WO2008014707A1 - Method, system and terminal of adjusting video quality - Google Patents
Method, system and terminal of adjusting video quality Download PDFInfo
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- WO2008014707A1 WO2008014707A1 PCT/CN2007/070335 CN2007070335W WO2008014707A1 WO 2008014707 A1 WO2008014707 A1 WO 2008014707A1 CN 2007070335 W CN2007070335 W CN 2007070335W WO 2008014707 A1 WO2008014707 A1 WO 2008014707A1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/75—Media network packet handling
- H04L65/752—Media network packet handling adapting media to network capabilities
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
- H04N21/25—Management operations performed by the server for facilitating the content distribution or administrating data related to end-users or client devices, e.g. end-user or client device authentication, learning user preferences for recommending movies
- H04N21/262—Content or additional data distribution scheduling, e.g. sending additional data at off-peak times, updating software modules, calculating the carousel transmission frequency, delaying a video stream transmission, generating play-lists
- H04N21/26208—Content or additional data distribution scheduling, e.g. sending additional data at off-peak times, updating software modules, calculating the carousel transmission frequency, delaying a video stream transmission, generating play-lists the scheduling operation being performed under constraints
- H04N21/26216—Content or additional data distribution scheduling, e.g. sending additional data at off-peak times, updating software modules, calculating the carousel transmission frequency, delaying a video stream transmission, generating play-lists the scheduling operation being performed under constraints involving the channel capacity, e.g. network bandwidth
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
- H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
- H04N21/442—Monitoring of processes or resources, e.g. detecting the failure of a recording device, monitoring the downstream bandwidth, the number of times a movie has been viewed, the storage space available from the internal hard disk
- H04N21/44209—Monitoring of downstream path of the transmission network originating from a server, e.g. bandwidth variations of a wireless network
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
- H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
- H04N21/442—Monitoring of processes or resources, e.g. detecting the failure of a recording device, monitoring the downstream bandwidth, the number of times a movie has been viewed, the storage space available from the internal hard disk
- H04N21/4425—Monitoring of client processing errors or hardware failure
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
- H04N21/45—Management operations performed by the client for facilitating the reception of or the interaction with the content or administrating data related to the end-user or to the client device itself, e.g. learning user preferences for recommending movies, resolving scheduling conflicts
- H04N21/462—Content or additional data management, e.g. creating a master electronic program guide from data received from the Internet and a Head-end, controlling the complexity of a video stream by scaling the resolution or bit-rate based on the client capabilities
- H04N21/4621—Controlling the complexity of the content stream or additional data, e.g. lowering the resolution or bit-rate of the video stream for a mobile client with a small screen
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/637—Control signals issued by the client directed to the server or network components
- H04N21/6373—Control signals issued by the client directed to the server or network components for rate control, e.g. request to the server to modify its transmission rate
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/637—Control signals issued by the client directed to the server or network components
- H04N21/6377—Control signals issued by the client directed to the server or network components directed to server
- H04N21/6379—Control signals issued by the client directed to the server or network components directed to server directed to encoder, e.g. for requesting a lower encoding rate
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
- H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
- H04N21/60—Network structure or processes for video distribution between server and client or between remote clients; Control signalling between clients, server and network components; Transmission of management data between server and client, e.g. sending from server to client commands for recording incoming content stream; Communication details between server and client
- H04N21/63—Control signaling related to video distribution between client, server and network components; Network processes for video distribution between server and clients or between remote clients, e.g. transmitting basic layer and enhancement layers over different transmission paths, setting up a peer-to-peer communication via Internet between remote STB's; Communication protocols; Addressing
- H04N21/643—Communication protocols
- H04N21/6437—Real-time Transport Protocol [RTP]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
Definitions
- the present invention relates to the field of video communications, and more particularly to a method, system and terminal for adjusting video quality. Background of the invention
- Video communication is typically carried out over a dedicated or public communication network by point-to-point or point-to-multipoint transmission of video data. From the perspective of users expecting wider communication and lower cost, the public network will be the user's first choice. Due to the real-time and interactivity requirements of video communication, the network carrying video communication must be able to guarantee the smoothness and quality of the video.
- the IP network is a connectionless network using the TCP/IP protocol.
- the H.323 standard is a video communication standard developed by ITU-T in 1996 for IP networks.
- the H.323 standard is a video communication with a large number of commercial users. Standards are widely recognized and widely used.
- the IP network carrying video communication must first face the bandwidth problem. Take video chat as an example.
- the network bandwidth is required to be 128 ⁇ 256kbit/s. Due to the interactive communication, plus the load of the network transmission, an actual point-to-point video call may occupy this bandwidth. 2.5 times (320 ⁇ 640kbit/s). Therefore, video communication needs to consume a large amount of network bandwidth.
- QoS quality of service
- video communication is sensitive to delay and jitter, so video transmission generally uses the real-time User Data Packet Protocol (UDP).
- UDP User Data Packet Protocol
- RTP/RTCP real-time transport control protocol
- RTP and RTCP combine to provide flow control and congestion control for the network.
- each participant periodically transmits RTCP packets, which contain statistics such as the number of transmitted packets and the number of lost packets, which are used to determine the network status for dynamic video quality adjustment. mostly according.
- the quality control of video transmission generally has the following options:
- the minimum transmitted video coding That is, the video stream is generated by the encoder for transmission according to the minimum acceptable bandwidth.
- the main disadvantage of this solution is that video transmission lacks the flexibility to adapt to dynamic network changes.
- Transcoding technology That is, a compressed video data of sufficient quality is stored in the server.
