CN1204764C - Processing method for implementing multiple voice services - Google Patents
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Abstract
The present invention discloses a method for realizing the processing of a plurality of speech services. On hardware, the method can alone support the sequential series connection of code conversion units of different speech services, and the number of the code conversion units in the series connection is the same as that of the supported speech services. In the processing of the speech services, each code conversion unit in the series connection judges if the received byte format of an uplink speech frame conforms with an established byte format of the code conversion unit, if true, then the code conversion unit is activated to process the speech frame, else the code conversion unit transparently transmits the speech frame. By utilizing the method, when a speech service is added, only hardware connection and part modules of DSP processing need simply modified, and redesign and great adjustment to a system are needless; thus, the upgrading cost is reduced, the system is convenient to maintain, and the inheritance to the original hardware is favorable.
Description
Technical field
The present invention relates to the speech business treatment technology, specifically, relate to a kind of processing method that realizes multiple speech business.
Background technology
In system of broadband wireless communication, speech coding is the technology of a key, its objective is under the prerequisite that guarantees certain subjective speech quality, reduces as far as possible and transmits the required communication bandwidth of voiceband user data, thereby effectively utilize the existing communication resource.It is purpose with low rate, high tone quality that speech coding is one, the technology of continuous development and perfection, along with the development in technology and market, full rate (FullRate), half rate (Half Rate), enhanced full rate three kinds of speech coding standards such as (Enhanced Full Rate) have successively been released.Therefore, the compatibility to multiple speech business just becomes one of basic demand of communication equipment.
In gsm system, transcoder (Transcoder) is a functional unit of realizing speech coding, and as shown in Figure 1, Fig. 1 is the gsm system schematic diagram.In gsm system, base transceiver station and base station controller are referred to as base station sub-system, and transcoder belongs to base station sub-system.Interface between base station sub-system and the mobile switching centre is disclosed standard interface, is called the A interface, and interface is an Abis interface between base transceiver station in the base station sub-system and the base station controller.Travelling carriage carries out encoding compression to the speech of input, forms speech frame.Speech frame is sent to base transceiver station through wave point, exchange to transcoder via base station controller again, by transcoder the speech frame decoding that receives is reduced into the pulse code modulation (pcm) signal, by mobile switching centre the PCM voice signal that receives is exchanged to public telephone network, be linked into user's phone set terminal at last.This communication direction abbreviates up direction as.In like manner, at down direction, the PCM voice signal is reduced into the PCM sample value via mobile station decodes again behind the transcoder coding.
Transcoder shown in Fig. 1 is made of a plurality of separate code conversion unit usually, and referring to shown in Figure 2, Fig. 2 is the structural representation of code conversion unit.Among the figure, a code conversion unit comprises Abis interface relay interface unit, digital signal processing unit (DSP), A interface relay interface unit.At up direction, the compressed speech frame that Abis interface TU Trunk Unit receives outputs to A interface TU Trunk Unit after carrying out decoding processing through the DSP unit; At down direction, then the data that the A interface is received are carried out encoding process.Because DSP unit and relay interface unit are fixedly connected, therefore if will support multiple speech business on the A interface physical link, then DSP need support the computing of multiple voice coding and decoding, and this internal memory and disposal ability to the DSP device has all proposed higher requirement.
At present, the code conversion unit supports the method for more voice business to have two kinds in the base station system, and a kind of is the method for DSP in the upgrading code conversion unit, referring to shown in Figure 3.When there is the demand of newly-increased voice service in system, for reaching the purpose of on the port of same TU Trunk Unit, supporting all voice services, often to upgrade, to satisfy the needed program of multiple speech coding algorithm, datarams and serviceability to the DSP unit.Upgrading DSP unit, the hardware designs that needs to change transcoder also can't be accomplished effective succession to original hardware simultaneously, upgrading has brought comparatively serious hardware cost loss to system business for this.
