




技术领域technical field
本发明涉及通信技术领域,特别是一种增强语音质量的方法和基站。The invention relates to the technical field of communication, in particular to a method and a base station for enhancing voice quality.
背景技术Background technique
在移动通信中,为了有效利用空中接口的传输带宽,现有技术中主要是采用对语音信号进行压缩编码以提高带宽利用率的方法。以GSM通信系统(Global System for Mobile communications,全球移动通信系统)为例,FR(Full Rate speech codec,全速率语音编解码)语音编解码是将64Kbps的PCM(Pulse Code Modulation,脉冲编码调制)语音信号压缩到13Kbps,HR(HalfRate speech codec,半速率语音编解码)语音业务则将64Kbps的PCM语音信号压缩到5.6Kbps。在半速率AMR(Adaptive Muti-Rate speech codec,自适应多速率语音编解码)业务下,则最低可将64Kbps的PCM语音信号压缩到4.75Kbps。半速率语音编码下,由于对原始语音的压缩比非常高,编码产生的语音参数都非常珍贵,一旦因传输误码等原因导致这些参数发生错误,将对语音质量产生很大的损伤。In mobile communication, in order to effectively utilize the transmission bandwidth of the air interface, in the prior art, a method of compressing and encoding voice signals is mainly used to improve bandwidth utilization. Taking the GSM communication system (Global System for Mobile communications, Global System for Mobile Communications) as an example, FR (Full Rate speech codec, full rate voice codec) voice codec is the 64Kbps PCM (Pulse Code Modulation, pulse code modulation) voice The signal is compressed to 13Kbps, and the HR (HalfRate speech codec, half-rate voice codec) voice service compresses the 64Kbps PCM voice signal to 5.6Kbps. Under the half-rate AMR (Adaptive Muti-Rate speech codec, adaptive multi-rate speech codec) service, the minimum 64Kbps PCM voice signal can be compressed to 4.75Kbps. Under half-rate speech coding, because the compression ratio of the original speech is very high, the speech parameters generated by the coding are very precious. Once these parameters are wrong due to transmission errors and other reasons, the speech quality will be greatly damaged.
如图1所示,是GSM系统的结构框图。其中BSS(Base StationSubsystem,基站子系统)通过无线方式实现发送和接收信息,并进行无线资源的管理;NSS(Network SubSystem,网络子系统)负责通信业务的管理,处理外部网络和移动用户呼叫的交换。OSS(Operating Support System,操作支持子系统)则为运营商提供对这些实际运行部分的控制、维护和管理。BSS由BTS(Base Transceiver Station,基站收发信台)和BSC(Base StationController,基站控制器)两部分功能实体组成。在BSS侧还包括TRAU(Transcoding and Rate Adaptation Unit,速率适配码变换单元),实现数据业务速率适配向A接口速率的适配,以及低速率语音编码方式向A接口64Kbps语音编码方式的转化。按照GSM协议规定,TRAU可以放置在BTS侧,也可以放置在BSC侧。为了节省BTS和BSC间传输带宽,一般将TRAU放置在BSC侧。TRAU和BTS之间采用串行的同步帧进行数据传输,称为TRAU帧。As shown in Figure 1, it is a structural block diagram of the GSM system. Among them, BSS (Base Station Subsystem, base station subsystem) realizes sending and receiving information wirelessly, and manages wireless resources; NSS (Network SubSystem, network subsystem) is responsible for the management of communication services, and handles the exchange of external networks and mobile user calls . OSS (Operating Support System, operating support subsystem) provides operators with control, maintenance and management of these actual operating parts. The BSS consists of two functional entities: BTS (Base Transceiver Station, Base Transceiver Station) and BSC (Base Station Controller, Base Station Controller). On the BSS side, it also includes TRAU (Transcoding and Rate Adaptation Unit, rate adaptation code conversion unit), which realizes the adaptation of data service rate adaptation to the rate of A interface, and the conversion of low-rate speech coding mode to 64Kbps speech coding mode of A interface . According to the GSM protocol, TRAU can be placed on the BTS side or on the BSC side. In order to save the transmission bandwidth between the BTS and the BSC, generally the TRAU is placed on the BSC side. Serial synchronous frames are used for data transmission between TRAU and BTS, which are called TRAU frames.