- the server When the output bit rate needs to be reduced, the server only needs to perform partial decoding and encoding to output a suitable video data stream.
- the server stores multiple copies of different qualities of the same video program.
- the I frames of these streams are aligned, and the server selects an appropriate code stream to send to the user according to the bandwidth of the user network.
- the I frame refers to a video coding type, also called a key frame, which can be accessed without being dependent on other frames, and can be accessed at any time by another stream.
- the transcoding technology and the code stream switching technology described in the schemes 2 and 3 greatly increase the storage cost of the server in order to increase the flexibility.
- the encoder parameters are automatically adjusted according to changes in the network environment.
- Number the encoding method that adapts the code rate of the output code stream to the network change.
- the sender divides the compressed data stream into RTP protocol packets and sends them to the receiver.
- the receiver monitors the transmission delay and packet loss rate of the RTP data packet, and feeds back the network transmission status to the sender through the RTCP protocol in the RTP protocol.
- the sender adjusts the data transmission rate according to the network bandwidth to match the network environment.
- This scheme is a more flexible one of the existing methods, but it needs to estimate the current network bandwidth value and adjust the data transmission rate according to this. The quality of the video transmission will be affected when the estimation is not accurate. Summary of the invention
- the main purpose of the embodiments of the present invention is to provide a method, a system, and a terminal for adjusting video quality, without estimating the current network bandwidth value, and ensuring video transmission quality.
- the method of the present invention includes the following steps:
- the sender extracts a default level entry from a preset encoding level table corresponding to different network conditions, establishes a video connection with the parameters in the entry, and sends a real-time transmission control protocol RTCP packet;
- the receiver judges the network status according to the received RTCP packet and returns it to the sender, and the sender adjusts the current coding level entry according to the judgment result.
- embodiments of the present invention also provide a sender terminal, a receiver terminal, and a corresponding system.
- the sender terminal includes: a coding unit, configured to: encode the video stream according to the input parameter, and send the encoded video code stream to the receiver terminal; and the RTCP packet sending unit is configured to send the RTCP packet to the receiver terminal;
- the sender terminal further includes:
- An encoding level table storage unit configured to preset a coding level table
- a level entry extraction unit configured to determine a result according to the received network condition, from the Page, and output the parameters in the extracted level table item to the coding unit.
- the receiving terminal includes: a decoding unit, configured to decode the received video code stream; and an RTCP packet receiving unit, configured to receive the RTCP packet, determine the network status according to the information in the RTCP packet, and feed back the determination result to the sender terminal;
- the receiver terminal further includes:
- the second terminal capability acquiring unit is configured to acquire the terminal capability of the local end, and send the terminal capability to the sender terminal.
- a system for adjusting video quality comprising: a receiver terminal, configured to decode a received video stream, and determine a network status according to the received information in the RTCP packet, and feed back the judgment result to the sender terminal;
- the system also includes:
- the sender terminal is configured to send the encoded video code stream and the RTCP packet, and extract the corresponding level entry from the preset coding level table according to the network status judgment result fed back by the receiver terminal, and use the level entry
- the parameters in the code are encoded.
- the present invention does not need to estimate the current network bandwidth value when performing video quality adjustment, thereby avoiding the problem that the video transmission quality is affected when the estimation is not accurate, thereby ensuring the video transmission quality.
- Figure 1 is a schematic structural view of a system of the present invention
- FIG. 2 is a schematic structural diagram of a sender terminal according to the present invention.
- FIG. 3 is a schematic structural diagram of a receiver terminal according to the present invention.
- FIG. 4 is a flow chart of the method of the present invention. Mode for carrying out the invention
- the present invention provides a system for adjusting video quality, as shown in FIG.
- the system includes: a sender terminal 1 and a receiver terminal 2.
- the sender terminal 1 is configured to send the encoded video code stream and the RTCP packet, and extract a corresponding level entry from the preset coding level table according to the network status judgment result fed back by the receiver terminal 2, and The parameters in this level entry are encoded.
- the receiver terminal 2 is configured to decode the received video code stream, determine the network status according to the information in the received RTCP packet, and feed back the determination result to the sender terminal 1.
- the sender terminal 1 (ie, the sender terminal of the present invention) in the above system mainly includes: an RTCP packet sending unit 11 and a level table entry extracting unit. 12, and an encoding level table storage unit 13 and an encoding unit 14 connected to the level entry extraction unit 12;
- the RTCP packet sending unit 11 is configured to send an RTCP packet to the receiver terminal 2.
- the level entry extraction unit 12 is configured to extract a corresponding level entry from the encoding level table stored by the encoding level table storage unit 13 according to the received network status determination result, and output the parameter in the extracted level entry to the Encoding unit 14.
- the encoding level table storage unit 13 is configured to store a preset encoding level table.
- the encoding unit 14 is configured to encode the video stream according to the input parameters and send it to the receiver terminal 2.
- the sender terminal further includes a first terminal capability acquiring unit 15 and a highest video frame rate determining unit 16, which are connected to each other, and the highest video frame.
- the rate determining unit 16 is also connected to the encoding level table storage unit 13;
- the first terminal capability acquiring unit 15 is configured to acquire the terminal capability of the local end.
- the highest video frame rate determining unit 16 is configured to determine a highest video frame rate by using the terminal capability of the local end and a lower terminal capability of the terminal capability sent by the receiving terminal, and storing the high video frame rate and the encoding level table.
- Unit 13 interacts to determine the level table entries available in the encoding level table accordingly.
- the sender terminal further includes an adaptation unit 17 connected to the RTCP packet sending unit 11 for controlling the RTCP packet sending unit at an adaptive time interval. 11 Send the RTCP packet.