Another kind is between TU Trunk Unit and DSP unit switching net to be set, and determines the method for connection between the two by software control net sheet, as shown in Figure 4.In transcoder, when voice link is set up, system is according to different voice services, corresponding D SP unit and TU Trunk Unit are linked up, for example, voice service 1 can only be supported in the DSP unit 1 among Fig. 4, and voice service 2 is supported in 2 of DSP unit, system is according to the difference of voice service kind, and decision TU Trunk Unit with which DSP unit links to each other.This method makes the flexibility of hardware configuration increase, and original hardware has also been had inheritance preferably.But the complexity of system greatly increases, and is higher to the designing requirement of hardware configuration and Control Software.
Summary of the invention
The object of the present invention is to provide a kind of processing method that realizes multiple speech business, make it can realize supporting on the same physical location port more voice business.
The present invention is achieved by the following technical solutions:
A kind of processing method that realizes multiple speech business, this method comprises the following steps:
A. in base station sub-system, independently support the code conversion unit of different phonetic business to be connected in series successively function;
B. whether the bit format of the ascending voice frame that receives of Chuan Jie each code conversion unit judges conforms to the bit format of making an appointment in this code conversion unit, if, then activate this code conversion unit, descending, upstream data are carried out speech coding and decoding processing respectively, otherwise the code conversion unit carries out transparent transmission with speech frame.
Wherein, described step B further comprises, whether comprise the bit format that causes next code conversion unit mistake to activate with the decoded pulse code modulation (pcm) speech data of the code conversion unit judges that is activated that the A interface does not directly link to each other, if, then revise frame synchronization, otherwise data are exported normally by the linear prediction of voice sampling point.
Comprise that further the set activation marker also detects first flag of frame after the described active coding converter unit of step B, judge that whether the current speech frame is first frame after the activation, if then carry out initialization.
The described transparent transmission of step B comprises the reset activation sign, data is not changed, and forward corresponding outlet to according to original data flow direction.
In addition, described initialization comprises the static variable that resets, all kinds of signs of set, and with the zero clearing of output buffers district.
The number of the code conversion unit that is connected in series equates with the number of the speech business kind of being supported, and will be on up direction in the last code conversion unit output interface be connected in series with the input interface of next code conversion unit.Described bit format is frame synchronization and frame ordering word.
Use method of the present invention, when increasing speech business newly, only need carry out the connection of hardware simply and revise DSP section processes process, do not need system is carried out big adjustment, reduced the required cost of upgrading, be convenient to safeguard that it is good to have the inheritance of original hardware, the software alteration amount is few, the simple advantage of implementation, and the propagation delay time that the serial connection back is increased is to the not influence of subjective quality of speech.
Description of drawings
Fig. 1 is the gsm system schematic diagram;
Fig. 2 is the structural representation of code conversion unit;
Fig. 3 is at present by the structural representation of upgrading DSP unit with upgrading code conversion unit;
Fig. 4 at present by software control net sheet to select the schematic diagram of DSP unit for use;
Fig. 5 is the structural representation of code conversion of the present invention unit serial connection;
Fig. 6 is code conversion cells D SP process chart under the code conversion unit single mode of operation;
Fig. 7 is a structural representation of supporting the code conversion unit of two kinds of speech businesses in the embodiment of the invention 2;
Fig. 8 is the DSP process chart of code conversion unit 1 under the serial connection mode of operation;
Fig. 9 is the DSP process chart of code conversion unit 2 under the serial connection mode of operation.
Embodiment
The multiple speech data of support of the present invention is for supporting one or more speech data, for make purpose of the present invention, technical scheme, and advantage clearer, followingly support a kind of embodiment 1 of speech business and the embodiment 2 of two kind of speech business with reference to accompanying drawing and with the code conversion unit, the present invention is described in more detail.