GSM系统中,BTS和TRAU单元之间通过TRAU帧来传送业务数据,TRAU帧每隔20ms发送一帧,在BTS和TRAU之间提供8Kbps(半速率语音编码方式下,每帧160个比特)或16Kbps(全速率语音编码方式下,每帧320个比特)的传输带宽。参考GSM08.60、08.61协议,TRAU帧有同步比特、控制比特和数据比特三部分组成。帧同步比特是TRAU帧中一定数量的0和1,如图2中灰色部分所示。TRAU帧0、1同步比特的数量和位置是用来判断帧的开始和结束位置的重要依据,称为帧同步格式。图2为半速率AMR业务7.4Kbps速率下的TRAU帧格式,不同业务的TRAU帧的格式不同。由于业务数据处理必须以完整的TRAU帧为单位,因此BTS和TRAU接收到完整的TRAU帧,是进行业务处理的前提。TRAU帧中除传送语音编码参数等业务数据之外,还传送基站与TRAU之间的控制信息。比如编解码的速率、当前帧是否可用等。In the GSM system, BTS and TRAU units transmit business data through TRAU frames, and TRAU frames are sent every 20ms, providing 8Kbps between BTS and TRAU (160 bits per frame in half-rate speech coding mode) or 16Kbps (320 bits per frame in full-rate speech coding mode) transmission bandwidth. Referring to the GSM08.60 and 08.61 protocols, the TRAU frame consists of three parts: synchronization bits, control bits and data bits. The frame synchronization bit is a certain number of 0s and 1s in the TRAU frame, as shown in the gray part in Fig. 2 . The number and position of the synchronization bits of
为了提升语音质量,GSM系统中引入AMR编解码方式。在半速率AMR业务下,支持4.75、5.15、5.9、6.7、7.4Kbps等5种速率的语音编码方式。对于不同的速率,每个用户在BTS和TRAU之间占用的传输带宽都是8Kbps。因此不同速率的语音编码TRAU帧同步格式一般也是不同的,速率越高,则同步比特越少。7.4Kbps速率的TRAU帧同步头格式如图2所示。可以看出,7.4Kbps速率TRAU帧由160个比特组成。这种TRAU帧的0,1同步比特非常少(灰色部分显示的为帧同步比特,只有6个),0,1出现的位置比较分散。In order to improve the voice quality, the AMR codec is introduced into the GSM system. Under the half-rate AMR service, it supports 5 kinds of voice coding modes including 4.75, 5.15, 5.9, 6.7, and 7.4Kbps. For different rates, the transmission bandwidth occupied by each user between BTS and TRAU is 8Kbps. Therefore, speech coding TRAU frame synchronization formats of different rates are generally different, and the higher the rate, the fewer the synchronization bits. Figure 2 shows the format of the TRAU frame synchronization header at a rate of 7.4Kbps. It can be seen that the 7.4Kbps rate TRAU frame consists of 160 bits. The 0, 1 synchronization bits of this TRAU frame are very few (the gray part shows the frame synchronization bits, only 6), and the positions where 0, 1 appear are scattered.
申请人在实现本发明的过程中发现,由于目前BTS和TRAU之间采用TRAU帧传送数据,其中采用CRC(Cyclic Redundancy Check,循环冗余校验)的只有检错功能,无纠错功能。而且,当传输误码导致接收端CRC计算不匹配时,即使只错了其中的1个比特,因CRC校验无纠错功能,也只能将这一帧丢弃。也就是说在基站和TRAU之间的数据传输无纠错编码功能。因此数据传输的错误率比较高,因此,在出现数据错误传输的情况下,导致语音质量降低。During the process of implementing the present invention, the applicant found that since the TRAU frame is used to transmit data between the BTS and TRAU, the CRC (Cyclic Redundancy Check) has only an error detection function and no error correction function. Moreover, when the transmission error causes the CRC calculation mismatch at the receiving end, even if only one bit is wrong, the frame can only be discarded because the CRC check has no error correction function. That is to say, the data transmission between the base station and the TRAU has no error correction coding function. Therefore, the error rate of data transmission is relatively high, and therefore, in the case of erroneous data transmission, the voice quality is degraded.