- the receiver terminal 2 (ie, the receiver terminal of the present invention) in the above system includes: a decoding unit 21 and an RTCP packet receiving unit 22;
- the decoding unit 21 is configured to decode the received video code stream.
- the RTCP packet receiving unit 22 is configured to receive the RTCP packet, determine the network status according to the information in the RTCP packet, and feed back the judgment result to the sender terminal 1.
- the receiving terminal further includes a second terminal capability acquiring unit 23, configured to acquire the terminal capability of the local end, and send the terminal capability to the transmitting terminal 1.
- FIG. 4 is a flow chart of the method of the present invention. Referring to FIG. 4, the following main steps are included:
- a coding level table is preset in the sender terminal.
- the coding level table includes several level entries to suit different network conditions. The higher the level entry, the better the network condition is applicable to each parameter value. On the contrary, the lower the level entry, the worse the network condition of each parameter value is.
- each level entry in the encoding level table can include the following parameters: Key frame I frame interval, quality priority or speed priority, quality priority bandwidth, quality priority frame rate, speed priority bandwidth, and speed priority frame rate; in addition, to increase rate control
- the flexibility of each level table can also include: horizontal scaling of the video window, and vertical scaling of the video window.
- Level 9 entry ⁇ 100, 0, 300, 12, 200, 13, 4/5, 1 ⁇ means that at level 9, the encoder will insert an I frame every 100 frames, select the speed priority mode.
- the bandwidth and frame rate under quality priority are 300k and 12 frames respectively
- the bandwidth and frame rate in speed priority mode are 200k and 13 frames respectively
- the horizontal scaling of video window is 4/5
- the vertical scaling of video window is 1. .
- the H.264 standard can increase coding efficiency by about 50%, but its terminal capability requirements will also increase exponentially. If the decoding capability of the receiver terminal is insufficient, the encoding frame rate of the sender terminal must be controlled to prevent the receiver terminal from dropping the frame due to decoding, resulting in a video that is not smooth.
- the terminal capabilities of both senders and receivers in order to achieve better video effects, Before establishing a video connection, you can choose to detect the terminal capabilities of both the sender and the receiver.
- the highest video frame rate is determined by the lower terminal capability of the terminal capabilities of the transmitting and receiving parties, and the level table item available in the encoding level table is determined correspondingly by the highest video frame rate.
- the terminal capability is mainly reflected in the processing power of the CPU.
- the processing power of the CPU is mainly determined by the CPU frequency, but in addition to the main frequency, other factors (such as the Cache size) also have an impact on the processing power of the CPU. Therefore, the highest video frame rate is correspondingly based on the CPU main frequency, for example: 2700 MHz main frequency corresponds to 15 frames, and 400 MHz main frequency corresponds to 1 frame. Further, comprehensively consider the characteristics of the CPU type and the size of the first-level cache Cache to correct the highest corresponding video frame rate, for example: For the AMD chip, the level 2 Cache size exceeds 128 and the frame rate is increased by 1, more than 256. The frame rate is increased by 2.
- the default level entry is extracted from the preset encoding level table by using the highest video frame rate.
- the default level entry should select a lower level entry to ensure that the video connection is successfully established.
- the sender terminal After the video connection is established, the sender terminal sends an RTCP packet to the receiver terminal.
- the transmitted RTCP packet contains information such as packet loss rate, network jitter, and delay time.
- This step can further adopt a method of transmitting an RTCP packet at a certain length of time.
- the time interval for sending packets is shorter to quickly determine the network status.
- the time interval for sending packets should be gradually lengthened to reduce the network load caused by the RTCP packet. Therefore, the present invention preferably transceives RTCP packets at adaptive time intervals.
- the receiver terminal After receiving the RTCP packet sent by the sender terminal, the receiver terminal obtains the relevant information, and if the received packet loss rate, network jitter, and delay time in the RTCP packet are less than the corresponding threshold, the network condition is determined to be good. Otherwise, it is determined that the network is not in good condition.
- the receiver terminal feeds back the network status judgment result to the sender terminal. If the judgment result received by the sender terminal indicates that the current network condition is good, and there is room for improving the video quality and fluency, the current coding level entry is used to be encoded, thereby generating a higher quality video bitstream, and receiving If the judgment result received by the sender terminal indicates that the current network status is not good, that is, the currently used coding level entry has exceeded the adaptability of the current network condition, the current coding level entry is lowered. The code generates a video stream of lower quality and transmits it to the receiver terminal.
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Abstract
Description
调节视频质量的方法、 系统及终端 技术领域 Method, system and terminal for adjusting video quality
本发明涉及视频通信领域, 尤指一种调节视频质量的方法、 系统及 终端。 发明背景 The present invention relates to the field of video communications, and more particularly to a method, system and terminal for adjusting video quality. Background of the invention
随着互联网的迅猛发展以及多媒体技术的日益成熟, 视频通信的市 场应用越来越广泛, 实现视频通信的方式也变得多种多样,如可视电话、 即时通信、 视频聊天、 IPTV、 远程监控, 以及远程医疗等。 视频通信发 展的广阔前景使其成为继话音通信之后又一重要的通信形式。 With the rapid development of the Internet and the maturity of multimedia technology, the market for video communication is becoming more and more widely used, and the ways to realize video communication have become various, such as videophone, instant messaging, video chat, IPTV, remote monitoring. , as well as telemedicine. The broad prospects for video communication development make it another important form of communication following voice communication.
