CN101765003A - Method for transmitting audio and video under environment of network with different speeds - Google Patents
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Abstract
The invention discloses a method for transmitting audio and video under an environment of a network with different speeds applied in a multimedia communication system which at least comprises a data transmitting terminal and a data receiving terminal. The data transmitting terminal at least comprises an encoder, a data buffering and a first control logic, and the data receiving terminal controls the encoder through the first control logic according to the buffer filling degree of the data buffering and selects the source code rate of a content source. The method can be used for automatically detecting the actual data transmitting speed so that the encoding code rate of the audio and video is suitable for the changing network environment and the actual data transmitting speed, therefore, a user enjoys the optimal multimedia quality under the changing network environment.
Description
Technical field
The present invention relates to the method for transmission of audio and video, particularly the method for transmission of audio and video under the different rates network environment.
Background technology
Mobile communication equipment and terminal, for example mobile phone, PDA (personal digital assistant), notebook computer, UMPC (super mobile personal computer), PMP (portable electronic device) etc. generally are to carry out data communication in the network environment that transmission rate changes.The variation of above-mentioned transmission rate generally is presented as two aspects:
1, the coverage difference of network infrastructure (for example cellular base station).For example, in 3G (third generation) cordless communication network, 3G signal (for example WCDMA) can provide more fast data transmission rate (for example 384kbps), but this signal only covers finite regions such as down town.In other comparatively remote zones, suburb for example, might be only cover by 2.5G signal (for example GPRS) message transmission rate lower (for example 48kbps).When portable terminal moves to from the zone that the 3G signal covers when having only regional that the 2.5G signal covers, the transmission rate of the data communication of well afoot will be forced to descend.
2, the signal quality of radio communication be subjected to the space, move, the influence of factor such as interference.For example, same under the covering of 3G signal, portable terminal is good at the signal quality that certain place receives, and what receive in certain place is of poor quality; Perhaps, under the environment of portable terminal (for example on the train) in fast moving, the poor signal quality of reception.The signal quality that receives when portable terminal is during by good slip, and the transmission rate of the data communication of well afoot will be forced to descend, and the 3G signal also might be downgraded to the 2.5G signal.
In present multimedia sound, video communication, generally the content source with sound, video is encoded to different source code rate (for example 128kbps), for example, and the clear version of high code rate and the trade edition of low code check.Before content of multimedia was play, the user selected certain version to play or communicate by letter, and in communication process, between the different editions, does not allow to switch.The shortcoming of above-mentioned communication mode is:
1, the user can not accurately learn the good and bad and actual message transmission rate of the residing network environment of portable terminal usually, thus the version in chosen content source correctly.Actual message transmission rate generally can not be simply from the power of signal, or which kind of signal to be covered (3G or 2.5G) by and learn.For example, when being covered, owing to disturb, cause the poor signal quality that receives, transmission rate even might be lower than typical rate when being covered by the 2.5G signal by the 3G signal.
2, after the content of selected certain version of user began broadcast or communication, in communication process, the transmission of data can not adapt to the variation of network environment.For example, after the user selects the version of high-quality, high code check to begin communication, if network environment worsens, message transmission rate will reduce, and the quality of the content of multimedia of user's acquisition will significantly descend, broadcast phenomenon such as will occur pausing, get stuck.Otherwise, if the user is in conservative consideration, select version than low quality, low code check, when network environment can be supported high data rate, the user then failed to make full use of the high bandwidth of network and enjoys better multimedia quality.
Summary of the invention
In view of this, the object of the present invention is to provide the method for transmission of audio and video under the different rates network environment, when transmitting in the network environment that Voice ﹠ Video is changing, method of the present invention can detect actual message transmission rate automatically, make the encoder bit rate of Voice ﹠ Video adapt to the network environment and actual message transmission rate that changes, thereby make the user under the network environment that changes, enjoy best multimedia quality.
The present invention is achieved by the following technical solutions:
A kind of under the different rates network environment method of transmission of audio and video, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal and data receiver at least, described data sending terminal comprises encoder, data buffering and first control logic at least, data sending terminal is according to the buffering full level of data buffering, by the first control logic controlled encoder, the source code rate of Dynamic Selection content source.
Further, when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
Further, the described buffering full level time representation of playing in real time with buffered data.
Further, described encoder comprises audio coder and video encoder, and described data buffering comprises voice data buffering and video data buffering.