发明内容Contents of the invention
本发明一个或多个实施例的目的在于提供一种增强语音质量的方法和基站,采用本发明的方法和基站,能够通过扩展数据传输带宽,在数据传输过程中增加纠错机制,以实现降低传输误码率,提高语音质量。The purpose of one or more embodiments of the present invention is to provide a method and base station for enhancing voice quality. By adopting the method and base station of the present invention, an error correction mechanism can be added in the data transmission process by expanding the data transmission bandwidth, so as to reduce the Transmission bit error rate, improve voice quality.
为解决上述问题,本发明实施例提供了一种增强语音质量的方法,包括In order to solve the above problems, an embodiment of the present invention provides a method for enhancing voice quality, including
步骤:step:
扩展BTS与BSC之间的传输带宽;Extend the transmission bandwidth between BTS and BSC;
在TRAU和所述BTS之间的TRAU帧中增加纠错机制。An error correction mechanism is added in the TRAU frame between the TRAU and the BTS.
还提供了一种增强语音质量的基站,包括:There is also provided a base station for enhanced speech quality, comprising:
传输带宽扩展单元,用于:扩展BTS与BSC之间的传输带宽;A transmission bandwidth extension unit, configured to: extend the transmission bandwidth between the BTS and the BSC;
TRAU传输纠错单元,用于:在TRAU和所述BTS之间的TRAU帧中增加纠错机制。The TRAU transmits an error correction unit, configured to: add an error correction mechanism in the TRAU frame between the TRAU and the BTS.
与现有技术相比,本发明实施例具有以下优点:Compared with the prior art, the embodiment of the present invention has the following advantages:
通过扩展BTS与BSC之间的传输带宽,在TRAU和BTS之间的TRAU帧中增加纠错机制,能够降低数据传输的因为带宽不够、没有纠错机制而导致的高概率的误码率,提升语音通信质量。By expanding the transmission bandwidth between BTS and BSC, an error correction mechanism is added to the TRAU frame between TRAU and BTS, which can reduce the high-probability bit error rate of data transmission due to insufficient bandwidth and no error correction mechanism, and improve Voice communication quality.
附图说明Description of drawings
图1所示是GSM系统的结构框图;Shown in Fig. 1 is the structural block diagram of GSM system;
图2所示是半速率AMR业务7.40Kbps速率TRAU帧格式结构图;Figure 2 is a frame format diagram of the half-rate AMR service 7.40Kbps rate TRAU frame;
图3所示是本发明的增强语音质量的方法的第一个实施例的流程图;Shown in Fig. 3 is the flowchart of the first embodiment of the method for enhancing voice quality of the present invention;
图4所示是本发明的增强语音质量的方法的第二个实施例的流程图;Shown in Fig. 4 is the flowchart of the second embodiment of the method for enhancing voice quality of the present invention;
图5所示是本发明的增强语音质量的基站的第一个实施例的框图;Fig. 5 is the block diagram of the first embodiment of the base station of enhancing voice quality of the present invention;
图6所示是本发明的各实施例对语音参数纠错编解码过程示意图。FIG. 6 is a schematic diagram of the speech parameter error correction encoding and decoding process according to various embodiments of the present invention.
具体实施方式Detailed ways
下面结合附图对本发明实施例的具体实施方式做进一步的详细阐述。The specific implementation manners of the embodiments of the present invention will be further described in detail below in conjunction with the accompanying drawings.
如图3所示,是本发明的增强语音质量的方法的第一个实施例,包括步骤:As shown in Figure 3, it is the first embodiment of the method for enhancing voice quality of the present invention, comprising steps:
S301、扩展BTS与BSC之间的传输带宽;S301. Extend the transmission bandwidth between the BTS and the BSC;
S302、在TRAU和BTS之间的TRAU帧中增加纠错机制。S302. Add an error correction mechanism in the TRAU frame between the TRAU and the BTS.
通过本发明的上述实施例,扩展BTS与BSC之间的传输带宽,在TRAU和BTS之间的TRAU帧中增加纠错机制,能够降低数据传输的误码率,提升语音通信质量。Through the above embodiments of the present invention, the transmission bandwidth between the BTS and the BSC is extended, and an error correction mechanism is added in the TRAU frame between the TRAU and the BTS, so that the bit error rate of data transmission can be reduced and the quality of voice communication can be improved.
其中,在上述实施例中,所述BTS与所述BSC之间的数据可以是半速率编码格式。Wherein, in the above embodiment, the data between the BTS and the BSC may be in a half-rate encoding format.
