CN1328891C - A semantic integrity ensuring method under IP network environment - Google Patents
A semantic integrity ensuring method under IP network environment Download PDFInfo
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- CN1328891C CN1328891C CNB2004100885942A CN200410088594A CN1328891C CN 1328891 C CN1328891 C CN 1328891C CN B2004100885942 A CNB2004100885942 A CN B2004100885942A CN 200410088594 A CN200410088594 A CN 200410088594A CN 1328891 C CN1328891 C CN 1328891C
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- 238000005538 encapsulation Methods 0.000 claims description 4
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Abstract
The present invention discloses a semantic integrity ensuring method under IP network environment. The method comprises that a speech data packet is divided and sealed by time length at the speech data transmitting end; after the speech data packet which is buffered at the speech data receiving end achieves a time delay number, the speech data packet buffers and plays speech data strictly according to time delay number; a delayed or lost speech data packet is replaced by noise data; when the quantity of continuous delayed or lost speech data packets is accumulated over a first critical numerical value, a speech playing thread of the speech data receiving end stops; after the quantity of the data packets waiting for being buffered is accumulated to the time delay number, the speech playing thread is started. In the process of speech playing, when the buffering position of the speech data packet is beyond the length of the playing position to exceed a second critical numerical value, a pointer in the speech playing thread is immediately turned to the specified position, so that the buffering number of the speech data packet is equal to the time delay number; the noise data is generated in the meantime so as to prompt that user data is damaged.
Description
Technical field
The present invention relates to the semantic integrity ensuring technology under a kind of IP network environment, promptly under the insecure host-host protocol of IP network, still can effectively guarantee the method for semantic integrity.
Background technology
Along with the develop rapidly of computer network, also more and more widely based on the application of the voice-transmission system of computer networking technology.Because the real-time of voice transfer makes its transmission rate to computer network, network condition that very high requirement be arranged.
Yet rare basically computer network can reach so high requirement at present.On the other hand, the computer network communication channel is a multiple users share, and in spare time when busy when Channel Sharing just makes the computer network channel, just bad during the channel conditions fashion, this has just further influenced performance, the effect of voice-transmission system.
Under the relatively poor situation of network condition, if do not take certain measure to be controlled, may there be many problems in the real-time voice transmission system.Show:
(1) network delay is long, and system real time is poor.Under the comparatively busy situation of computer network, to receiving speech data each node transmission course that sends, all time of length may be waited for because this node is comparatively busy from dispatching station.Perhaps because certain section network is relatively poor, and have to select other the long route of physical circuit, this just may cause packet to arrive bigger delay, and this can cause the problem that system real time is relatively poor, and may be transmitting terminal at the voice of receiving terminal uppick sends before for a long time.
(2) the receiving terminal speech play is not smooth.Under the little stable situation of computer network performance, in a certain period, when network condition was better, packet arrives the time delay of receiving terminal may be very little; In another period, when network condition was relatively poor, packet arrives the time delay of receiving terminal may be relatively large, if do not handled at receiving terminal, then when playing, just can produce off and on voice, causes speech play not smooth.
(3) during loss of data, the semantic integrity of the voice that transmit is relatively poor.In most of computer network, if the network system situation is relatively poor, when network blockage occurring, some node wherein may discard some packet selectively, if VoP is dropped, because the continuity and the real-time of voice transfer, may not can resend the packet that these are dropped at transmitting terminal, the packet that makes receiving terminal receive is not complete, continuous, thereby may make the speech data of receiving lose semantic integrality, cause semantic error.
By way of example: " enter ... there are companies such as Microsoft, IBM, Dell, SUN in the enterprise of list " if will transmit one section such voice.If these two sounds of a data packet transmission " Dell " are arranged, in network, be dropped, the voice of receiving at receiving terminal are companies such as " entering ... there are Microsoft, IBM, SUN in the enterprise of list " so.So just, lost semantic integrality.The semantic integrity of voice is very important, if the voice that receiving terminal is received are incomplete, ambiguity is arranged, just probably give rise to misunderstanding.