Further, described first control logic is: buffering full level f and predefined thresholding f1, and f2 ... fn compares, to the source code rate V1 of content source, and V2 ... Vn selects, and as buffering full level f 〉=fn, selects source code rate V1; As fn-1≤f＜fn, select source code rate V2; As f＜f1, select source code rate Vn, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
Further, described content source is by the encoder real-time coding.
Further, the code rate of temporal resolution, picture coding quality or the audio frequency of the spatial resolution that the source code rate of described content source can be by changing video, video is adjusted.
Another technical scheme of the present invention is as follows:
A kind of under the different rates network environment method of transmission of audio and video, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal, transfer server and data receiver at least, described data sending terminal comprises encoder, first data buffering and first control logic at least, described transfer server comprises second data buffering and second control logic at least
Data sending terminal is according to the buffering full level of first data buffering, by the first control logic controlled encoder, and the first source code rate of Dynamic Selection content source;
Transfer server is according to the buffering full level of second data buffering, by the second source code rate of the second control logic Dynamic Selection content source;
Data sending terminal is selected the source code rate of the smaller value of the described first source code rate and the second source code rate as the content source of data sending terminal.
Further, when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
Further, the described buffering full level time representation of playing in real time with buffered data.
Further, described encoder comprises audio coder and video encoder, and described first data buffering comprises voice data buffering and video data buffering, and described second data buffering comprises voice data buffering and video data buffering.
Further, described first control logic and second control logic are: buffering full level f and predefined thresholding f1, f2, ... fn compares, to the source code rate V1 of content source, V2, ... Vn selects, and as buffering full level f 〉=fn, selects source code rate V1; As fn-1≤f＜fn, select source code rate V2; As f＜f1, select source code rate Vn, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
Further, described content source is by the encoder real-time coding.
Further, the code rate of temporal resolution, picture coding quality or the audio frequency of the spatial resolution that the source code rate of described content source can be by changing video, video is adjusted.
Further, described data receiver is to drag the mode of getting from described transfer server reading of data.
A technical scheme more of the present invention is as follows:
A kind of under the different rates network environment method of transmission of audio and video, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal and data receiver at least, described data receiver comprises the version selector at least, described data receiver comprises the 3rd data buffering and the 3rd control logic at least, data receiver is according to the buffering full level of the 3rd data buffering, source code rate by the 3rd control logic chosen content source, the version selector of control data transmitting terminal, the version of Dynamic Selection content source.
Further, when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
Further, the described buffering full level time representation of playing in real time with buffered data.
Further, described the 3rd data buffering comprises voice data buffering and video data buffering.
Further, described the 3rd control logic is: buffering full level f and predefined thresholding f1, and f2 ... fn compares, to the source code rate V1 of content source, and V2 ... Vn selects, and as buffering full level f 〉=fn, selects source code rate Vn; As fn-1≤f＜fn, select source code rate Vn-1; As f＜f1, select source code rate V1, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
Further, described content source is made in advance.
Further, the different editions of described content source is prepared the critical data frame on synchronous time point, and described data sending terminal carries out the switching between the different editions on described critical data frame.
Further, described critical data frame is infra-frame prediction frame in the video coding or the independently frame in the audio coding.
Further, described data receiver is to drag the mode of getting from described data sending terminal reading of data.
Beneficial effect of the present invention is:
Good when network environment, when message transmission rate was high, the user can farthest experience best audio visual effect; When network environment worsens, when message transmission rate reduces, the user can be under lower quality the audio-visual content source, and be unlikely to be subjected to the puzzlement of playing pause, getting stuck; When network environment returned to optimum state, the user can enjoy best multimedia quality rapidly again.
The user does not need the version in chosen content source, and method of the present invention can be done adaptively selected.
It is seamless and mild that content source is switched between the version of different quality, and the user can not feel unexpected variation or pause.
Description of drawings
Fig. 1 is the data flow block diagram of data sending terminal and data receiver direct communication;
Fig. 2 data flow block diagram that to be data sending terminal communicate by letter by transfer server with data receiver;
Fig. 3 makes the data sending terminal of content source version and the data flow block diagram of data receiver direct communication in advance.