其中,在进行上述实施例之前,可以包括步骤:打开预设的抗误码开关,所述抗误码开关可以设置在BTS上,所述抗误码开关包括站点抗误码开关或用户抗误码开关。Wherein, before carrying out the above-mentioned embodiment, it may include the step of: opening a preset anti-error switch, the anti-error switch may be set on the BTS, and the anti-error switch includes a site anti-error switch or a user anti-error switch code switch.
其中,如果所述BTS与所述BSC之间采用微波传输,或所述BTS与所述BSC之间存在传输误码,或其他需要提高所述BTS与所述BSC之间数据传输质量,则修改所述BTS或BSC配置数据,打开站点ABIS接口抗误码开关或用户ABIS接口抗误码开关。其中,所述BTS与所述BSC之间出现传输误码的原因可能是传输设备故障等原因。Wherein, if microwave transmission is used between the BTS and the BSC, or there is a transmission error between the BTS and the BSC, or other needs to improve the quality of data transmission between the BTS and the BSC, modify For the configuration data of the BTS or BSC, turn on the anti-error switch of the station ABIS interface or the anti-error switch of the user ABIS interface. Wherein, the cause of a transmission error between the BTS and the BSC may be a transmission equipment failure or the like.
其中,所述抗误码开关可以是站点抗误码开关或用户抗误码开关,如果是站点抗误码开关打开,则对于在本站点下所有采用半速率语音通话的用户,将所述站点的所述BTS与所述BSC之间的传输带宽由8Kbps扩展为16Kbps,采用类似全速率AMR业务的帧同步格式传送半速率下语音的数据,并且在TRAU和BTS之间的TRAU帧中增加纠错机制;如果是用户抗误码开关打开,则对于此用户采用半速率语音通话时,BTS和TRAU只是针对此用户进行抗误码处理,将此用户数据传输过程中所经过的所有所述BTS与所述BSC之间的传输带宽由8Kbps扩展为16Kbps,采用类似全速率AMR业务的帧同步格式传送半速率下语音的数据,并且在TRAU和BTS之间的TRAU帧中增加纠错机制;如果打开的所述抗误码开关是站点抗误码开关和用户抗误码开关,则将所述站点的、所述用户数据经过的所有所述BTS与所述BSC之间的传输带宽由8Kbps扩展为16Kbps,在所述站点的、所述用户数据经过的所有TRAU和所述BTS之间的TRAU帧中增加纠错机制。Wherein, the anti-error switch can be a site anti-error switch or a user anti-error switch, if the site anti-error switch is turned on, then for all users who use half-rate voice calls under this site, the described The transmission bandwidth between the BTS and the BSC of the site is expanded from 8Kbps to 16Kbps, and the frame synchronization format similar to the full-rate AMR service is used to transmit voice data at half-rate, and the TRAU frame between TRAU and BTS is increased Error correction mechanism; if the user's anti-error code switch is turned on, when the user uses half-rate voice calls, BTS and TRAU only perform anti-error The transmission bandwidth between the BTS and the BSC is expanded from 8Kbps to 16Kbps, using a frame synchronization format similar to the full-rate AMR service to transmit voice data at half-rate, and adding an error correction mechanism in the TRAU frame between the TRAU and the BTS; If the anti-error switch that is turned on is a site anti-error switch and a user anti-error switch, then the transmission bandwidth between all the BTSs and the BSC that the user data passes through in the station is changed from 8Kbps It is extended to 16Kbps, and an error correction mechanism is added in TRAU frames between all TRAUs of the station and the user data passing through and the BTS.
