CN1614994A - Audio and visual frequencies synchronizing method for IP network conference - Google Patents
Audio and visual frequencies synchronizing method for IP network conference Download PDFInfo
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Abstract
The method includes following steps: the data channels are separately built up for video data and audio data; in the data channel the separate thread is used to collect, compress, package, transmit, receive, unpackgae, uncompress the data. When coding and packaging, the audio data package and its relevant video data package are encoded with same sequence number so that at the reception end the audio and video data can be displayed according to sequence numbers; in the displaying procedure, when the video and audio data can not keep the consistent of sequence number, the sequence number of audio data is taken as norm.
Description
Technical Field
The invention relates to a method for processing audio and video data in an IP network environment, in particular to a method for synchronizing video and audio of an IP network conference.
Technical Field
With the development of computer and communication technologies, multimedia communication technologies have become a hot spot of research. In the technology, how to correctly recover the time relationship among a plurality of media at the destination, that is, to realize the synchronization among the media, becomes one of the key technologies for realizing the multimedia communication.
In multimedia network communication, synchronization refers to temporal relationships existing among all media objects in a system and inside the media objects, and exists in a series of processes of acquiring, storing, transmitting, presenting and the like of multimedia information, wherein the synchronization problem in transmission is the most complicated due to instability of network conditions.
The essence of synchronization is that the various media remain constrained in terms of time and space after transmission. In a distributed multimedia communication network, multimedia synchronization includes two types: intra-stream synchronization and inter-stream synchronization.
Intra-stream synchronization refers to the temporal relationship that exists between media units within a single media stream. For example, it is ensured that the transmitted audio is played at the receiving end in a time relationship, and the problem of "out of order" that the transmitted audio is played first does not occur, which means in-stream synchronization. The in-stream synchronization is relatively easy to realize, the simplest method is to number data packets to be sent in sequence, and the in-stream synchronization can be realized only by setting a buffer area at a receiving end and unpacking and playing the data packets in sequence strictly according to the sequence numbers of the data packets.
The inter-stream synchronization refers to a temporal relationship existing among the basic media units of a plurality of related media streams, that is, various data such as video, audio and the like collected at the same time are still played at the same time at a receiving end. The video and audio synchronization technique we studied is inter-stream synchronization. The synchronization between streams is more complex to achieve than the intra-stream co-frequency.
In a conference system, sound and video data are sequentially subjected to the processing processes of data sampling, data compression, packaging, data packet sending, data packet receiving, data packet unpacking, decompression and playback.
Through the processing of the above links, the audio and video signals captured at the same time are often out of synchronization when the audio and video signals are played back by the receiving terminal. According to the above process, the reasons for causing the asynchronism can be divided into two types: firstly, the terminal processes the time delay caused by data; second, network transmission delay. The time delay caused by the terminal comprises the time difference caused by the difference of the data volume of audio and video and the encoding algorithm when the sending terminal processes the modules of sampling, compressing, packaging and the like and when the receiving terminal processes the modules of unpacking, decompressing, playback and the like. The network transmission delay is affected by many factors such as the real-time transmission bandwidth of the network, the transmission distance, and the processing speed of the network node.
In fact, the delay caused by the terminal processing procedure is fixed, can be estimated, and is easier to solve. What is really difficult to solve is the problem that the network delay changes randomly according to the network condition, thereby causing asynchronism.
In order to solve the audio and video frequency problem, a great deal of related research is carried out by experts. At present, a widely used method is a block packing transmission method. The main idea of the block packaging transmission method is as follows:
A. at a sending end, video data and audio data to be sent are respectively sampled at intervals of time T (T is small).
B. And directly printing the video and audio data obtained at the same time into the same packet. I.e. the video and audio data sampled each time are packed in the same packet, and the packets are numbered.
C. And transmitting each data packet respectively.
D. And arranging a buffer area at the receiving end, sequencing the received packets according to the numbers and playing the packets. Since the video and audio are in the same packet, the synchronization problem is naturally solved.
