CN103209442A - Method and terminal for dynamically setting voice service transmission parameters - Google Patents
Method and terminal for dynamically setting voice service transmission parameters Download PDFInfo
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Abstract
The invention discloses a method and a terminal for dynamically setting voice service transmission parameters. The method comprises the steps of obtaining encoding and decoding types of voice service transmission; and setting transmission parameters required by transmission of voice services according to encoding and decoding types. By the aid of the method and the terminal, corresponding transmission parameters can be set for voice services according to different encoding and decoding types, so that waste of limited air interface resources caused by the fact that the same bandwidth is allocated to different encoding and decoding types in the prior art is prevented, and limited air interface resources can be saved.
Description
Technical field
The present invention relates to the communications field, particularly relate to a kind of method and terminal that dynamically arranges speech business transmission parameter.
Background technology
(increasing user brings into use wireless network to carry out the business of in the past carrying out usually in cable network for Worldwide Interoperability for Microwave Access, WiMAX) the scale commercialization of network along with worldwide interoperability for microwave inserts.For example, carry out Video chat by wireless network, download mass data etc.The bandwidth resources that these business take will be far longer than the shared bandwidth resources of common voice calling service.Especially in wireless networks such as WiMAX, because the interface-free resources (being the bandwidth resources in the wireless network) that permission is used is limited, so more should rationally utilize bandwidth resources.
In the prior art, the terminal equipment of WiMAX network mostly has IP phone (Voice over Internet Portocol, VOIP) function.VOIP is exactly with simulated sound signal (Voice) digitlization in brief, does real-time transmission with the form of data packet (Data Packet) at IP data network (IPNetwork).In the VOIP technology, have multiple for the code encoding/decoding mode of speech data.Than higher, the occupied bandwidth resource is less for the compression degree of speech data for the code encoding/decoding mode that has, and speech quality is poor slightly; The code encoding/decoding mode that has the then compression degree for speech data is lower, and the occupied bandwidth resource is more, but speech quality is better.
The method that speech business transmission parameter is set of the prior art, when terminal was initiated dynamic service flow, network side was that the parameters such as bandwidth resources of terminal distribution are fixed.That is, no matter which kind of code encoding/decoding mode is reality adopt, and all can be the terminal distribution bandwidth based on the needed bandwidth resources of a certain code encoding/decoding mode (the normally more mode of occupied bandwidth resource).Therefore, the bandwidth allocation methods of wireless network speech business of the prior art has caused waste to limited empty resource.
Summary of the invention
The purpose of this invention is to provide a kind of method and terminal that dynamically arranges speech business transmission parameter, can be according to the code encoding/decoding mode of each speech business, determine transmission parameter required under this code encoding/decoding mode, and then distribute corresponding bandwidth for each wireless network speech business, can save limited empty resource.
For achieving the above object, the embodiment of the invention provides following scheme:
The embodiment of the invention provides a kind of method that dynamically arranges speech business transmission parameter, comprising: the code/decode type that obtains transmitting voice service; According to described code/decode type the required transmission parameter of the described speech business of transmission is set.
The embodiment of the invention also provides a kind of terminal, comprising: the code/decode type acquiring unit, for the code/decode type that obtains transmitting voice service; The transmission parameter settings unit is used for according to described code/decode type the required transmission parameter of the described speech business of transmission being set.
According to specific embodiment provided by the invention, the embodiment of the invention realizes following technique effect:
The method that dynamically arranges speech business transmission parameter disclosed in this invention has been considered different these factors of transmission parameter corresponding under the different code/decode types, by obtaining the code/decode type of transmitting voice service; According to described code/decode type the required transmission parameter of the described speech business of transmission is set; Can be at different code/decode types, for this speech business arranges corresponding transmission parameter, thereby avoided the waste to limited empty resource that distributes identical bandwidth to cause for different code/decode types in the prior art, can save limited empty resource.
Description of drawings
In order to be illustrated more clearly in the embodiment of the invention or technical scheme of the prior art, to do to introduce simply to the accompanying drawing of required use among the embodiment below, apparently, accompanying drawing in describing below only is some embodiments of the present invention, for those of ordinary skills, under the prerequisite of not paying creative work, can also obtain other accompanying drawing according to these accompanying drawings.
The method flow diagram that dynamically arranges speech business transmission parameter that Fig. 1 provides for the embodiment of the invention one;
The method flow diagram that dynamically arranges speech business transmission parameter that Fig. 2 provides for the embodiment of the invention two;
The structure chart of the terminal that Fig. 3 provides for the embodiment of the invention three;
The structure chart of the terminal that Fig. 4 provides for the embodiment of the invention four.