Embodiment
The network environment that changes for adapting to and the data transmission bauds of variation, content source when coding, may be encoded as multiple different source code rates version (V1, V2 ... Vn represents different source code rates, V1＜V2＜...＜Vn).For example be encoded to high, medium and low three kinds of versions.The version of above-mentioned different source code rates can be made (for example, a film in the video-on-demand service) in advance before transmission, also can be real-time (for example, the match of the live telecast in the direct broadcast service; Video telephone).
When multimedia communication began, transmitting terminal was according to the accepting state (certainly, above-mentioned state can not be represented to entirely accurate actual message transmission rate) of radiofrequency signal, the source code rate of initial option content source.For example, when radio-frequency module receives 3G WCDMA (Wideband CDMA, broadband demal multiplex (MUX) access) signal, select high-quality content source version; When receiving EDGE (Enhanced Data Rate for GSMEvolution, enhanced data rates for gsm evolution) signal, the content source version of quality in the selection; When receiving GPRS (General Packet Radio Service, GPRS) signal, select low-quality content source version.
In multimedia communication, after sound, video content source code, generally the data that coding is finished are put into the data buffering of transmitting terminal earlier, data are transferred to the network layer transmission again.For receiving terminal, the data that receive from network layer are put into the data buffering of receiving terminal earlier, play again.Multimedia communications system for using transfer server also comprises data buffering on the transfer server.Above-mentioned data buffering is first in first out (FIFO, a First In First Out) memory.
During the communication beginning, according to transmitting terminal or/and transfer server or/and the full level f of the data buffering of receiving terminal, the transmission rate of the source code rate of comparison content source and transmitting terminal, receiving terminal reality, thereby suitably the source code rate in adjustment or switch contents source is not higher than actual message transmission rate with the source code rate that guarantees content source.The time representation that the full level f of data buffering can play in real time with buffered data (for example, cushioned 3 seconds data, show that the data in this data buffering can be play for 3 seconds in real time before exhausting).
For transmitting terminal, when keeping overfill within a certain period of time, its data buffering (surpasses pre-set threshold), illustrate that the source code rate is higher than transmission rate, data can not in time be exported.Transmitting terminal then is adjusted into lower source code rate to guarantee smooth transfer of data.Otherwise, when the data buffering of transmitting terminal kept empty within a certain period of time (being lower than pre-set threshold), illustrating that the source code rate is lower than transmission code rate, data can be exported smoothly, but transmission bandwidth is underused.Transmitting terminal then is adjusted into higher source code rate to transmit higher-quality content source.
For drag the receiving terminal that fetches data from content source, when its data buffering kept empty within a certain period of time (being lower than pre-set threshold), illustrate that the source code rate is higher than transmission rate, data can not in time be imported.Transmitting terminal then is adjusted into lower source code rate to guarantee smooth transfer of data.Otherwise, when keeping overfill within a certain period of time, the data buffering of receiving terminal (surpasses pre-set threshold), illustrate that the source code rate is lower than transmission rate, and data can be imported smoothly, but transmission bandwidth is underused.Transmitting terminal then is adjusted into higher source code rate to transmit higher-quality content source.
By above-mentioned self adaptation adjustment, can realize the source code rate of content source and the dynamic equilibrium of transmission rate, thereby guarantee under the network environment that changes, to obtain best multimedia quality.
In multimedia communication, the adjustment of the source code rate of content source and switching are dynamic, promptly in playing process, need not to interrupt playing or reinitializing, content source that can the seamless switching different editions, and the user can't feel the unexpected variation or the pause of Voice ﹠ Video.
For the content source of real-time coding, encoder can be adjusted the source code rate at any time.For example, can adjust by spatial resolution, the temporal resolution of video, picture coding quality, or the code rate of audio frequency is adjusted, and then realization is to the adjustment of source code rate video.
For the content source of making in advance before the transmission, content source is encoded to the version of different quality, different code checks in advance.Different versions need to be prepared the critical data frame on synchronous time point, for example, different versions is simultaneously at the 10th second, 20 seconds, 30 seconds ... time point on prepare key frame, transmitting terminal only can carry out the switching between the different editions on above-mentioned key frame.The coding of key frame is independently, and there is not dependence in the frame data of front and back.For example, can be the infra-frame prediction frame in the video coding (I frame), the perhaps independently frame in the audio coding.