视频通信一般是在专用的或公用的通信网络上, 通过点对点或点对 多点来传送视频数据的。 从用户期望更广泛的交流和更低的成本的角度 来看, 公用网络将是用户的首选。 由于视频通信对实时性和交互性的要 求, 承载视频通信的网络必须能够保证视频的流畅性和质量。 Video communication is typically carried out over a dedicated or public communication network by point-to-point or point-to-multipoint transmission of video data. From the perspective of users expecting wider communication and lower cost, the public network will be the user's first choice. Due to the real-time and interactivity requirements of video communication, the network carrying video communication must be able to guarantee the smoothness and quality of the video.
目前, 通信网络的 IP化成为网络发展的主流, 基于 IP网络的视频 通信得到了普遍应用和广泛发展。 IP网络是使用 TCP/IP协议的面向无 连接的网络, H.323标准是 ITU-T在 1996年制定的应用于 IP网络的视 频通信标准, H.323 标准是目前拥有大量商业用户的视频通信标准, 得 到了普遍的认可和广泛的应用。 At present, the IP of communication networks has become the mainstream of network development, and video communication based on IP networks has been widely applied and widely developed. The IP network is a connectionless network using the TCP/IP protocol. The H.323 standard is a video communication standard developed by ITU-T in 1996 for IP networks. The H.323 standard is a video communication with a large number of commercial users. Standards are widely recognized and widely used.
承载视频通信的 IP 网络首先要面对的是带宽问题, 以视频聊天为 例, 根据经验数值, 使用流行的 H.263编码标准, 要保证可接受的质量 和流畅度, 需要网络带宽为 128 ~ 256kbit/s。 由于是交互式通信, 加上 网络传送的负荷, 一个实际的点对点的视频通话, 可能要占据此带宽的 2.5倍( 320 ~ 640kbit/s )。 因此, 视频通信需要消耗大量的网络带宽, 要 在各种网络下如互联网、 局域网保障服务质量(QoS )参数, 就需要通 过视频质量调节来进行带宽控制。 The IP network carrying video communication must first face the bandwidth problem. Take video chat as an example. According to the empirical value, using the popular H.263 coding standard, to ensure acceptable quality and smoothness, the network bandwidth is required to be 128 ~ 256kbit/s. Due to the interactive communication, plus the load of the network transmission, an actual point-to-point video call may occupy this bandwidth. 2.5 times (320 ~ 640kbit/s). Therefore, video communication needs to consume a large amount of network bandwidth. To ensure quality of service (QoS) parameters under various networks, such as the Internet and the local area network, it is necessary to perform bandwidth control through video quality adjustment.
其次, 视频通信对时延和抖动 4艮敏感, 因此视频传输一般都采用实 时性高的用户数据包协议(UDP )。 但由于 UDP协议没有任何拥塞控制 算法, 因此容易造成网络过载和高丢包率, 从而影响视频的传输质量。 因此, 视频通信还需要实时传输协议 /实时传输控制协议( RTP/RTCP ) 的支持, RTP与 RTCP结合可以为网络提供流量控制和拥塞控制。在 RTP 会话期间, 各参与者周期性地传送 RTCP包, RTCP包中含有已发送的 数据包的数量、 丢失的数据包的数量等统计资料, 这些信息是判断网络 状况以进行动态视频质量调节的主要依据。 Second, video communication is sensitive to delay and jitter, so video transmission generally uses the real-time User Data Packet Protocol (UDP). However, since the UDP protocol does not have any congestion control algorithm, it is easy to cause network overload and high packet loss rate, thus affecting the video transmission quality. Therefore, video communication also requires real-time transport protocol/real-time transport control protocol (RTP/RTCP) support. RTP and RTCP combine to provide flow control and congestion control for the network. During the RTP session, each participant periodically transmits RTCP packets, which contain statistics such as the number of transmitted packets and the number of lost packets, which are used to determine the network status for dynamic video quality adjustment. mostly according.
目前, 视频传输的质量控制一般有以下几种方案: At present, the quality control of video transmission generally has the following options:
1、 最小传输的视频编码。 即按照最低可接受的带宽由编码器产生 视频码流进行传输。 该方案的主要缺点是视频传输缺少适应网络动态变 化的灵活性。 1. The minimum transmitted video coding. That is, the video stream is generated by the encoder for transmission according to the minimum acceptable bandwidth. The main disadvantage of this solution is that video transmission lacks the flexibility to adapt to dynamic network changes.
2、 转码技术。 即在服务器中保存一份质量足够好的压缩视频数据, 当需要降低输出码率时, 服务器只需进行部分的解码和编码就可以输出 合适的视频数据流。 2. Transcoding technology. That is, a compressed video data of sufficient quality is stored in the server. When the output bit rate needs to be reduced, the server only needs to perform partial decoding and encoding to output a suitable video data stream.
3、 码流切换技术。 服务器中保存同一个视频节目的不同质量的多 个拷贝, 这些码流的 I帧是对齐的, 服务器根据用户网络带宽情况选择 一个适当的码流发送给用户。 其中, I帧是指一种视频编码类型, 也叫 关键帧,它可以不依赖于别的帧解码,通过它可以随时接入另一个码流。 3. Code stream switching technology. The server stores multiple copies of different qualities of the same video program. The I frames of these streams are aligned, and the server selects an appropriate code stream to send to the user according to the bandwidth of the user network. The I frame refers to a video coding type, also called a key frame, which can be accessed without being dependent on other frames, and can be accessed at any time by another stream.
方案 2和方案 3所述的转码技术和码流切换技术, 其为了增加灵活 性而大大提高了服务器的存储成本。 The transcoding technology and the code stream switching technology described in the schemes 2 and 3 greatly increase the storage cost of the server in order to increase the flexibility.