Because voice transfer is too strong to the dependence of network performance, a good voice-transmission system must be considered at following aforementioned variety of issue that may cause of the relatively poor situation of network.In order to overcome these problems, just must the processing playout software of receiving terminal be improved,, can be provided real-time, smooth voice to greatest extent so that when network problem occurring, and lose at speech data, when semantic integrity goes wrong, can remind the user.
In order to overcome above-mentioned problem, can carry out some processing to the data that receive at receiving terminal, so that better service to be provided.The main method that can Gong take at present is mainly
(1) data buffering method.This is the method that early stage transmission system is used commonplacely.If under the relatively poor situation of network, the voice that the user hears not are continuous, but off and on, in this case, the user will feel to be difficult to be subjected to.Based on user's this needs, the data buffering method mainly is devoted to ensure the continuous, smooth of voice.
The main thought of data buffering method is: at receiving terminal, handling procedure is not to unpack immediately, decode, play with the data mistake that receives, but be put into earlier in the system buffer, when the number of packet in the buffering area reaches certain value M, just can be once to all data decodes, broadcast.When system carried out data buffering, system was quiet to user's broadcast, and is shown to the prompting that the user " cushions ".
Wherein, the value of M can be decided in its sole discretion according to network condition by system.When network condition is better, just M is made as a bigger value, so once play long voice, so that voice sound smooth more.When network condition is relatively poor, be unlikely to for a long time sky etc. in order to make the user, M can be made as a smaller value, to guarantee the real-time of system.
Data buffering method has promptly ensured the fluency of receiving terminal voice from user's angle, has taken into account the real-time of receiving terminal voice again.Yet also there are a lot of shortcomings in data buffering method.At first, the data buffering method has consumed system's memory source greatly because the size of buffering area is indefinite, and this often needs a bigger buffering area to carry out work.Secondly, in the data buffering method, being provided with of M value is very crucial, will directly have influence on the quality of system to the whether reasonable of M value setting, how to hold, control the value of M, is a difficult problem.In addition, because the buffering of packet may make that the real-time of system is not very strong.Particularly importantly, the data buffering method is not considered " semantic integrity " this problem, and incomplete voice make the user when listening to voice, probably can produce wrong understanding.(2) real-time playing system.In some voice-transmission system, very strict to the real-time requirement of voice, the data that transmitting terminal sent must strictly receive in the time that system allowed and broadcast.On the other hand, development of computer network makes that the speed of transmission data is more and more higher in the network.100,000,000, the network of gigabit or even TB level occurs in succession and popularizes, and this makes network condition that very big improvement arranged.Like this, in many cases, broadcast in real time just is enough to satisfy user's needs.
The real-time playing system main method is: A. divides the data that will send at transmitting terminal with isometric time T.The requirement of T viewing system and deciding can be very short, also can be longer.B. the data number that will send, packing and transmission.C. at receiving terminal, received data not cushioned, perhaps carry out the buffering of very short time (Millisecond) after, play.If the late situation of individual packet, then with the quiet expression of respective length.D. when receiving late packet,, can directly it be abandoned because the context dependent packet of this packet is play.
Real-time playing system has fully ensured the real-time of voice-transmission system.Under most situation, real-time playing system has been obtained good effect.Yet, if the relatively poor situation of network, the voice that real-time playing system plays out at receiving terminal conveniently off and on, this runs counter to " data buffering method ".
In addition, because real-time playing system only is simply with late data packet discarding, and replace quietly, and occur when quiet, it is speaker's pause that the user can think it by mistake very much, and can not expect having occurred loss of data, may receive garbled voice.Obviously can not ensure semantic integrality.
Summary of the invention
The objective of the invention is: at the deficiencies in the prior art, provide a kind of under the insecure host-host protocol of IP network, still can effectively guarantee the method for semantic integrity.