Referring to shown in Figure 5, will be on function separate code conversion unit 1, code conversion unit 2 ... code conversion unit N is connected in series successively, promptly the A interface TU Trunk Unit in the last code conversion of the up direction of communication unit and the Abis interface TU Trunk Unit of next code conversion unit are serially connected, for example, A interface TU Trunk Unit 512 in the code conversion unit 1 is connected in series with the Abis interface TU Trunk Unit 520 in the code conversion unit 2, and available like this N code conversion unit is connected in series successively and forms one with the synthetic code conversion unit of cascade system.Wherein, 1 the support voice business 1 in code conversion unit, 2 the support voice business 2 in code conversion unit ... the professional N of N the support voice in code conversion unit, N code conversion unit is connected in series to support the speech business of N kind simultaneously on same port.
The serial connection of code conversion unit needs the support of DSP (Digital Signal Processing) unit, and the DSP unit relies on the entrained business information of speech frame of Abis interface to activate the coding and decoding operation.Speech frame generally is divided into three parts: frame synchronization, frame ordering word, frame data.Frame synchronization is the bit series through specific arrangement, indicates the initial of a frame and bit synchronous position.The frame ordering word is carrying the control information relevant with speech business.Frame data then are through the frame bit stream after the speech coding.The activation of service of transcoder and deexcitation are to rely on frame ordering word and frame synchronization in the Abis interface speech frame to control respectively.
Embodiment 1:
A kind of speech data is supported in each code conversion unit, therefore, when needs are supported a kind of speech data, then only needs a code conversion unit, needn't be connected in series a plurality of code conversions unit, and at this moment, the code conversion cell operation is in single-mode.
Referring to shown in Figure 6, Fig. 6 is code conversion cells D SP process chart under the code conversion unit single mode of operation.The DSP handling process is divided into following steps:
(1) execution in step 601, and at up direction, the Abis interface rate of received data is the compress speech frame data of 16kbps, and at down direction, the A interface receives voice PCM data, and the speed of data is 64kbps;
(2) execution in step 602, the code conversion unit carries out frame synchronization and frame ordering word analysis with the data of the up direction that receives, whether judgment frame conforms to the frame ordering word with the frame synchronization of code conversion unit agreement with the frame ordering word synchronously, if be not inconsistent with agreement, then execution in step 607, the speech business activation marker that resets returns step 601 then; If receive the frame synchronization and the frame ordering word of agreement, then execution in step 603, and the set activation marker detects first flag of frame;
(3) execution in step 604, whether code conversion unit judges current speech frame is first frame after activating, if first frame after activating, then execution in step 605, and first frame is carried out initialization, i.e. static variable in the reset algorithm, all kinds of signs of set, and execution in step 606 again behind the zero clearing output buffer, first frame after activate is direct execution in step 606 then, and descending, upstream data are carried out speech coding, decoding processing respectively.
(4) execution in step 608, data were exported by set frame format after the code conversion unit will carry out computing, and the data of down direction after with encoding compression are formed compressed speech frame output, and bit rate is 16kbps, up direction returns step 601 at last for the 64kbps PCM speech data that the decoding back generates.If end of conversation, then Abis interface can't receive speech frame, the Program reset activation marker.
Embodiment 2:
Referring to shown in Figure 7.When needs are supported two kinds of speech businesses, the Abis interface relay interface unit 720 in A interface relay interface unit in the code conversion unit 1 712 and the code conversion unit 2 is serially connected, form a synthetic code conversion unit.At up direction, the speech frame after the compression is by 1 input of code conversion unit; PCM sampling point after the reduction is by 2 outputs of code conversion unit, and vice versa.On function, code conversion unit 1 is separate with code conversion unit 2, for example, 1 the support voice business 1 in code conversion unit, 2 the support voice business 2 in code conversion unit, both are connected in series to support two kinds of speech businesses simultaneously on same port.
The code conversion unit is under the mode of operation of serial connection, and when conversation was speech business 1, code conversion unit 1 was in activated state, and code conversion unit 2 is in the transparent transmission state, and promptly the uplink and downlink data do not process, and directly transmits; If be in the talking state of speech business 2, then code conversion unit 2 activates code conversion unit 1 transparent transmission.For supporting this function, the DSP cell processing of two kinds of code conversion unit need be done some adjustment.
