证视频数据的实时传输,我们采用RTP作为视频数据流的传输协议。使用RTP作为视频数据传输协议能够更加方便的使用本发明,因为RTP协议规定了流媒体信息传输时RTP包头所携带的信息,包括该RTP数据包的来源,数据包的序列号以及时间戳等。根据RTP数据包的来源,流媒体传输系统在接收端能够同时接收多路媒体源;根据RTP数据包的序列号,可以计算出流媒体传输系统在某一个时间段内的丢包率。通过获取RTP数据包业务数据的长度,可以计算出流媒体传输系统在一个时间段内业务数据通过率。系统在该模块将一帧视频数据按照RTP进行切割打包,然后交付给UDP进行传输。(2) Video data sending module. In order to ensure the real-time transmission of video data, we use RTP as the transmission protocol of video data stream. Using RTP as the video data transmission protocol can use the present invention more conveniently, because the RTP protocol stipulates the information carried by the RTP header when the streaming media information is transmitted, including the source of the RTP data packet, the serial number and the time stamp of the data packet. According to the source of the RTP data packet, the streaming media transmission system can receive multiple media sources at the same time at the receiving end; according to the sequence number of the RTP data packet, the packet loss rate of the streaming media transmission system within a certain period of time can be calculated. By obtaining the length of the service data of the RTP data packet, the service data throughput rate of the streaming media transmission system within a period of time can be calculated. In this module, the system cuts and packages a frame of video data according to RTP, and then delivers it to UDP for transmission.
视频接收端:Video receiver:
(1)数据接收模块。该模块功能是接收RTP数据包,进而对RTP数据包进行解包,将同一帧的RTP数据包负载组合在一起成为视频帧。(1) Data receiving module. The function of this module is to receive RTP data packets, and then unpack the RTP data packets, and combine the RTP data packet payloads of the same frame into a video frame.
(2)视频解码显示模块。数据接收模块将接收到的RTP数据合成被压缩的视频帧之后,由解码模块对其进行H264视频格式解码。由于解码后的视频数据是YUV420格式,因此需要将其转换为RGB格式之后由LCD进行显示。(2) Video decoding display module. After the data receiving module synthesizes the received RTP data into a compressed video frame, the decoding module decodes it in H264 video format. Since the decoded video data is in YUV420 format, it needs to be converted to RGB format and displayed by LCD.
(3)速率控制模块。该模块是本发明提供的方法在该实施例中的体现,是保障该系统实时视频传输服务质量的重要模块。该模块主要由3个子模块组成,包括丢包率及业务数据通过率获取子模块、改进的AIMD速率调整子模块以及系统速率切换判定子模块。三个子模块的描述如下:(3) Speed control module. This module is the embodiment of the method provided by the present invention in this embodiment, and is an important module for guaranteeing the service quality of real-time video transmission of the system. This module is mainly composed of three sub-modules, including the packet loss rate and business data pass rate acquisition sub-module, the improved AIMD rate adjustment sub-module and the system rate switching judgment sub-module. The descriptions of the three sub-modules are as follows:
■丢包率以及业务数据通过率获取子模块。该子模块在数据接收模块进行RTP数据包接收的同时,按照本发明提供的方法对传输参数进行统计,从而获取丢包率以及业务数据通过率。系统将上述参数获取的周期设定为接收端接收到30帧视频的时间,由于系统视频帧率为15帧/s保持不变,因此时间约为2s。该子模块按照本发明提供的公式(2)和(3)计算丢包率以及业务数据通过率时参数的取值如下:■Packet loss rate and business data pass rate acquisition sub-module. The sub-module counts the transmission parameters according to the method provided by the present invention while the data receiving module receives the RTP data packets, so as to obtain the packet loss rate and the service data pass rate. The system sets the cycle of obtaining the above parameters as the time when the receiving end receives 30 frames of video. Since the system video frame rate remains unchanged at 15 frames/s, the time is about 2s. The value of parameter when this submodule calculates packet loss rate and service data passing rate according to formula (2) and (3) provided by the present invention is as follows:
参与计算的时间间隔个数N=3; The number of time intervals involved in the calculation N=3;
丢包率平滑因子α0=0.5,α1=0.3,α2=0.2; Packet loss rate smoothing factor α 0 =0.5, α 1 =0.3, α 2 =0.2;
μ0=0.7,μ1=0.2,μ2=0.1。 μ 0 =0.7, μ 1 =0.2, μ 2 =0.1.
■改进的AIMD速率调整子模块。该子模块在获取丢包率以及业务数据通过率之后,按照本发明提供的改进的AIMD算法对传输速率进行调整,各种参数的取值如下:■Improved AIMD rate adjustment submodule. This sub-module adjusts the transmission rate according to the improved AIMD algorithm provided by the present invention after obtaining the packet loss rate and the service data passing rate, and the values of various parameters are as follows:
丢包率上下门限值,Ploss_max=0.03,Ploss_min=0.008; Upper and lower thresholds of packet loss rate, P loss_max = 0.03, P loss_min = 0.008;
最大业务数据速率Rmax=2048kb/s,最小业务数据速率Rmin=256kb/s; The maximum service data rate R max =2048kb/s, the minimum service data rate R min =256kb/s;
首次信道超载速率降低因子β0=0.93; The first channel overload rate reduction factor β 0 =0.93;
速率增加因子R0=100。 Rate increase factor R 0 =100.