Embodiment
Below in conjunction with the accompanying drawing in the embodiment of the invention, the technical scheme in the embodiment of the invention is clearly and completely described, obviously, described embodiment only is the present invention's part embodiment, rather than whole embodiment.Based on the embodiment among the present invention, those of ordinary skills belong to the scope of protection of the invention not making the every other embodiment that obtains under the creative work prerequisite.
For above-mentioned purpose of the present invention, feature and advantage can be become apparent more, the present invention is further detailed explanation below in conjunction with the drawings and specific embodiments.
Embodiment one
The method flow diagram that dynamically arranges speech business transmission parameter that Fig. 1 provides for the embodiment of the invention one.The executive agent of present embodiment can be a kind of wireless Internet access terminal equipment.As shown in Figure 1, the method comprising the steps of:
S101: the code/decode type that obtains transmitting voice service;
In the practical application, initiate professional terminal can with the server negotiate code encoding/decoding mode of network side.Behind terminal and the intact code/decode type of server negotiate, negotiation result is sent to server, optionally, terminal can pass to negotiation result the server of WiMAX side by THP message (Tomlinson-Harashima Precoding, Tomlinson-Harrar is wished the agate precoding).The server of WiMAX side can be dispatched this speech business according to negotiation result.
Concrete, when consulting code/decode type, the code encoding/decoding mode that can support according to described terminal or the business of current initiation are selected suitable code encoding/decoding mode to the requirement of signal quality.For example, when terminal and server are all supported the silence compression function, the code encoding/decoding mode that can select to have the silence compression function; When this business is higher to demand on signal quality, can select more but the code encoding/decoding mode that signal quality is high of occupied bandwidth.
S102: the required transmission parameter of the described speech business of transmission is set according to described code/decode type.
According to described code/decode type the required transmission parameter of the described speech business of transmission is set, comprises: transmission parameters such as the required peak transfer rate of the described speech business of transmission, minimum transmission rate, propagation delay time, shake are set according to described code/decode type.
Wherein, transmission such as propagation delay time, shake parameter is that network quality levels by the required bandwidth correspondence of this speech business determines.Optionally, parameters such as the propagation delay time of described network quality levels correspondence, shake can be with reference to following table, and following table is from communication industry standard YD/T 1071-2000 " requirement of IP phone gateway equipment and technology ".
| Network quality levels | One Way Delay (ms) | Loss rate | Shake (ms) |
| Well (self-defined) | ≤40 | ≈0 | ≤10 |
| Relatively poor | ≤100 | ≤1％ | ≤20 |
| Badly | ≤400 | ≤5％ | ≤60 |
In the WiMAX system, G.711 code encoding/decoding mode can adopt, G.729, G.723 wait encoding and decoding technique.Concrete, (Pulse-code modulation PCM), is the cover voice compression that International Telecommunications Union stipulates out, is mainly used in phone G.711 to be also referred to as pulse code modulation.It mainly with pulse code modulation to audio sample, sample rate is the 8k per second.It utilizes not pressure channel transferring voice signal of a 64Kbps.Playing compression ratio is 1: 2, namely 16 bit data is compressed into 8.G.723.1 be that International Telecommunications Union's telecommunication standardsization is organized in a kind of multi-media voice encoding and decoding standard of working out moulding in 1996.Its typical case uses and comprises VoIP service, visual telephone, radio telephone, digital satellite system, number electric multiplication equipment, public switch telephone network and various multi-media voice information products.G.723.1 standard transmission code check has two kinds of 5.3kb/s and 6.3kb/s.G.729 encoding scheme is the standard of the speech signal coding of telephone bandwidth, and to analog signal 8kHz, the sampling of input voice character, 16 bit linear PCM quantize.
As seen, under the different code encoding/decoding modes, be different for the compression degree of speech data, therefore, the actual required bandwidth of different code encoding/decoding modes is also inequality.So, according to described code/decode type the required transmission parameter of the described speech business of transmission is set and comprises: calculate the required bandwidth of the described speech business of transmission according to described code/decode type.
Every kind of code encoding/decoding mode has all been stipulated corresponding encoding and decoding standard.Wherein, comprise packing cycle and per second packing length.Because speech data normally transmits with the form of packet, so the packing cycle is just represented the encapsulation required time of each packet, unit is generally Millisecond.Per second packing length is exactly the payload data amount that all packets of each encapsulation in second comprise.