其中,上述实施例中所述的增加的纠错机制具体为:Wherein, the added error correction mechanism described in the above embodiment is specifically:
当TRAU和BTS之间的带宽由8Kbps扩展为16Kbps之后,由于带宽的限制,无法对所有的业务数据增加纠错机制,本方案优选一些重要的业务数据如控制比特、CRC校验比特和语音参数进行分组纠错编解码,当然,也可以选择上述三种业务数据中的一种、二种或全部,以提高数据传输的质量。进行分组纠错编解码的方法可以采用比较通行的行之有效的方法,这属于所属领域技术人员的常识。对于所属领域的技术人员而言,也可以只对重要的语音参数进行纠错编码,重要的语音参数是根据相关的标准协议的规定,根据语音参数的重要程度,将语音参数分为IA、IB、II类。当然,与滤波器有关的语音参数也是重要的语音参数。其中当然在其他实施例中,如果认为有必要,可以将TRAU和BTS之间的带宽扩展到更宽的程度,从而对所有的传输数据增加纠错机制,全面提高数据传输的质量。对所属领域的技术人员而言,这种变化没有超出本发明的保护范围。When the bandwidth between TRAU and BTS is expanded from 8Kbps to 16Kbps, due to bandwidth limitations, it is impossible to add error correction mechanisms to all business data. This solution prefers some important business data such as control bits, CRC check bits and voice parameters. To perform packet error correction codec, of course, one, two or all of the above three types of service data can also be selected to improve the quality of data transmission. The method for performing packet error correction encoding and decoding can be a relatively popular and effective method, which belongs to the common knowledge of those skilled in the art. For those skilled in the art, it is also possible to only perform error correction coding on important speech parameters. The important speech parameters are divided into IA and IB according to the importance of the speech parameters according to the provisions of relevant standard protocols. , Class II. Of course, the speech parameters related to the filter are also important speech parameters. Of course, in other embodiments, if necessary, the bandwidth between the TRAU and the BTS can be extended to a wider extent, so as to add an error correction mechanism to all transmitted data and improve the quality of data transmission in an all-round way. For those skilled in the art, such changes do not go beyond the protection scope of the present invention.
如图4所示,通过在BTS中增加站点抗误码开关配置,选择有抗误码开头的特定的基站打开Abis接口抗误码功能,就得到了是本发明的增强语音质量的方法的第二个实施例,对于半速率7.4Kbps AMR业务,本实施例中采用艾布朗森码进行纠错编码,本方法实施例包括步骤:As shown in Figure 4, by increasing the station anti-error switch configuration in the BTS, selecting the specific base station with the beginning of the anti-error code to open the Abis interface anti-error function, the first method of enhancing the voice quality of the present invention is obtained Two embodiments, for the half-rate 7.4Kbps AMR service, the Abronson code is used for error correction coding in this embodiment, and this method embodiment includes steps:
S401、当某BTS与BSC之间的传输采用微波时,或者因传输设备故障,BTS和BSC之间存在传输误码时,修改该BSC数据配置,根据修改后的BSC配置,BTS打开站点Abis接口抗误码开关;当然,在其他实施例中,也可以将BSC中的数据配置设置成站点Abis接口抗误码开关为常开状态,只要BSS进行初始化,就打开站点Abis接口抗误码开关;S401. When the transmission between a BTS and the BSC uses microwaves, or when there is a transmission error between the BTS and the BSC due to transmission equipment failure, modify the data configuration of the BSC, and according to the modified BSC configuration, the BTS opens the site Abis interface Anti-error switch; Certainly, in other embodiments, the data configuration in the BSC can also be set to the station Abis interface anti-error switch as a normally open state, as long as the BSS is initialized, the site Abis interface anti-error switch is turned on;
S402、所述BTS通过TRAU帧带内信令通知其对应的TRAU Abis接口抗误码开关打开;TRAU默认抗误码开关关闭,BTS通过TRAU帧带内信令通知TRAUAbis接口抗误码开关是否打开,如果站点Abis接口抗误码开关打开,则通过TRAU帧带内信令通知TRAU Abis接口抗误码开关打开;当然,在其他实施例中,TRAU默认抗误码开关状态也可以为开启。只有站点Abis接口抗误码开关和TRAU Abis接口抗误码开关都打开,才能启动后续的扩展传输带宽和增加纠错机制步骤。S402, the BTS notifies its corresponding TRAU Abis interface anti-error switch to open through the TRAU frame in-band signaling; the TRAU default anti-error switch is closed, and the BTS notifies the TRAU Abis interface whether the anti-error switch is turned on through the TRAU frame in-band signaling , if the station Abis interface anti-error switch is turned on, the TRAU frame in-band signaling is used to notify TRAU that the Abis interface anti-error switch is turned on; of course, in other embodiments, the default state of the TRAU anti-error switch can also be on. Only when both the anti-error switch of the site Abis interface and the anti-error switch of the TRAU Abis interface are turned on, can the subsequent steps of expanding the transmission bandwidth and adding the error correction mechanism be started.