In order to solve the problems of the technologies described above, the technical solution used in the present invention is: a kind of method of the semantic integrity ensuring under the IP network environment comprises the steps:
Step 1, preestablish the value of time span at the speech data transmitting terminal, and with described time span divide, the encapsulation VoP;
Step 2, preestablish the value of time-delay number, after the VoP of speech data rx-side buffering reaches described time-delay number, according to this time-delay number buffering, play speech data at the speech data receiving terminal;
Step 3, replace with noise data for VoP late or that lose;
Step 4, preestablish first critical numerical value at the speech data receiving terminal, when continuous VoP number accumulation late or that lose surpasses described first critical numerical value, then stop the speech play thread of speech data receiving terminal, the number of data packets of pending buffer is restarted the speech play thread after running up to described time-delay number;
Step 5, preestablish second critical numerical value at the speech data receiving terminal, in the process of speech play, when the length of leading over play position when the buffer position of VoP surpasses described second critical numerical value, then immediately the pointer skips in the speech play thread is arrived assigned address, make VoP buffering number equal described time-delay number, and produce one section noise data during this period.
The described span that is used for the time span of the division of speech data transmitting terminal, encapsulation VoP can be between 50 milliseconds to 150 milliseconds.
The value of the predefined time-delay number of described speech data receiving terminal can be for making the voice time-delay between 0.5 second to 1 second.
The span of described first critical numerical value can be for making the accumulated time of voice between 0.5 second to 2 seconds.
Described second critical numerical value can be greater than described time-delay number.
In technique scheme, the present invention is owing to divide the data that will send at the voice transmitting terminal with the isometric time, no matter and to make the speech data receiving terminal be under the situation of normal play, still occurring under the late or situation of losing of packet, perhaps the buffer position at VoP substantially exceeds under the situation of speech play position, all in strict accordance with the time-delay number buffer memory determined with play the VoP that is received, thus real-time that can the strict guarantee speech data and property at the uniform velocity.Simultaneously, in the real-time that guarantees language data with at the uniform velocity under the prerequisite of property, when guaranteeing that further semantic integrality, the present invention adopt the packet of the voice receiving terminal being found to lose or being late, produce this section of method prompting user voice generation problem of noise, thereby avoid the generation of ambiguity.Therefore relative prior art, the present invention not only has real-time, the smooth characteristics of speech play, also has the characteristics that can effectively guarantee complete semantic.
Embodiment
Below in conjunction with specific embodiment the present invention is described in further detail.
A kind of semantic integrity ensuring method under the IP network environment provided by the present invention comprises the steps:
Step 1, preestablish the value of time span L at the speech data transmitting terminal, this time span L span is between 50 milliseconds to 150 milliseconds.Divide, encapsulate VoP at the speech data transmitting terminal with this time span L.
Step 2, preestablish the value of time-delay number T at the speech data receiving terminal.For the fluency that guarantees that voice are linked up, the span of the VoP number T of speech data rx-side buffering is for making the voice time-delay between 0.5 second to 1 second.After the VoP of speech data rx-side buffering reaches time-delay number T, in strict accordance with this time-delay number T buffering, broadcast speech data.
Step 3, then replace with noise data for VoP late or that lose.
Step 4, preestablish the first critical numerical value M at the speech data receiving terminal.When continuous VoP number accumulation late or that lose surpasses the described first critical numerical value M, then stop the speech play thread of speech data receiving terminal, the number of data packets of pending buffer runs up to after the described time-delay number T, restarts the speech play thread.
In order to guarantee the sensitivity of system responses, the span that VoP number late continuously or that lose is accumulated the first critical numerical value M is to make the accumulated time of voice between 0.5 second to 2 seconds.
Step 5, preestablish the second critical numerical value N at the speech data receiving terminal, the described second critical numerical value N is greater than described time-delay number T.In process in speech play, when the length that the buffer position of VoP is led over play position surpasses the described second critical numerical value N, then immediately the pointer skips in the speech play thread is arrived assigned address, make VoP buffering number equal described time-delay number T, and produce one section noise data during this period, destroyed with the prompting user data.
Can reach by this method under the insecure host-host protocol of IP network, still can effectively guarantee the purpose of semantic integrity.
It below only is a kind of specific embodiment of the present invention.In the practical application, also can design according to the present invention make corresponding change, reach same function and effect, as the way that replaces with noise data for VoP late or that lose, also can adopting tolls waits other voice prompt data to replace.Therefore, all changes of doing according to technical solution of the present invention when the function that is produced does not exceed the scope of technical solution of the present invention, all belong to protection scope of the present invention.