However, there are many disadvantages to the block and packet transmission method. The most important disadvantages are:
first, it is a method for transmitting end control, which makes the transmitting end have a large load. For an IP network conference, a large amount of audio and video acquisition is often performed at a sending end. The block packing transmission method requires that a sending end host not only undertakes the work of data acquisition, compression, sending to each receiving end and the like, but also takes charge of synchronous control, which is a high requirement on computer hardware participating in a network conference.
Secondly, stability and real-time are not high. The block packing transmission method is to pack audio and video data into the same packet, so that each data packet is very large, which results in a large data volume, and when the data is transmitted on the network, the data is likely to have more packet loss, delay and other problems due to unstable network conditions, which is not favorable for the stability and real-time performance of the system.
Third, the data buffer at the receiving end must be large. Since each data packet is large and the time required for each packet to reach the receiving end is variable, this may require the receiving end to have a large data buffer to store the "send-after-first" data packets while waiting for the "send-after-first" data packets.
In view of the above disadvantages of the block packaging method, in practical use, partial improvement is made on the block packaging method according to practical situations. The main measures for improvement are:
A. the amount of data is reduced. In order to avoid the situation that the sending end is too busy due to too large data volume, the sending of the data volume can be considered to be reduced, which is at the expense of the quality of the IP network conference. The methods for reducing the data volume mainly include: a. the acquisition frequency is reduced-this may result in audio and video discontinuities. b. The capture of video is reduced while maintaining continuity of audio. In IP netconferencing, if the audio is intermittent, it is difficult for people to receive, and video discontinuities seem to be tolerable. For this reason, it is considered to reduce the acquisition of video, increase the compression ratio of video data (at the expense of picture quality), and reduce the data amount.
B. And opening up multiple transmission and multiple transmission channels. Under the condition that the hardware of the computer at the sending end is good enough, the method can be considered in order to overcome the problems of data sending caused by over busy sending threads and unstable network conditions of the sending end.
Opening up multiple routes of transmission is to create multiple transmission threads at the transmitting end, and each thread is responsible for transmitting data to several corresponding receiving ends. The multipath transmission channel means that two or more virtual connection channels are created between a sending end and a receiving end in a connection-oriented computer network, and data to be sent can be transmitted through different channels. This increases the system cost.
Therefore, the improved block packing method is only a change and improvement of the packing method under different practical conditions and different practical requirements. But its implementation is achieved at the expense of some other benefit or cost.
Disclosure of Invention
The purpose of the invention is: aiming at the defects of the prior art, the method for synchronizing the video and the audio of the IP network conference, which has stable and real-time data transmission, good audio and video playing effect and low system cost, is provided.
In order to solve the technical problems, the technical scheme adopted by the invention is as follows: a method for synchronizing audio and video of an IP network conference specifically comprises the following steps: establishing a video channel and an audio channel which are mutually independent between a video data sending end and an audio data receiving end, wherein the two channels respectively comprise the processing processes of data sampling, data compression, packaging, data packet sending, data packet receiving, data packet unpacking, decompression and playback; wherein,
adopting a thread in an audio channel, setting a first time parameter, coding the collected audio data in the thread by taking the first time parameter as a time interval, then packaging the coded data, and marking a sequence number by adopting an accumulated variable as an audio data packet during packaging;
while collecting audio, a thread is adopted in a video channel, a second time parameter is set, in the thread, the second time parameter is taken as a time interval, collected video data are coded, the coded data are packaged, and the accumulated variable when the video data start to be collected is taken as a sequence number when a key frame of the video data is packaged;
before the video and audio data receiving end starts to play data, a section of video and audio data is respectively buffered, and the serial numbers of the video and audio data at the initial position of the buffer area are the same; when the audio data is stored in the buffer for a certain time, the video and audio data are played.
In the process of playing the video and audio data, when the video and audio data sequence numbers cannot be kept consistent, the audio data sequence number can be used as the standard.
In the process of playing the video data and the audio data, if the video data sequence number is less than the audio data sequence number, the video data is directly discarded, and if the video data sequence number is greater than the audio data sequence number, the video data is delayed to be played until the video data sequence number is equal to the audio data sequence number and then is played.