Referring to shown in Figure 8, Fig. 8 is the DSP cell processing flow chart of code conversion unit 1 under the serial connection mode of operation.For code conversion unit 1, DSP unit 711 handling processes are divided into following steps:
(1) execution in step 801, and at up direction, the Abis interface rate of received data is 16kbps compress speech frame data, and at down direction, A interface rate of received data is the voice PCM data of 64kbps;
(2) execution in step 802, the code conversion unit carries out frame synchronization and frame ordering word analysis with the data of the up direction that receives, whether judgment frame conforms to the frame ordering word with the frame synchronization of code conversion unit agreement with the frame ordering word synchronously, if be not inconsistent with agreement, then execution in step 811, the speech business activation marker resets, follow execution in step 812, with data transparency transmission up, descending reception, promptly the data of receiving are not changed, and according to original data flow direction, forward in the corresponding outlet, return step 801 then; If receive the frame synchronization and the frame ordering word of agreement, then execution in step 803, and the set activation marker detects first flag of frame;
(3) execution in step 804, whether code conversion unit judges current speech frame is first frame after activating, if first frame after activating, then execution in step 805, and first frame is carried out initialization, i.e. static variable in the reset algorithm, all kinds of signs of set, and execution in step 806 again behind the zero clearing output buffer, first frame after activate is direct execution in step 806 then, and descending, upstream data are carried out speech coding, decoding computing respectively.
(4) execution in step 807, and for example GSM 08.60 agreement is carried out frame synchronization and the frame ordering word format detects according to band inner control agreement to decoded PCM speech data in the code conversion unit;
(5) execution in step 808, and whether the up PCM speech data of code conversion unit judges comprises the frame synchronization that can cause code conversion unit 2 mistakes to activate, if the result who detects demonstration does not comprise the frame synchronization that can cause code conversion unit 2 mistakes to activate, then forward step 810 to; Otherwise execution in step 809, by the mode of linear prediction, according to band inner control agreement for example GSM 08.60 agreement this bit format is made amendment, do not activated by mistake and do not influenced subjective speech quality to guarantee code conversion unit 2, execution in step 810 again.
(6) execution in step 810, data were exported by set frame format after the code conversion unit will carry out computing, and the data of down direction after with encoding compression are formed compressed speech frame output, and bit rate is 16kbps, up direction returns step 801 at last for the 64kbps PCM speech data that the decoding back generates.If end of conversation, then Abis interface can't receive speech frame, the Program reset activation marker.
Because code conversion unit 2 is the last code converter units that are connected in series, promptly this code conversion unit directly and the A interface join, therefore the handling process of code conversion unit 2 is similar to single mode of operation, and just under the unactivated situation of speech business, the uplink and downlink data should be carried out transparent transmission.DSP unit 721 treatment steps are as follows:
(1) execution in step 901, and at up direction, the Abis interface rate of received data is 16kbps compress speech frame data, and at down direction, A interface rate of received data is 64kbps voice PCM data;
(2) execution in step 902, the code conversion unit carries out frame synchronization and frame ordering word analysis with the data of the up direction that receives, whether judgment frame conforms to the frame ordering word with the frame synchronization of code conversion unit agreement with the frame ordering word synchronously, if be not inconsistent with agreement, then execution in step 908, the speech business activation marker resets, follow execution in step 909, with data transparency transmission up, descending reception, promptly the data of receiving are not changed, and according to original data flow direction, forward in the corresponding outlet, return step 901 then; If receive the frame synchronization and the frame ordering word of agreement, then execution in step 903, and the set activation marker detects first flag of frame;
(3) execution in step 904, whether code conversion unit judges current speech frame is first frame after activating, if first frame after activating, then execution in step 905, and first frame is carried out initialization, i.e. static variable in the reset algorithm, all kinds of signs of set, and execution in step 906 again behind the zero clearing output buffer, first frame after activate is direct execution in step 906 then, and descending, upstream data are carried out speech coding, decoding computing respectively.