During transmitting terminal real-time coding content source,, judge the message transmission rate of the reality that network environment can provide, thereby adjust the source code rate of content source in real time according to its data buffering full level f.Buffering full level f and predefined thresholding f1, f2 ... fn (f1＜f2＜...＜fn) compare.Control logic is as follows: as f 〉=fn, the data buffering overfill is described, network environment is not good, and the message transmission rate of transmitting terminal is low, therefore selects minimum V1 source code rate; Similarly, as fn-1≤f＜fn, select only than the big V2 of V1 source code rate; By that analogy, as f＜f1, illustrate that data buffering is empty excessively, network condition is good, and the message transmission rate height of transmitting terminal should be selected maximum Vn source code rate.
See also Fig. 1, Fig. 1 is the data flow block diagram of data sending terminal and data receiver direct communication.Data sending terminal comprises video encoder, the video data buffering of audio coder, voice data buffering and the processing video data of processing audio data.Data sending terminal is according to buffering full level f, by above-mentioned control logic difference control audio encoder and video encoder.Audio coder and video encoder are selected suitable source code rate according to the buffering full level of voice data buffering and video data buffering respectively.Method of the present invention both had been applicable to the identical situation of message transmission rate of data sending terminal and data receiver, was applicable to the different situation of message transmission rate of data sending terminal and data receiver again.
See also Fig. 2, the data flow block diagram that Fig. 2 communicates by letter by transfer server for data sending terminal and data receiver.Data sending terminal exports voice data and video data earlier to the voice data buffering and the video data buffering of transfer server.Transfer server also comprises control logic, and its principle is identical with the principle of the control logic of data sending terminal.Transfer server is data cached, obtains data to drag the mode of getting from transfer server by data receiver.The transmission rate of the buffering full level reflection data sending terminal of the data buffering of data sending terminal, and the transmission rate of the buffering full level of the data buffering of transfer server reflection data receiver that is to say the ability of fetching data.When the control logic of data sending terminal is selected source code rate Vs, and the control logic of transfer server is selected the source code rate V ' during s, the selection of transfer server transfers to data sending terminal by control signal, data sending terminal is then selected source code rate min (Vs lower among both, V ' s), also be transmission rate lower in data sending terminal, the data receiver.
See also Fig. 3, Fig. 3 is for making the data sending terminal of content source version and the data flow block diagram of data receiver direct communication in advance.Data sending terminal is a server, has to make different quality, the content source version 1 of different source code rates, content source version 2, the content source version 3 that coding is finished in advance.Data receiver fetches data from data sending terminal in the mode of dragging, and exports the voice data buffering and the video data buffering of data receiver to, carries out the broadcast of sound, video again.The buffering full level of the data buffering of data receiver reflects the height of its transmission rate with respect to the source code rate.The control logic of the control logic of data receiver and Fig. 1, Fig. 2 is opposite: buffering full level f and predefined thresholding f1, and f2 ... fn (f1＜f2＜...＜fn) compare.As f 〉=fn, the data buffering overfill is described, therefore the message transmission rate height of data receiver selects maximum Vn source code rate; Similarly, as fn-1≤f＜fn, select only than the low Vn-1 of Vn source code rate; By that analogy, as f＜f1, illustrate that data buffering is empty excessively, the message transmission rate of data receiver is low, should select minimum V1 source code rate.Data receiver is according to the version selector of the source code rate control data transmitting terminal of selecting, thus the content source version of the most suitable current network transmission environment of selection.
That more than introduces only is based on preferred embodiment of the present invention, can not limit scope of the present invention with this.Any to the invention process step do well know in the art be equal to change or replace all do not exceed exposure of the present invention and protection range.
Claims (24)
1. the method for transmission of audio and video under a different rates network environment, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal and data receiver at least, described data sending terminal comprises encoder, data buffering and first control logic at least, it is characterized in that data sending terminal is according to the buffering full level of data buffering, by the first control logic controlled encoder, in the multimedia communication process, the source code rate of Dynamic Selection content source.
2. as claimed in claim 1 under the different rates network environment method of transmission of audio and video, it is characterized in that when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
3. as claimed in claim 1 under the different rates network environment method of transmission of audio and video, it is characterized in that the time representation that described buffering full level is play in real time with buffered data.
4. as claimed in claim 1 under the different rates network environment method of transmission of audio and video, it is characterized in that described encoder comprises audio coder and video encoder, described data buffering comprises that voice data buffering and video data cushion.