Claims (4)
1, a kind of method of the semantic integrity ensuring under the IP network environment comprises the steps:
Step 1, preestablish the value of time span at the speech data transmitting terminal, and with described time span divide, the encapsulation VoP;
Step 2, preestablish the value of time-delay number, after the VoP of speech data rx-side buffering reaches described time-delay number, according to this time-delay number buffering, play speech data at the speech data receiving terminal;
Step 3, replace with noise data for VoP late or that lose;
Step 4, preestablish first critical numerical value at the speech data receiving terminal, when continuous VoP number accumulation late or that lose surpasses described first critical numerical value, then stop the speech play thread of speech data receiving terminal, the number of data packets of pending buffer is restarted the speech play thread after running up to described time-delay number;
Step 5, preestablish second critical numerical value at the speech data receiving terminal, described second critical numerical value is greater than described time-delay number, in the process of speech play, when the length of leading over play position when the buffer position of VoP surpasses described second critical numerical value, then immediately the pointer skips in the speech play thread is arrived assigned address, make VoP buffering number equal described time-delay number, and produce one section noise data during this period.
2, the method for the semantic integrity ensuring under the IP network environment according to claim 1 is characterized in that: describedly be used for that the speech data transmitting terminal is divided, the span of the time span of encapsulation VoP is between 50 milliseconds to 150 milliseconds.
3, the method for the semantic integrity ensuring under the IP network environment as claimed in claim 1 or 2 is characterized in that: the value of the predefined time-delay number of described speech data receiving terminal is for making the voice time-delay between 0.5 second to 1 second.
4, as the method for the semantic integrity ensuring under the IP network environment as described in the claim 3, it is characterized in that: the span of described first critical numerical value is to make the accumulated time of voice between 0.5 second to 2 seconds.
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| CNB2004100885942A CN1328891C (en) | 2004-11-09 | 2004-11-09 | A semantic integrity ensuring method under IP network environment |
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| CNB2004100885942A CN1328891C (en) | 2004-11-09 | 2004-11-09 | A semantic integrity ensuring method under IP network environment |
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| CN1328891C true CN1328891C (en) | 2007-07-25 |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN102238076A (en) * | 2010-04-29 | 2011-11-09 | 财团法人工业技术研究院 | Method for performing transmission scheduling by taking block as unit and computer program product |
| CN105845138A (en) * | 2016-03-25 | 2016-08-10 | 乐视控股（北京）有限公司 | Voice signal processing method and apparatus |
Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1346198A (en) * | 2000-09-30 | 2002-04-24 | 华为技术有限公司 | Anti-loss treating method for IP speech sound data package |
| CN1364289A (en) * | 2000-03-06 | 2002-08-14 | 扎林克半导体公司 | Sub-packet insertion for packet loss compensation in voice over IP net works |
| JP2003273914A (en) * | 2002-03-13 | 2003-09-26 | Oki Electric Ind Co Ltd | Voice packet communication equipment, traffic prediction method and optimal control method of call quality in voice packet communication equipment |
| US20040120309A1 (en) * | 2001-04-24 | 2004-06-24 | Antti Kurittu | Methods for changing the size of a jitter buffer and for time alignment, communications system, receiving end, and transcoder |
| JP2004229115A (en) * | 2003-01-24 | 2004-08-12 | Japan Radio Co Ltd | Control method for receiving buffer in voice packet data receiving device, and voice packet data receiving device |
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Patent Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1364289A (en) * | 2000-03-06 | 2002-08-14 | 扎林克半导体公司 | Sub-packet insertion for packet loss compensation in voice over IP net works |
| CN1346198A (en) * | 2000-09-30 | 2002-04-24 | 华为技术有限公司 | Anti-loss treating method for IP speech sound data package |
| US20040120309A1 (en) * | 2001-04-24 | 2004-06-24 | Antti Kurittu | Methods for changing the size of a jitter buffer and for time alignment, communications system, receiving end, and transcoder |
| JP2003273914A (en) * | 2002-03-13 | 2003-09-26 | Oki Electric Ind Co Ltd | Voice packet communication equipment, traffic prediction method and optimal control method of call quality in voice packet communication equipment |
| JP2004229115A (en) * | 2003-01-24 | 2004-08-12 | Japan Radio Co Ltd | Control method for receiving buffer in voice packet data receiving device, and voice packet data receiving device |
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