■系统速率切换判定子模块。该模块主要是为了防止系统出现频繁切换而增加系统开销。按照本发明提供的系统速率切换判定方法,其判定的阈值Rth设定为60Kb/s。■The system rate switching judgment sub-module. This module is mainly to prevent frequent switching of the system from increasing system overhead. According to the method for judging system rate switching provided by the present invention, the threshold R th for judging is set to 60Kb/s.
如图6所示,该系统具有由两条无线传输链路。第一条是基于RTP/UDP/IP的视频数据传输链路,主要负责实时的视频传输;第二条是基于TCP/IP的反馈控制链路,主要负责速率控制模块的速率调整信息的可靠传输。As shown in Figure 6, the system has two wireless transmission links. The first is a video data transmission link based on RTP/UDP/IP, which is mainly responsible for real-time video transmission; the second is a feedback control link based on TCP/IP, which is mainly responsible for the reliable transmission of the rate adjustment information of the rate control module .
为了评估上述系统性能,在该系统进行视频传输过程中统计了丢包率、业务数据通过率以及视频传输的PSNR(峰值信噪比)等参数。系统性能的测试环境是有着许多不同无线接入点的走廊,不同无线信号之间干扰比较大,发送端系统与接收端系统相距42m。性能测试的结果如下。In order to evaluate the performance of the above system, parameters such as packet loss rate, business data pass rate, and PSNR (peak signal-to-noise ratio) of video transmission were counted during the video transmission process of the system. The test environment for system performance is a corridor with many different wireless access points, the interference between different wireless signals is relatively large, and the distance between the sending end system and the receiving end system is 42m. The results of the performance tests are as follows.
如图7所示,是计算系统发送端H264压缩前与接收端进行H264解码后视频数据的PSNR值。因此,图7所示视频质量的损失由两部分组成:传输损失与H264编解码损失。As shown in Figure 7, it is the PSNR value of the video data after H264 decoding at the sending end and H264 decoding at the receiving end of the calculation system before H264 compression. Therefore, the loss of video quality shown in Figure 7 consists of two parts: transmission loss and H264 codec loss.
视频传输的PSNR值与人们对视频感知质量的评价有着一定的关系,将图7中的PSNR值进行分段统计如下:The PSNR value of video transmission has a certain relationship with people's evaluation of video perception quality. The PSNR value in Figure 7 is segmented and counted as follows:
PSNR值大于37,视频等级为5,视频质量非常好,占总数88.312%; The PSNR value is greater than 37, the video level is 5, and the video quality is very good, accounting for 88.312% of the total;
PSNR值位于区间(31,37],视频等级为4,视频质量好,占总数9.5238%; The PSNR value is in the interval (31, 37], the video level is 4, and the video quality is good, accounting for 9.5238% of the total;
PSNR值位于区间(25,31],视频等级为3,视频质量一般,占总数0%; The PSNR value is in the interval (25, 31], the video level is 3, and the video quality is average, accounting for 0% of the total;
PSNR值位于区间(20,25],视频等级为2,视频质量较差,占总数0.8658%; The PSNR value is in the interval (20, 25], the video level is 2, and the video quality is poor, accounting for 0.8658% of the total;
PSNR值小于20,视频等级为4,视频质量差,占总数1.2987%; The PSNR value is less than 20, the video level is 4, and the video quality is poor, accounting for 1.2987% of the total;
从统计的结果可以看出,该系统能够提供较高的视频传输服务质量,97.8358%情况下能提供好的视频质量,只有当无线信道衰落很大时,会短暂的引起视频质量的降低。From the statistical results, it can be seen that the system can provide high video transmission service quality, 97.8358% of the cases can provide good video quality, only when the wireless channel fading is very large, it will temporarily cause the video quality to decrease.
图8是系统进行视频传输时统计的业务数据速率以及丢包率的参数图。从图中可以看出,在大部分时间内,系统的丢包率都被控制在2%以下,然而由于信道的时变特性,在某些时候系统的丢包率比较大,此时无线信道进入了比较大的衰落中,通过本发明的速率控制方法,业务数据速率会立即响应无线信道的变化,降低传输速率,从而保证较低的丢包率。FIG. 8 is a parameter diagram of the service data rate and packet loss rate counted when the system performs video transmission. It can be seen from the figure that in most of the time, the packet loss rate of the system is controlled below 2%. In relatively large fading, through the rate control method of the present invention, the service data rate will immediately respond to the change of the wireless channel and reduce the transmission rate, thereby ensuring a lower packet loss rate.
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