The packaged data owner of each packet is wanted to be divided into two parts.A part is fixed field, and another part is payload.Described fixed field mainly refers to the protocol fields that each packet carries, for example: real time transport protocol field, User Datagram Protocol field, procotol field and Ethernet field etc.Because the length of these fields is all fixed in every kind of code encoding/decoding mode, so the shared field length of these fields also is known, is called fixed field.Payload refers to represent that part of data of voice messaging.In different code encoding/decoding modes, the payload in each packet is also inequality.Payload=packing cycle (be unit with the second) * per second packing length.
Therefore, can calculate the required bandwidth value of described wireless network speech business according to the fixed field length of described per second packing length and packing cycle and each packet; The field length of the protocol fields that described fixed field length is carried for each packet;
Concrete, G.711, G.729, and G.723.1 (5.3kbit/s), G.723.1 in the coded systems such as (6.3Kbit/s), the length of fixed field can be calculated.Usually, fixed field mainly comprises: real time transport protocol field, User Datagram Protocol field, procotol field and Ethernet field.Wherein, real time transport protocol field (Real-time Transport Protocol, RTP) length is 96bit, User Datagram Protocol field (User Datagram Protocol, UDP) length is 64bit, the procotol field (Internet Protocol, length IP) is 160bit, the length of Ethernet (Ethernet) field is 208bit.The length summation of above-mentioned four fields is 528bit.So the length of fixed field can be represented with 528bit.
How to describe in detail below according to the fixed field length of described per second packing length and packing cycle and each packet, calculate the required bandwidth value of described wireless network speech business.
Required bandwidth=individual data packet length * per second packing number.Individual data packet length=fixed field length+payload wherein, per second packing number=1/ packing cycle.Payload=packing cycle (be unit with the second) * per second packing length.Therefore, after the fixed field length of known described per second packing length and packing cycle and each packet, can multiply by the described packing cycle with described per second packing length, obtain payload, described payload is the data of expression voice messaging in the packet; With described payload and the addition of described fixed field length, obtain the individual data packet length; Obtain per second packing number according to the conversion of described packing cycle; Multiply by described individual data packet length with described per second packing number, obtain described bandwidth value.
As seen, the method that dynamically arranges speech business transmission parameter disclosed in this invention has been considered different these factors of transmission parameter corresponding under the different code/decode types, by obtaining the code/decode type of transmitting voice service; According to described code/decode type the required transmission parameter of the described speech business of transmission is set; Can be at different code/decode types, for this speech business arranges corresponding transmission parameter, thereby avoided the waste to limited empty resource that distributes identical bandwidth to cause for different code/decode types in the prior art, can save limited empty resource.
Embodiment two
The method flow diagram that dynamically arranges speech business transmission parameter that Fig. 2 provides for the embodiment of the invention two.As shown in Figure 2, the method comprising the steps of:
S201: the code/decode type that obtains transmitting voice service;
S202: calculate the required bandwidth of the described speech business of transmission according to described code/decode type;
S203: according to the network quality levels of described required bandwidth correspondence, the required transmission parameter of the described speech business of transmission is set;
S204: obtain described speech business and whether support the silence compression function;
In the voice call process, the pause (for example because the silence that thinking etc. cause) of blink may appear.Like this, when of short duration pause appears in a side of conversation, because the characteristic of speech business, under the situation without any voice messaging, still need be to side's transmission information of answering conversation, this information is used for expression speaker's this moment and does not send any voice.Silence compression just refers to, when the caller does not send voice, represents that with a sign caller is in silent status, only needs to send this sign, and no longer sends the background sound of caller's environment of living in.Because the data volume size of sign will be much smaller than the data volume size of background sound information, so the silence compression function can be saved bandwidth resources.
S205: when the silence compression function is supported in described speech business, described scheduling type is set is the real-time polling service ERTPS of expansion;
(Extended Rt-Polling ERTPS) is a kind of dispatching method of supporting the silence compression function of the prior art to the real-time polling service of expansion.
S206: when the silence compression function was not supported in described speech business, it was Unsolicited Grant Service UGS that described scheduling type is set.
Initiatively (Unsolicited Grant Service UGS) is a kind of dispatching method of not supporting the silence compression function of the prior art to the grant bandwidth business.
Present embodiment, also can be at different code/decode types, for this speech business arranges corresponding transmission parameter, thereby avoid the waste to limited empty resource that distributes identical bandwidth to cause for different code/decode types in the prior art, can save limited empty resource.
Present embodiment is compared with a last embodiment, relate to the dispatching method whether terminal of described wireless network speech business supports to have the silence compression function by judgement, when terminal is supported silence compression, adopt the dispatching method of the real-time polling service of expansion, when terminal is not supported silence compression, adopt the dispatching method of active grant bandwidth business that described wireless network speech business is dispatched, can also further save the interface-free resources of wireless network.