5. as claimed in claim 1 under the different rates network environment method of transmission of audio and video, it is characterized in that, described first control logic is: buffering full level f and predefined thresholding f1, f2 ... fn compares, to the source code rate V1 of content source, V2, ... Vn selects, and as buffering full level f 〉=fn, selects source code rate V1; As fn-1≤f＜fn, select source code rate V2; As f＜f1, select source code rate Vn, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
6. as claimed in claim 1 under the different rates network environment method of transmission of audio and video, it is characterized in that described content source is by the encoder real-time coding.
7. as claimed in claim 6 under the different rates network environment method of transmission of audio and video, it is characterized in that the code rate of the spatial resolution that the source code rate of described content source can be by changing video, temporal resolution, picture coding quality or the audio frequency of video is adjusted.
8. the method for transmission of audio and video under a different rates network environment, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal, transfer server and data receiver at least, described data sending terminal comprises encoder, first data buffering and first control logic at least, described transfer server comprises second data buffering and second control logic at least, it is characterized in that
Data sending terminal is according to the buffering full level of first data buffering, by the first control logic controlled encoder, and in the multimedia communication process, the first source code rate of Dynamic Selection content source;
Transfer server is according to the buffering full level of second data buffering, by second control logic, and in the multimedia communication process, the second source code rate of Dynamic Selection content source;
Data sending terminal is selected the source code rate of the smaller value of the described first source code rate and the second source code rate as the content source of data sending terminal.
9. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
10. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that the time representation that described buffering full level is play in real time with buffered data.
11. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that, described encoder comprises audio coder and video encoder, described first data buffering comprises voice data buffering and video data buffering, and described second data buffering comprises voice data buffering and video data buffering.
12. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that, described first control logic and second control logic are: buffering full level f and predefined thresholding f1, f2 ... fn compares, to the source code rate V1 of content source, V2, ... Vn selects, and as buffering full level f 〉=fn, selects source code rate V1; As fn-1≤f＜fn, select source code rate V2; As f＜f1, select source code rate Vn, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
13. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that described content source is by the encoder real-time coding.
14. as claimed in claim 13 under the different rates network environment method of transmission of audio and video, it is characterized in that the code rate of the spatial resolution that the source code rate of described content source can be by changing video, temporal resolution, picture coding quality or the audio frequency of video is adjusted.
15. as claimed in claim 8 under the different rates network environment method of transmission of audio and video, it is characterized in that described data receiver is to drag the mode of getting from described transfer server reading of data.
16. the method for transmission of audio and video under a different rates network environment, be applied in the multimedia communications system, described multimedia communications system comprises data sending terminal and data receiver at least, described data sending terminal comprises the version selector at least, and prepare the version of the content source of a plurality of different qualities and source code rate in advance, described data receiver comprises the 3rd data buffering and the 3rd control logic at least, it is characterized in that, data receiver is according to the buffering full level of the 3rd data buffering, source code rate by the 3rd control logic chosen content source, the version selector of control data transmitting terminal, in the multimedia communication process, the version of Dynamic Selection content source.
17. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that when multimedia communication began, transmitting terminal was according to the accepting state of radiofrequency signal, the source code rate of initial option content source.
18. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that the time representation that described buffering full level is play in real time with buffered data.
19. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that described the 3rd data buffering comprises that voice data buffering and video data cushion.
20. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that, described the 3rd control logic is: buffering full level f and predefined thresholding f1, f2 ... fn compares, to the source code rate V1 of content source, V2, ... Vn selects, and as buffering full level f 〉=fn, selects source code rate Vn; As fn-1≤f＜fn, select source code rate Vn-1; As f＜f1, select source code rate V1, wherein, and f1＜f2＜...＜fn, V1＜V2＜...＜Vn, n are natural numbers.
21. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that described content source is made in advance.
22. as claimed in claim 21 under the different rates network environment method of transmission of audio and video, it is characterized in that, the different editions of described content source is prepared the critical data frame on synchronous time point, described data sending terminal carries out the switching between the different editions on described critical data frame.
23. as claimed in claim 22 under the different rates network environment method of transmission of audio and video, it is characterized in that described critical data frame is infra-frame prediction frame in the video coding or the independently frame in the audio coding.
24. as claimed in claim 16 under the different rates network environment method of transmission of audio and video, it is characterized in that described data receiver is to drag the mode of getting from described data sending terminal reading of data.
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2008
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