4、 自适应视频编码。 即根据网络环境的变化来自动调整编码器的参 数, 使得输出码流的码率适应网络变化的编码方法。 一般的做法是发送 方将压缩数据流划分成 RTP协议包发送给接收方, 接收方监测 RTP数据 包的传输时延和丢包率, 通过 RTP协议中的 RTCP协议将网络传输情况 反馈给发送方, 发送方根据网络带宽情况调整数据发送码率使之与网络 环境相匹配。 该方案是现有方法中较为灵活的一种方案, 但其需要估计 当前的网络带宽值, 并据此调整数据发送码率, 当估计不准时视频传输 的质量将受到影响。 发明内容 4. Adaptive video coding. That is, the encoder parameters are automatically adjusted according to changes in the network environment. Number, the encoding method that adapts the code rate of the output code stream to the network change. Generally, the sender divides the compressed data stream into RTP protocol packets and sends them to the receiver. The receiver monitors the transmission delay and packet loss rate of the RTP data packet, and feeds back the network transmission status to the sender through the RTCP protocol in the RTP protocol. The sender adjusts the data transmission rate according to the network bandwidth to match the network environment. This scheme is a more flexible one of the existing methods, but it needs to estimate the current network bandwidth value and adjust the data transmission rate according to this. The quality of the video transmission will be affected when the estimation is not accurate. Summary of the invention
有鉴于此, 本发明实施例的主要目的在于提供一种调节视频质量的 方法、 系统及终端, 无需估计当前的网络带宽值, 保证视频传输质量。 In view of this, the main purpose of the embodiments of the present invention is to provide a method, a system, and a terminal for adjusting video quality, without estimating the current network bandwidth value, and ensuring video transmission quality.
为达到上述目的, 本发明实施例的技术方案具体是这样实现的: 本发明方法包括下列步骤: To achieve the above objective, the technical solution of the embodiment of the present invention is specifically implemented as follows: The method of the present invention includes the following steps:
发送方从预设的对应不同的网络状况的编码级别表中提取默认级 别表项, 以该表项中的参数建立视频连接并发送实时传输控制协议 RTCP包; The sender extracts a default level entry from a preset encoding level table corresponding to different network conditions, establishes a video connection with the parameters in the entry, and sends a real-time transmission control protocol RTCP packet;
接收方根据接收到的 RTCP包判断网络状况并返回给发送方, 发送 方根据判断结果调整当前编码级别表项。 The receiver judges the network status according to the received RTCP packet and returns it to the sender, and the sender adjusts the current coding level entry according to the judgment result.
为了支撑本发明方法, 本发明实施例还提供了发送方终端、 接收方 终端和相应的系统。 In order to support the method of the present invention, embodiments of the present invention also provide a sender terminal, a receiver terminal, and a corresponding system.
其中, 发送方终端包括: 编码单元, 用于根据输入的参数, 将视频 流编码, 并向接收方终端发送编码后的视频码流; RTCP 包发送单元, 用于向接收方终端发送 RTCP包; 所述发送方终端还包括: The sender terminal includes: a coding unit, configured to: encode the video stream according to the input parameter, and send the encoded video code stream to the receiver terminal; and the RTCP packet sending unit is configured to send the RTCP packet to the receiver terminal; The sender terminal further includes:
编码级别表存储单元, 用于预设编码级别表; An encoding level table storage unit, configured to preset a coding level table;
级别表项提取单元, 用于根据收到的网络状况判断结果, 从所述编 页, 并将提取 的级别表项中的参数输出到编码单元。 a level entry extraction unit, configured to determine a result according to the received network condition, from the Page, and output the parameters in the extracted level table item to the coding unit.
接收方终端包括: 解码单元, 用于解码收到的视频码流; RTCP 包 接收单元,用于接收 RTCP包, 并根据 RTCP包中的信息判断网络状况, 并将判断结果反馈给发送方终端; 所述接收方终端还包括: The receiving terminal includes: a decoding unit, configured to decode the received video code stream; and an RTCP packet receiving unit, configured to receive the RTCP packet, determine the network status according to the information in the RTCP packet, and feed back the determination result to the sender terminal; The receiver terminal further includes:
第二终端能力获取单元, 用于获取本端的终端能力, 并发送给发送 方终端。 The second terminal capability acquiring unit is configured to acquire the terminal capability of the local end, and send the terminal capability to the sender terminal.
调节视频质量的系统, 包括: 接收方终端, 用于对接收到的视频码 流解码, 以及^ ^据收到的 RTCP包中的信息判断网络状况, 并将判断结 果反馈给发送方终端; 所述系统还包括: a system for adjusting video quality, comprising: a receiver terminal, configured to decode a received video stream, and determine a network status according to the received information in the RTCP packet, and feed back the judgment result to the sender terminal; The system also includes:
发送方终端, 用于发送编码后的视频码流及 RTCP包, 以及根据接 收方终端反馈的网络状况判断结果, 从预设的编码级别表中提取相应的 级别表项, 并以该级别表项中的参数进行编码。 The sender terminal is configured to send the encoded video code stream and the RTCP packet, and extract the corresponding level entry from the preset coding level table according to the network status judgment result fed back by the receiver terminal, and use the level entry The parameters in the code are encoded.
由上述技术方案可见, 本发明在进行视频质量调节时, 无需估计当 前的网络带宽值, 从而避免了当估计不准时, 视频传输质量受到影响的 问题, 从而保证了视频传输质量。 It can be seen from the above technical solution that the present invention does not need to estimate the current network bandwidth value when performing video quality adjustment, thereby avoiding the problem that the video transmission quality is affected when the estimation is not accurate, thereby ensuring the video transmission quality.