S403、当站点抗误码开关打开时,则对于在本站点下所有采用半速率语音通话的用户,将BTS和TRAU之间的传输带宽由8Kbps扩展为16Kbps,采用类似全速率AMR业务的帧同步格式传送半速率下语音的数据;S403. When the site anti-error switch is turned on, for all users who use half-rate voice calls under this site, the transmission bandwidth between BTS and TRAU is expanded from 8Kbps to 16Kbps, and frames similar to full-rate AMR services are used The synchronous format transmits voice data at half rate;
S404、在将BTS和TRAU之间的传输带宽扩展为16KBbps后同时,在TRAU和BTS之间的TRAU帧中增加纠错机制。TRAU和BTS之间的纠错机制如下:S404. After expanding the transmission bandwidth between the BTS and the TRAU to 16KBbps, add an error correction mechanism in the TRAU frame between the TRAU and the BTS. The error correction mechanism between TRAU and BTS is as follows:
1)当TRAU和BTS之间的带宽由8Kbps扩展为16Kbps之后,由于带宽的限制,无法对所有的业务数据增加纠错功能,选择对语音质量影响重大的数据进行保护。当然,所属领域的技术人员根据本发明的方法,如果将带宽进一步拓宽,可以选择对更多的业务数据或所有的业务数据增加纠错功能,这种变化没有超出本发明的保护范围,是在本发明的方法启示下进行逻辑推导必然得出的。在本方案中,进行纠错编码处理的、重要的数据如下表所示:1) When the bandwidth between TRAU and BTS is expanded from 8Kbps to 16Kbps, due to bandwidth limitations, it is impossible to add error correction functions to all business data, and choose to protect data that has a significant impact on voice quality. Of course, according to the method of the present invention, if the bandwidth is further widened, those skilled in the art can choose to add an error correction function to more business data or all business data. This change does not exceed the scope of protection of the present invention. Under the enlightenment of the method of the present invention, logical derivation must be obtained. In this scheme, the important data for error correction coding processing are shown in the following table:
2)重要比特的分组纠错编解码技术2) Packet error correction codec technology for important bits
首先,将上述的64个重要比特进行排列,以3比特为单位进行分组。共可以分为22组。最后1组只有1个比特,后边补两个比特0组成一组。其次,综合考虑Abis接口传输误码模型,传输带宽和算法复杂度,选择合适的纠错编码技术对每个分组进行纠错编码。本方案选择的纠错编码技术为[7,3,4]艾布朗姆森码。其生成多项式为:First, the above-mentioned 64 important bits are arranged and grouped in units of 3 bits. There are 22 groups in total. The last group has only 1 bit, followed by two 0 bits to form a group. Secondly, comprehensively consider the Abis interface transmission error model, transmission bandwidth and algorithm complexity, and select the appropriate error correction coding technology to perform error correction coding on each packet. The error correction coding technology selected in this scheme is [7, 3, 4] Abramson code. Its generating polynomial is:
g(x)=(x+1`)(x3+x2+1)g(x)=(x+1`)(x3 +x2 +1)
艾布朗姆森码纠错编码技术具备单位错误纠错能力,并可以检测所有的单位错、两位错和奇数位错,长度≤4的突发错以及长度≤2的两个突发错。采用[7,3,4]艾布朗姆森码对22组重要比特进行编码,每组编码生成7个比特,共产生154个比特。The Abramson code error correction coding technology has the ability to correct single errors, and can detect all single errors, two-digit errors and odd-number errors, burst errors of length ≤ 4, and two burst errors of length ≤ 2. [7, 3, 4] Abramson codes are used to encode 22 groups of important bits, each group of codes generates 7 bits, and a total of 154 bits are generated.
除同步比特之外,其它的重要比特编帧方式如下:In addition to synchronization bits, other important bits are framed as follows:
1)控制比特C1~C5设置1) Control bit C1~C5 setting
如下表所示,控制比特C1~C5为当前TRAU帧的业务类型标志,前11种为目前GSM系统已有的TRAU帧类型,第12和第13种是新增加带纠错的半速率AMR、带纠错的HR两种帧类型;As shown in the table below, the control bits C1~C5 are the service type signs of the current TRAU frame, the first 11 are the existing TRAU frame types in the current GSM system, the 12th and 13th are newly added half-rate AMR with error correction, Two frame types of HR with error correction;
2)控制比特C6~C25设置2) Control bit C6 ~ C25 setting
对于带纠错的半速率AMR TRAU帧,C6~C25的设置方法和全速率AMR业务相同。For the half-rate AMR TRAU frame with error correction, the setting method of C6~C25 is the same as that of the full-rate AMR service.