The second time parameter may be an integer multiple of the first time parameter.
The first time parameter may range between 50 milliseconds and 150 milliseconds.
The value range of the second time parameter can be 0.8-1.2 seconds.
The variable used as the sequence number can adopt an unsigned integer type variable.
When the unsigned integer type variable is adopted as a data packet marking sequence number, 1 accumulation can be carried out on the unsigned integer type variable after each operation, and the value of the unsigned integer type variable is recovered to 0 when overflowing.
In the technical scheme, the two mutually independent channels are adopted, and the two different threads are used for respectively transmitting the audio data packet and the video data packet with the same serial number, so that the receiving end can carry out synchronous processing operation according to the serial number, thereby not only reducing the workload of the transmitting end, but also controlling the transmission delay of the network. Meanwhile, because the audio data and the video data are not encapsulated in one data packet at the same time, the first time parameter and the second time parameter can be adjusted, so that the audio data packet and the video data packet are smaller, the data packet transmission has better stability and real-time performance, and larger data volume can be transmitted to obtain better image and sound playing effect. In addition, the two threads have lower requirements on system hardware of a receiving end, the system cost can be saved, and the application range is wider. Therefore, compared with the prior art, the invention has the characteristics of strong practicability, low cost, good audio and video data transmission stability and real-time performance, good synchronization effect, better playing effect and the like.
Detailed Description
The invention will be described in further detail with reference to the drawings and specific examples.
The invention provides a video and audio synchronization method for an IP network conference, which aims to overcome the problems that a sending thread of a sending end is too busy, the data sending is influenced due to unstable network conditions, and the inapplicable common users caused by too high system cost are avoided. And, when packing and coding, the audio data packet and the corresponding video data packet are coded by the same sequence number, so that the receiving end can synchronously play the audio and video data according to the sequence number.
The embodiment specifically includes: the method comprises the steps of establishing a video channel and an audio channel which are mutually independent between a sending end and a receiving end, wherein the two channels respectively comprise the processing processes of data sampling, data compression, packaging, data packet sending, data packet receiving, data packet unpacking, decompression and playback.
The method comprises the steps that a thread is adopted in an audio channel, a first time parameter is set, collected audio data are coded in the thread by taking the first time parameter as a time interval, then the coded data are packaged, and an accumulated variable is used as an audio data package to mark a serial number during packaging;
the first time parameter has a small value, so that the data volume in a single data packet can be reduced, and the stability and the real-time performance of data packet transmission are ensured. In this embodiment, the value range of the first time parameter may be between 50 milliseconds and 150 milliseconds.
In this embodiment, the accumulated variable may adopt an unsigned integer type variable.
When the accumulated unsigned integer type variable is adopted as a data packet marking sequence number, 1 accumulation is carried out on the unsigned integer type variable after each operation, and the value of the unsigned integer type variable is recovered to 0 when overflowing, and no system exception is caused.
And simultaneously acquiring the audio, adopting a thread in the video channel, and setting a second time parameter. In the thread, the second time parameter is used as a time interval, the collected video data is coded, the coded data is packaged, and when a key frame of the video data is packaged, an unsigned integer type variable when the video data starts to be collected is used as a sequence number.
The second time parameter is an integral multiple of the first time parameter.
The value range of the second time parameter is 0.8-1.2 seconds, and the preferable value is about 1 second.
By adjusting the values of the first time parameter and the second time parameter, the receiving end can adopt a smaller data buffer area to fully buffer the received audio and video data.
Before the video and audio data receiving end starts playing data, a section of video and audio data is respectively buffered, and the serial numbers of the video and audio data at the initial position of the buffer area are the same, so that the video and audio data played immediately are synchronous.
When the audio data is stored in the buffer for a certain time, the video and audio data are played.