(4) execution in step 907, data were exported by set frame format after the code conversion unit will carry out computing, and the data of down direction after with encoding compression are formed compressed speech frame output, and bit rate is 16kbps, up direction returns step 901 at last for the 64kbps PCM speech data that the decoding back generates.If end of conversation, then Abis interface can't receive speech frame, the Program reset activation marker.
For being connected in series the independently generated code converter unit of code conversion unit formation of two above functions, the DSP cell processing flow process in the code conversion unit that is connected in series is similar to embodiment 2, and promptly DSP cell processing flow process mainly comprises:
Under nonactivated operating state, the uplink and downlink data that receive are carried out transparent transmission, promptly the data of receiving are not changed, and, forward in the corresponding outlet, in order to avoid the Data Receiving and the analysis of next unit impacted according to original data flow direction.
Under the operating state that activates, for the transcoder that directly links to each other with the A interface, for example code conversion unit N among Fig. 5 or the code conversion unit 2 among Fig. 7 are the same under its workflow and the single mode of operation.For the code conversion unit that does not directly link to each other with the A interface, for example code conversion unit 1 among Fig. 7 or the code conversion unit 1 among Fig. 5, code conversion unit 2.... are until code conversion unit N-1, need carry out frame synchronization and the detection of frame ordering word to decoded PCM speech data, in order to avoid the frame synchronization coincidence of the frame synchronization in the PCM speech data and next code conversion unit, thereby cause the mistake of next code conversion unit to activate.If the result who detects shows that up PCM speech data does not comprise the frame synchronization that can cause the next unit mistake to activate, then data are normally exported; Otherwise to this frame synchronization be made amendment by the mode of voice sampling point linear prediction, to guarantee that next unit is not activated by mistake.
Claims (8)
1, a kind of processing method that realizes multiple speech business is characterized in that, this method comprises the following steps:
A. in base station sub-system, will independently support the code conversion unit of different phonetic business to be connected in series successively;
B. whether the bit format of the ascending voice frame that receives of Chuan Jie each code conversion unit judges conforms to the bit format of making an appointment in this code conversion unit, if, then activate this code conversion unit, descending, upstream data are carried out encoding and decoding speech respectively handle, otherwise the code conversion unit carries out transparent transmission with speech frame.
2, processing method according to claim 1, it is characterized in that, described step B further comprises, whether comprise the bit format that causes next code conversion unit mistake to activate with the decoded pulse code modulation speech data of the code conversion unit judges that is activated that the A interface does not directly link to each other, if, then revise bit format, otherwise data are exported normally by the linear prediction of voice sampling point.
3, processing method according to claim 1 and 2, it is characterized in that, further comprise after the described active coding converter unit of the step B step, the set activation marker also detects first flag of frame, judge that whether the current speech frame is first frame after activating, if then carry out initialization, otherwise carry out described step of descending, upstream data being carried out the encoding and decoding speech processing respectively.
4, processing method according to claim 3 is characterized in that, described initialization comprises the static variable that resets, all kinds of signs of set, and with the zero clearing of output buffers district.
5, processing method according to claim 1 and 2 is characterized in that, the described transparent transmission of step B comprises the reset activation sign, data is not changed, and forward corresponding outlet to according to original data flow direction.
6, processing method according to claim 1 is characterized in that, the number of the code conversion unit that is connected in series equates with the number of the speech business kind of being supported.
7, according to claim 1 or 6 described processing methods, it is characterized in that, will on up direction, be connected in series with next code conversion unit input interface by the output interface in the last code conversion unit.
8, processing method according to claim 1 is characterized in that, described bit format is frame synchronization and frame ordering word.
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| CN 02121475 CN1204764C (en) | 2002-06-25 | 2002-06-25 | Processing method for implementing multiple voice services |
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Effective date of registration: 20210425 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: 518057 HUAWEI building, road, Shenzhen science and Technology Park Patentee before: HUAWEI TECHNOLOGIES Co.,Ltd. |
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