附图简要说明 BRIEF DESCRIPTION OF THE DRAWINGS
图 1为本发明系统的结构示意图; Figure 1 is a schematic structural view of a system of the present invention;
图 2为本发明发送方终端的结构示意图; 2 is a schematic structural diagram of a sender terminal according to the present invention;
图 3为本发明接收方终端的结构示意图; 3 is a schematic structural diagram of a receiver terminal according to the present invention;
图 4为本发明方法的流程图。 实施本发明的方式 Figure 4 is a flow chart of the method of the present invention. Mode for carrying out the invention
为了在进行视频质量调节时, 无需估计当前的网络带宽值, 从而避 免当估计不准时, 视频传输质量受到影响的问题, 本发明提供了一种调 节视频质量的系统, 参见图 1所示, 该系统包括: 发送方终端 1和接收 方终端 2。 In order to perform video quality adjustment, it is not necessary to estimate the current network bandwidth value, thereby avoiding the problem that the video transmission quality is affected when the estimation is not accurate. The present invention provides a system for adjusting video quality, as shown in FIG. The system includes: a sender terminal 1 and a receiver terminal 2.
其中, 发送方终端 1 , 用于发送编码后的视频码流及 RTCP包, 以 及根据接收方终端 2反馈的网络状况判断结果, 从预设的编码级别表中 提取相应的级别表项, 并以该级别表项中的参数进行编码。 The sender terminal 1 is configured to send the encoded video code stream and the RTCP packet, and extract a corresponding level entry from the preset coding level table according to the network status judgment result fed back by the receiver terminal 2, and The parameters in this level entry are encoded.
所述接收方终端 2, 用于对接收到的视频码流解码, 以及根据收到 的 RTCP包中的信息判断网络状况,并将判断结果反馈给发送方终端 1。 The receiver terminal 2 is configured to decode the received video code stream, determine the network status according to the information in the received RTCP packet, and feed back the determination result to the sender terminal 1.
图 2为本发明发送方终端的结构示意图, 参见图 2所示, 上述系统 中的发送方终端 1 (即本发明的发送方终端)中主要包括: RTCP包发送 单元 11、 级别表项提取单元 12, 以及与级别表项提取单元 12相连的编 码级别表存储单元 13和编码单元 14; 其中, 2 is a schematic structural diagram of a sender terminal according to the present invention. Referring to FIG. 2, the sender terminal 1 (ie, the sender terminal of the present invention) in the above system mainly includes: an RTCP packet sending unit 11 and a level table entry extracting unit. 12, and an encoding level table storage unit 13 and an encoding unit 14 connected to the level entry extraction unit 12;
RTCP包发送单元 11 , 用于向接收方终端 2发送 RTCP包。 The RTCP packet sending unit 11 is configured to send an RTCP packet to the receiver terminal 2.
级别表项提取单元 12, 用于根据收到的网络状况判断结果, 从编码 级别表存储单元 13存储的编码级别表中提取相应的级别表项, 并将提 取的级别表项中的参数输出到编码单元 14。 The level entry extraction unit 12 is configured to extract a corresponding level entry from the encoding level table stored by the encoding level table storage unit 13 according to the received network status determination result, and output the parameter in the extracted level entry to the Encoding unit 14.
编码级别表存储单元 13, 用于存储预设的编码级别表。 The encoding level table storage unit 13 is configured to store a preset encoding level table.
编码单元 14, 用于根据输入的参数, 将视频流编码, 并向接收方终 端 2发送。 The encoding unit 14 is configured to encode the video stream according to the input parameters and send it to the receiver terminal 2.
进一步, 为了减小收发双方的终端能力对视频质量和流畅度的影 响, 所述发送方终端还包括相互连接的第一终端能力获取单元 15 和最 高视频帧率确定单元 16, 所述最高视频帧率确定单元 16还与编码级别 表存储单元 13相连; 其中, 第一终端能力获取单元 15, 用于获取本端的终端能力。 Further, in order to reduce the impact of the terminal capabilities of the transmitting and receiving parties on the video quality and the fluency, the sender terminal further includes a first terminal capability acquiring unit 15 and a highest video frame rate determining unit 16, which are connected to each other, and the highest video frame. The rate determining unit 16 is also connected to the encoding level table storage unit 13; The first terminal capability acquiring unit 15 is configured to acquire the terminal capability of the local end.
最高视频帧率确定单元 16,用于以本端的终端能力和接收方终端发 来的终端能力中较低的终端能力确定最高视频帧率, 并以该高视频帧率 与所述编码级别表存储单元 13 交互, 相应确定编码级别表中可用的级 别表项。 The highest video frame rate determining unit 16 is configured to determine a highest video frame rate by using the terminal capability of the local end and a lower terminal capability of the terminal capability sent by the receiving terminal, and storing the high video frame rate and the encoding level table. Unit 13 interacts to determine the level table entries available in the encoding level table accordingly.
进一步, 为了减少收发 RTCP包对网络负载的影响, 所述发送方终 端还包括与所述 RTCP包发送单元 11相连的自适应单元 17, 用于以自 适应的时间间隔控制所述 RTCP包发送单元 11发送 RTCP包。 Further, in order to reduce the impact of the RTCP packet on the network load, the sender terminal further includes an adaptation unit 17 connected to the RTCP packet sending unit 11 for controlling the RTCP packet sending unit at an adaptive time interval. 11 Send the RTCP packet.