In the process of playing the video and audio data, when the video and audio data sequence numbers cannot be kept consistent, since discontinuous video playing can be generally received by people, the principle of ensuring continuous playing of audio as much as possible is adopted in the embodiment, and the audio data sequence numbers are taken as the standard. The method specifically comprises the following steps: if the video data sequence number is less than the audio data sequence number, the video data is directly discarded, and if the video data sequence number is greater than the audio data sequence number, the video data is delayed to be played until the video data sequence number is equal to the audio data sequence number.
Claims (8)
1. A method for synchronizing audio and video of an IP network conference specifically comprises the following steps: establishing a video channel and an audio channel which are mutually independent between a video data sending end and an audio data receiving end, wherein the two channels respectively comprise the processing processes of data sampling, data compression, packaging, data packet sending, data packet receiving, data packet unpacking, decompression and playback; wherein,
adopting a thread in an audio channel, setting a first time parameter, coding the collected audio data in the thread by taking the first time parameter as a time interval, then packaging the coded data, and marking a sequence number by adopting an accumulated variable as an audio data packet during packaging;
while collecting audio, a thread is adopted in a video channel, a second time parameter is set, in the thread, the second time parameter is taken as a time interval, collected video data are coded, the coded data are packaged, and the accumulated variable when the video data start to be collected is taken as a sequence number when a key frame of the video data is packaged;
before the video and audio data receiving end starts to play data, a section of video and audio data is respectively buffered, and the serial numbers of the video and audio data at the initial position of the buffer area are the same; when the audio data is stored in the buffer for a certain time, the video and audio data are played.
2. The method for synchronizing audio and video of an IP network conference as claimed in claim 1, wherein: in the process of playing the video and audio data, when the video and audio data sequence numbers can not be kept consistent, the audio data sequence number is taken as the standard.
3. The method for synchronizing audio and video of an IP network conference as claimed in claim 2, wherein: in the process of playing the video data and the audio data, if the video data sequence number is less than the audio data sequence number, the video data is directly discarded, and if the video data sequence number is greater than the audio data sequence number, the video data is delayed to be played until the video data sequence number is equal to the audio data sequence number and then is played.
4. The method for video and audio synchronization of IP network conference according to claim 1, 2 or 3, characterized in that: the second time parameter is an integral multiple of the first time parameter.
5. The method for video and audio synchronization in an IP network conference as claimed in claim 4, wherein: the first time parameter has a value ranging from 50 milliseconds to 150 milliseconds.
6. The method for synchronizing audio and video of an IP network conference as claimed in claim 5, wherein: the value range of the second time parameter is 0.8-1.2 seconds.
7. The method for video and audio synchronization for IP network conferencing as claimed in claim 6, wherein: the variable used as the sequence number adopts an unsigned integer type variable.
8. The method for video and audio synchronization for IP network conferencing as claimed in claim 7, wherein: when the unsigned integer type variable is adopted as a data packet marking sequence number, 1 accumulation is carried out on the unsigned integer type variable after each operation, and the value of the unsigned integer type variable is recovered to 0 when overflowing.
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| CN102137274A (en) * | 2010-07-22 | 2011-07-27 | 华为终端有限公司 | Video transmission method, video reception method, video transmission equipment, video reception equipment and video transmission system based on wireless fidelity (WiFi) |
| CN102497578A (en) * | 2011-11-25 | 2012-06-13 | 武汉大学 | Mobile audio and video real-time communication method in 3G network environment |
| CN102547482A (en) * | 2011-12-30 | 2012-07-04 | 北京锐安科技有限公司 | Synchronous playing method of multi-path IP (Internet Protocol) audio-video stream |
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| US5923655A (en) * | 1997-06-10 | 1999-07-13 | E--Net, Inc. | Interactive video communication over a packet data network |
| CN1292588C (en) * | 2001-10-29 | 2006-12-27 | 松下电器产业株式会社 | Video-voice synchronizer |
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| CN102137274A (en) * | 2010-07-22 | 2011-07-27 | 华为终端有限公司 | Video transmission method, video reception method, video transmission equipment, video reception equipment and video transmission system based on wireless fidelity (WiFi) |
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