Embodiment three
The structure chart of the terminal that Fig. 3 provides for the embodiment of the invention three.As shown in Figure 3, this terminal comprises:
Code/decode type acquiring unit 301 is for the code/decode type that obtains transmitting voice service;
Transmission parameter settings unit 302 is used for according to described code/decode type the required transmission parameter of the described speech business of transmission being set.
As seen, terminal disclosed in this invention has been considered different these factors of transmission parameter corresponding under the different code/decode types, by obtaining the code/decode type of transmitting voice service; According to described code/decode type the required transmission parameter of the described speech business of transmission is set; Can be at different code/decode types, for this speech business arranges corresponding transmission parameter, thereby avoided the waste to limited empty resource that distributes identical bandwidth to cause for different code/decode types in the prior art, can save limited empty resource.
Embodiment four
The structure chart of the terminal that Fig. 4 provides for the embodiment of the invention four.As shown in Figure 4, this terminal comprises:
Code/decode type acquiring unit 401 is for the code/decode type that obtains transmitting voice service;
Transmission parameter settings unit 402, described transmission parameter comprises: one or more in the network parameters such as peak transfer rate, minimum transmission rate, propagation delay time, shake, scheduling type.
Described transmission parameter settings unit 402 can comprise:
Transmission rate arranges subelement 4021, is used for determining required peak transfer rate and the minimum transmission rate of the described speech business of transmission according to described code/decode type.
Other transmission parameter settings subelements 4023 are used for the network quality levels according to described required bandwidth correspondence, and required propagation delay time, the shake of the described speech business of transmission is set.
Silence compression function support information acquiring unit 403 is used for obtaining described speech business and whether supports the silence compression function;
Whether scheduling type arranges unit 404, be used for supporting the silence compression function setting to transmit the required scheduling type of described speech business according to described speech business;
Described scheduling type arranges unit 404 and can comprise:
The real-time polling service scheduling type arranges subelement 4041, is used for when the silence compression function is supported in described speech business, described scheduling type is set is the real-time polling service ERTPS of expansion;
The Unsolicited Grant Service scheduling type arranges subelement 4042, is used for when the silence compression function is not supported in described speech business, and it is Unsolicited Grant Service UGS that described scheduling type is set.
Present embodiment is compared with a last embodiment, relate to the dispatching method whether terminal of described wireless network speech business supports to have the silence compression function by judgement, when terminal is supported silence compression, adopt the dispatching method of the real-time polling service of expansion, when terminal is not supported silence compression, adopt the dispatching method of active grant bandwidth business that described wireless network speech business is dispatched, can also further save the interface-free resources of wireless network.
Each embodiment adopts the mode of going forward one by one to describe in this specification, and what each embodiment stressed is and the difference of other embodiment that identical similar part is mutually referring to getting final product between each embodiment.For the disclosed terminal of embodiment, because it is corresponding with the embodiment disclosed method, so description is fairly simple, relevant part partly illustrates referring to method and gets final product.
The professional can also further recognize, unit and the algorithm steps of each example of describing in conjunction with embodiment disclosed herein can be realized in the mode that electronic hardware or electronic hardware combine with computer software.For the interchangeability of declaratives hardware and software clearly, composition and the step of each example described in general manner according to function in the above description.These functions still are that way of hardware and software combination is carried out with hardware actually, depend on application-specific and the design constraint of technical scheme.The professional and technical personnel can specifically should be used for using distinct methods to realize described function to each, but this realization should not thought and exceeds scope of the present invention.
The method of describing in conjunction with embodiment disclosed herein or the step of algorithm can directly use the software module of hardware, processor execution, and perhaps the combination of the two is implemented.Software module can place the storage medium of any other form known in random asccess memory (RAM), internal memory, read-only memory (ROM), electrically programmable ROM, electrically erasable ROM, register, hard disk, moveable magnetic disc, CD-ROM or the technical field.
To the above-mentioned explanation of the disclosed embodiments, make this area professional and technical personnel can realize or use the present invention.Multiple modification to these embodiment will be apparent concerning those skilled in the art, and defined General Principle can realize under the situation that does not break away from the spirit or scope of the present invention in other embodiments herein.Therefore, the present invention will can not be restricted to these embodiment shown in this article, but will meet the wideest scope consistent with principle disclosed herein and features of novelty.
Claims (12)
1. one kind dynamically arranges the method that parameter is transmitted in speech business, it is characterized in that described method comprises:
Obtain the code/decode type of transmitting voice service;
According to described code/decode type the required transmission parameter of the described speech business of transmission is set.