图 3为本发明接收方终端的结构示意图, 参见图 3所示, 上述系统 中的接收方终端 2 (即本发明的接收方终端) 中包括: 解码单元 21 和 RTCP包接收单元 22; 其中, 3 is a schematic structural diagram of a receiver terminal according to the present invention. Referring to FIG. 3, the receiver terminal 2 (ie, the receiver terminal of the present invention) in the above system includes: a decoding unit 21 and an RTCP packet receiving unit 22;
解码单元 21 , 用于将收到的视频码流解码。 The decoding unit 21 is configured to decode the received video code stream.
RTCP包接收单元 22, 用于接收 RTCP包, 并根据 RTCP包中的信 息判断网络状况, 以及将判断结果反馈给发送方终端 1。 The RTCP packet receiving unit 22 is configured to receive the RTCP packet, determine the network status according to the information in the RTCP packet, and feed back the judgment result to the sender terminal 1.
进一步地, 为了减少收发双方的终端能力对视频质量和流畅度的影 响, 所述接收方终端还包括第二终端能力获取单元 23, 用于获取本端的 终端能力, 并向发送方终端 1发送。 Further, in order to reduce the impact of the terminal capability of the transmitting and receiving parties on the video quality and the fluency, the receiving terminal further includes a second terminal capability acquiring unit 23, configured to acquire the terminal capability of the local end, and send the terminal capability to the transmitting terminal 1.
应用上述系统及终端, 本发明提供了一种调节视频质量的方法, 图 4为本发明方法的流程图, 参见图 4所示, 包括下列主要步骤: Applying the above system and terminal, the present invention provides a method for adjusting video quality, and FIG. 4 is a flow chart of the method of the present invention. Referring to FIG. 4, the following main steps are included:
S 1、 从预设的编码级别表中提取默认级别表项。 S1. Extract a default level entry from a preset encoding level table.
首先, 在发送方终端中预设编码级别表。 该编码级别表中包括若干 级别表项, 以适应不同的网络状况。 级别表项越高, 其中的各参数数值 适用的网络状况越好; 相反, 级别表项越低, 其中的各参数数值适应的 网络状况越差。 First, a coding level table is preset in the sender terminal. The coding level table includes several level entries to suit different network conditions. The higher the level entry, the better the network condition is applicable to each parameter value. On the contrary, the lower the level entry, the worse the network condition of each parameter value is.
参见表一所示, 编码级别表中的每一级别表项可以包括下列参数: 关键帧 I帧间隔、 质量优先或速度优先、 质量优先情况下的带宽、 质量 优先情况下的帧率、 速度优先情况下的带宽, 以及速度优先情况下的帧 率; 此外, 为了增加码率控制的灵活性, 每一级别表项中还可以包括: 视频窗口横向缩放比例, 以及视频窗口纵向缩放比例。 例如: 第 9级表 项 = { 100, 0, 300, 12, 200, 13, 4/5, 1 } 表示在第 9级上,编码器将每隔 100 帧插入一个 I帧, 选择速度优先方式, 质量优先情况下的带宽和帧率分 别为 300k和 12帧,速度优先方式下的带宽和帧率分别为 200k和 13帧, 视频窗口横向缩放比例为 4/5 , 视频窗口纵向缩放比例为 1。As shown in Table 1, each level entry in the encoding level table can include the following parameters: Key frame I frame interval, quality priority or speed priority, quality priority bandwidth, quality priority frame rate, speed priority bandwidth, and speed priority frame rate; in addition, to increase rate control The flexibility of each level table can also include: horizontal scaling of the video window, and vertical scaling of the video window. For example: Level 9 entry = { 100, 0, 300, 12, 200, 13, 4/5, 1 } means that at level 9, the encoder will insert an I frame every 100 frames, select the speed priority mode. The bandwidth and frame rate under quality priority are 300k and 12 frames respectively, the bandwidth and frame rate in speed priority mode are 200k and 13 frames respectively, the horizontal scaling of video window is 4/5, and the vertical scaling of video window is 1. .
表一 Table I
其次, 可选择检测收发双方的终端能力。 Second, you can choose to detect the terminal capabilities of both the sender and the receiver.
H.264标准与传统的编码标准相比, 编码效率能够提高 50%左右, 但其对终端能力的要求也将成倍地增加。 如果接收方终端的解码能力不 足, 就必须控制发送方终端的编码帧率, 以免接收方终端来不及解码而 丢帧, 造成视频的不流畅。 Compared with the traditional coding standard, the H.264 standard can increase coding efficiency by about 50%, but its terminal capability requirements will also increase exponentially. If the decoding capability of the receiver terminal is insufficient, the encoding frame rate of the sender terminal must be controlled to prevent the receiver terminal from dropping the frame due to decoding, resulting in a video that is not smooth.
因此, 为了顾及收发双方的终端能力, 以达到更好的视频效果, 在 建立视频连接之前, 可选择检测收发双方的终端能力。 以收发双方的终 端能力中较低的终端能力确定最高视频帧率, 并以该最高视频帧率为 限, 相应确定编码级别表中可用的级别表项。 Therefore, in order to take into account the terminal capabilities of both senders and receivers, in order to achieve better video effects, Before establishing a video connection, you can choose to detect the terminal capabilities of both the sender and the receiver. The highest video frame rate is determined by the lower terminal capability of the terminal capabilities of the transmitting and receiving parties, and the level table item available in the encoding level table is determined correspondingly by the highest video frame rate.
在视频流编解码过程中, 终端能力主要体现在 CPU 的处理能力。 In the video stream encoding and decoding process, the terminal capability is mainly reflected in the processing power of the CPU.