3)数据比特设置3) Data bit setting
采用[7,3,4]艾布朗姆森码对22组(64个比特)重要比特编码共产生154个比特。这些比特填充在下表中D1~D154位置。[7,3,4] Abramson codes are used to encode 22 groups (64 bits) of important bits to generate 154 bits in total. These bits are filled in positions D1~D154 in the table below.
此外,还有87个比特(即151-64)未经上述纠错编码处理,将其填充在D155~D241位置。In addition, there are 87 bits (namely 151-64) that have not been subjected to the above error correction coding process, and are filled in positions D155-D241.
D242~D256位置没有使用,默认填1。如下表所示,就是扩展带宽后,经过纠错机制处理和未经纠错机制的所有待传输数据结构。D242~D256 positions are not used, fill in 1 by default. As shown in the table below, after the bandwidth is expanded, all data structures to be transmitted after being processed by the error correction mechanism and without the error correction mechanism.
对于其它速率的半速率AMR业务及HR业务,也可采用类似的方法进行填充。这些填充方法应用于不同速率的业务中,是没有超出本发明的保护范围的。For half-rate AMR services and HR services of other rates, a similar method can also be used for filling. The application of these filling methods to services of different rates does not exceed the scope of protection of the present invention.
当然,在实际运用中,还可以选择:循环码、汉明码等其他的编码方式。对于所属领域的技术人员而言,这种编码方式变化本身没有超出本发明的保护范围,是在本发明的方法启示下进行逻辑推导必然得出的,都构成了对本发明的实施例的等同替换;Of course, in practical applications, other encoding methods such as cyclic codes and Hamming codes can also be selected. For those skilled in the art, the change in the encoding method itself does not exceed the scope of protection of the present invention, and is necessarily obtained through logical derivation under the inspiration of the method of the present invention, and all constitute equivalent replacements for the embodiments of the present invention ;
S405、采用GSM系统类似全速率AMR业务的TRAU帧格式来传送上述经过纠错机制处理过的半速率语音数据和其他未经纠错机制处理过的半速率语音数据。此时的TRAU帧的结构分布如步骤403中的经过纠错机制处理后的表格所示,此TRAU帧共由320个比特组成,每20ms接收一帧。表中灰色部分为TRAU帧的同步比特,共有37个0、1组成。因同步比特数目比原来的增加了不少,减少了伪同步的可能性。即使其中的部分数据比特(非灰色部分)因传输误码发生改变,由于同步比特数据量大,部分数据比特的改变很难符合同步的要求,避免了伪同步现象的出现。基于同样的原因,在因传输误码或发生切换导致失步之后,根据37个同步比特,也能快速找到正确的同步头,恢复同步。S405. Transmit the half-rate voice data processed by the error correction mechanism and other half-rate voice data not processed by the error correction mechanism by adopting the TRAU frame format similar to the full-rate AMR service in the GSM system. The structure distribution of the TRAU frame at this time is shown in the table processed by the error correction mechanism in step 403. The TRAU frame consists of 320 bits in total, and one frame is received every 20 ms. The gray part in the table is the synchronization bit of the TRAU frame, which consists of 37 0s and 1s. Because the number of synchronization bits has increased a lot compared with the original one, the possibility of false synchronization is reduced. Even if some of the data bits (non-gray parts) are changed due to transmission errors, due to the large amount of synchronization bit data, it is difficult for the change of some data bits to meet the synchronization requirements, thereby avoiding the occurrence of false synchronization. For the same reason, after out-of-synchronization due to transmission error or handover, the correct synchronization head can be quickly found and synchronization can be restored according to the 37 synchronization bits.
利用本实施例,实现了对于需要提供高质量语音服务的站点,将经过所述站点的业务数据扩展传输带宽后,增加纠错机制处理重要数据,减少了伪同步的可能性,降低了语音损伤,提高了经由此站点传输的所有语音服务的质量。Utilizing this embodiment, it is realized that for a site that needs to provide high-quality voice services, after expanding the transmission bandwidth of the business data passing through the site, an error correction mechanism is added to process important data, reducing the possibility of false synchronization and voice damage , improving the quality of all voice services transmitted through this site.