2. as any described method in the claim 1, it is characterized in that described transmission parameter comprises: one or more in the network parameters such as peak transfer rate, minimum transmission rate, propagation delay time, shake, scheduling type.
3. as any described method among the claim 1-2, it is characterized in that, describedly according to described code/decode type transmission described speech business required transmission parameter be set and comprise:
Determine required peak transfer rate and the minimum transmission rate of the described speech business of transmission according to described code/decode type.
4. as any described method among the claim 1-2, it is characterized in that, describedly according to described code/decode type transmission described speech business required transmission parameter be set and comprise:
Calculate the required bandwidth of the described speech business of transmission according to described code/decode type;
According to the network quality levels of described required bandwidth correspondence, required propagation delay time, the shake of the described speech business of transmission is set.
5. as any described method among the claim 1-4, it is characterized in that described method also comprises:
Obtain described speech business and whether support the silence compression function;
Whether support the silence compression function setting to transmit the required scheduling type of described speech business according to described speech business.
6. whether method as claimed in claim 5 is characterized in that, describedly support the silence compression function setting to transmit the required scheduling type of described speech business according to described speech business to comprise:
When the silence compression function is supported in described speech business, described scheduling type is set is the real-time polling service ERTPS of expansion;
When the silence compression function was not supported in described speech business, it was Unsolicited Grant Service UGS that described scheduling type is set.
7. a terminal is characterized in that, comprising:
The code/decode type acquiring unit is for the code/decode type that obtains transmitting voice service;
The transmission parameter settings unit is used for according to described code/decode type the required transmission parameter of the described speech business of transmission being set.
8. terminal as claimed in claim 7 is characterized in that, described transmission parameter comprises: one or more in the network parameters such as peak transfer rate, minimum transmission rate, propagation delay time, shake, scheduling type.
9. as the described terminal of claim 7-8, it is characterized in that described transmission parameter settings unit comprises:
Transmission rate arranges subelement, is used for determining required peak transfer rate and the minimum transmission rate of the described speech business of transmission according to described code/decode type.
10. according to any described terminal among the claim 7-8, it is characterized in that described transmission parameter settings unit comprises:
The bandwidth calculation subelement is used for calculating the required bandwidth of the described speech business of transmission according to described code/decode type;
Other transmission parameter settings subelements are used for the network quality levels according to described required bandwidth correspondence, and required propagation delay time, the shake of the described speech business of transmission is set.
11. according to any described terminal among the claim 7-10, it is characterized in that described terminal also comprises:
Silence compression function support information acquiring unit is used for obtaining described speech business and whether supports the silence compression function;
Whether scheduling type arranges the unit, be used for supporting the silence compression function setting to transmit the required scheduling type of described speech business according to described speech business.
12. terminal according to claim 11 is characterized in that, described scheduling type arranges the unit and comprises:
The real-time polling service scheduling type arranges subelement, is used for when the silence compression function is supported in described speech business, described scheduling type is set is the real-time polling service ERTPS of expansion;
The Unsolicited Grant Service scheduling type arranges subelement, is used for when the silence compression function is not supported in described speech business, and it is Unsolicited Grant Service UGS that described scheduling type is set.
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Address after: 518129 Building 2, B District, Bantian HUAWEI base, Longgang District, Shenzhen, Guangdong. Patentee after: Huawei terminal (Shenzhen) Co.,Ltd. Address before: 518129 Building 2, B District, Bantian HUAWEI base, Longgang District, Shenzhen, Guangdong. Patentee before: HUAWEI DEVICE Co.,Ltd. |
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Effective date of registration: 20181218 Address after: 523808 Southern Factory Building (Phase I) Project B2 Production Plant-5, New Town Avenue, Songshan Lake High-tech Industrial Development Zone, Dongguan City, Guangdong Province Patentee after: HUAWEI DEVICE Co.,Ltd. Address before: 518129 Building 2, B District, Bantian HUAWEI base, Longgang District, Shenzhen, Guangdong. Patentee before: Huawei terminal (Shenzhen) Co.,Ltd. |
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Effective date of registration: 20210420 Address after: Unit 3401, unit a, building 6, Shenye Zhongcheng, No. 8089, Hongli West Road, Donghai community, Xiangmihu street, Futian District, Shenzhen, Guangdong 518040 Patentee after: Honor Device Co.,Ltd. Address before: Metro Songshan Lake high tech Industrial Development Zone, Guangdong Province, Dongguan City Road 523808 No. 2 South Factory (1) project B2 -5 production workshop Patentee before: HUAWEI DEVICE Co.,Ltd. |