CPU的处理能力主要由 CPU主频决定, 但除主频之外, 其他因素 (如 Cache大小 )也对 CPU的处理能力有一定影响。 因此根据 CPU主频初 步对应出最高视频帧率, 例如: 2700MHZ主频对应 15帧, 400MHZ主 频对应 1 帧。 进一步, 再综合考虑 CPU类型和一二级高速緩存 Cache 大小等特性, 以修正初步对应出的最高视频帧率, 例如: 对于 AMD芯 片, 2级 Cache大小超过 128则帧率加 1 , 超过 256则帧率加 2。 The processing power of the CPU is mainly determined by the CPU frequency, but in addition to the main frequency, other factors (such as the Cache size) also have an impact on the processing power of the CPU. Therefore, the highest video frame rate is correspondingly based on the CPU main frequency, for example: 2700 MHz main frequency corresponds to 15 frames, and 400 MHz main frequency corresponds to 1 frame. Further, comprehensively consider the characteristics of the CPU type and the size of the first-level cache Cache to correct the highest corresponding video frame rate, for example: For the AMD chip, the level 2 Cache size exceeds 128 and the frame rate is increased by 1, more than 256. The frame rate is increased by 2.
最后, 以所述最高视频帧率为限, 从预设的编码级别表中提取默认 级别表项, 所述默认级别表项应选择较低的级别表项, 以保证顺利建立 视频连接。 Finally, the default level entry is extracted from the preset encoding level table by using the highest video frame rate. The default level entry should select a lower level entry to ensure that the video connection is successfully established.
52、 以默认级别表项中的参数建立视频连接。 52. Establish a video connection with the parameters in the default level entry.
53、 发送 RTCP包。 53. Send the RTCP packet.
建立视频连接后, 发送方终端向接收方终端发送 RTCP包。 所述发 送的 RTCP包中含有丢包率、 网络抖动和延迟时间等信息。 After the video connection is established, the sender terminal sends an RTCP packet to the receiver terminal. The transmitted RTCP packet contains information such as packet loss rate, network jitter, and delay time.
本步骤可进一步采用每隔一定的时长发送一个 RTCP包的方式。 当视频连接刚建立时, 发包的时间间隔较短为宜, 以快速判断网络 状况; 而在视频连接稳定之后, 发包的时间间隔应逐渐加长, 以减少 RTCP 包带来的网络负载。 所以本发明优选以自适应的时间间隔收发 RTCP包。 所述自适应的时间间隔通过以下公式计算: 发送 RTCP包的 时间间隔= (视频已建立的时间) X ( EXP ( I 帧间隔时间 /125 ) -1 ), 即所述发送 RTCP包的时间间隔为: 计算 I帧间隔时间与常数 125之商 的指数, 所得的值减常数一得到差值, 再计算所得差值与与视频已建立 的时间之积。 This step can further adopt a method of transmitting an RTCP packet at a certain length of time. When the video connection is just established, the time interval for sending packets is shorter to quickly determine the network status. After the video connection is stable, the time interval for sending packets should be gradually lengthened to reduce the network load caused by the RTCP packet. Therefore, the present invention preferably transceives RTCP packets at adaptive time intervals. The adaptive time interval is calculated by the following formula: Time interval for transmitting RTCP packets = (time when video has been established) X ( EXP (I frame interval time / 125 ) -1 ), that is, the time interval for transmitting the RTCP packets For: Calculate the exponent of the quotient of the I frame interval time and the constant 125, and obtain the difference between the obtained value and the constant, and then calculate the difference and the video has been established. The product of time.
S4、 接收 RTCP包, 以判断网络状况。 S4. Receive an RTCP packet to determine the network status.
接收方终端收到发送方终端发来的 RTCP包后,解包获取相关信息, 若收到的 RTCP包中含有的丢包率、 网络抖动和延迟时间同时小于相应 的阈值时, 判定网络状况良好; 否则, 判定网络状况不好。 After receiving the RTCP packet sent by the sender terminal, the receiver terminal obtains the relevant information, and if the received packet loss rate, network jitter, and delay time in the RTCP packet are less than the corresponding threshold, the network condition is determined to be good. Otherwise, it is determined that the network is not in good condition.
S5、 根据判断结果相应调整当前编码级别表项。 S5. Adjust the current coding level entry according to the judgment result.
接收方终端将网络状况判断结果反馈给发送方终端。 若发送方终端 收到的判断结果表明当前网络状况良好, 还有提升视频质量和流畅度的 余地, 则调高当前所用的编码级别表项, 从而编码生成质量更高的视频 码流, 向接收方终端发送; 若发送方终端收到的判断结果表明当前网络 状况不好, 即当前所用的编码级别表项已经超过了当前网络状况的适应 能力, 则调低当前所用的编码级别表项, 从而编码生成质量较低的视频 码流, 向接收方终端发送。 The receiver terminal feeds back the network status judgment result to the sender terminal. If the judgment result received by the sender terminal indicates that the current network condition is good, and there is room for improving the video quality and fluency, the current coding level entry is used to be encoded, thereby generating a higher quality video bitstream, and receiving If the judgment result received by the sender terminal indicates that the current network status is not good, that is, the currently used coding level entry has exceeded the adaptability of the current network condition, the current coding level entry is lowered. The code generates a video stream of lower quality and transmits it to the receiver terminal.
以上所述, 仅为本发明的较佳实施例而已, 并非用于限定本发明的 保护范围, 凡在本发明的精神和原则之内所做的任何修改、 等同替换、 改进等, 均应包含在本发明的保护范围之内。 The above is only the preferred embodiment of the present invention, and is not intended to limit the scope of the present invention. Any modifications, equivalents, improvements, etc., which are made within the spirit and principles of the present invention, should be included. It is within the scope of the invention.
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