针对如果有的用户对于语音质量有较高要求时,在BSC中增加用户抗误码开关配置,选择对特定的用户打开Abis接口抗误码功能,就得到了本发明提供的增强语音质量的方法的第三个实施例,包括步骤:For if some users have higher requirements for voice quality, increase the user anti-error switch configuration in the BSC, select to open the Abis interface anti-error function for specific users, and then obtain the method for enhancing voice quality provided by the present invention A third embodiment includes the steps of:
S501、当某用户对语音质量有较高要求时,修改BSC数据配置,对此用户打开用户Abis接口抗误码开关,并通过呼叫信令通知所述用户数据要经过的BTS;S501. When a certain user has higher requirements on voice quality, modify the BSC data configuration, and for this user, turn on the anti-error switch of the user Abis interface, and notify the BTS that the user data will pass through through call signaling;
S502、所述BTS接收到所述呼叫信令后同,通过TRAU帧带内信令通知其对应的TRAU Abis接口抗误码开关打开;TRAU默认抗误码开关关闭,BTS通过TRAU帧带内信令通知TRAU Abis接口抗误码开关是否打开,如果用户Abis接口抗误码开关打开,则通过TRAU帧带内信令通知TRAU Abis接口抗误码开关打开;S502. After the BTS receives the call signaling, it notifies its corresponding TRAU Abis interface anti-error switch to open through the TRAU frame in-band signaling; the TRAU default anti-error switch is closed, and the BTS passes the TRAU frame in-band signal Command to notify TRAU Abis interface whether the anti-error switch is turned on, if the user Abis interface anti-error switch is turned on, then notify TRAU Abis interface anti-error switch to turn on through TRAU frame in-band signaling;
S503、当用户抗误码开关打开时,则对于此用户采用半速率语音通话时。BTS和TRAU对此用户的传输数据进行抗误码处理,将BTS和TRAU之间的传输带宽由8Kbps扩展为16Kbps,采用类似全速率AMR业务的帧同步格式传送半速率下语音的数据,同时在TRAU和BTS之间的TRAU帧中增加纠错机制;TRAU和BTS之间的纠错机制与上一实施例中步骤403中完全一致,此不赘述;S503. When the anti-error switch of the user is turned on, a half-rate voice call is used for the user. BTS and TRAU perform anti-error processing on the transmission data of this user, expand the transmission bandwidth between BTS and TRAU from 8Kbps to 16Kbps, adopt the frame synchronization format similar to full-rate AMR service to transmit voice data at half-rate, and at the same time An error correction mechanism is added in the TRAU frame between the TRAU and the BTS; the error correction mechanism between the TRAU and the BTS is completely consistent with that in step 403 in the previous embodiment, and will not be repeated here;
S504、采用GSM系统类似全速率AMR业务的TRAU帧格式来传送上述经过纠错机制处理过的半速率语音数据和其他未经纠错机制处理过的半速率语音数据。此时的TRAU帧的结构分布如步骤403中的经过纠错机制处理后的表格所示,此TRAU帧共由320个比特组成,每20ms接收一帧。表中灰色部分为TRAU帧的同步比特,共有37个0、1组成。因同步比特数目比原来的增加了不少,减少了伪同步的可能性。即使其中的部分数据比特(非灰色部分)因传输误码发生改变,由于同步比特数据量大,部分数据比特的改变很难符合同步的要求,避免了伪同步现象的出现。基于同样的原因,在因传输误码或发生切换导致失步之后,根据37个同步比特,也能快速找到正确的同步头,恢复同步。S504. Transmit the half-rate voice data processed by the error correction mechanism and other half-rate voice data not processed by the error correction mechanism by adopting the TRAU frame format similar to the full-rate AMR service in the GSM system. The structure distribution of the TRAU frame at this time is shown in the table processed by the error correction mechanism in step 403. The TRAU frame consists of 320 bits in total, and one frame is received every 20 ms. The gray part in the table is the synchronization bit of the TRAU frame, which consists of 37 0s and 1s. Because the number of synchronization bits has increased a lot compared with the original one, the possibility of false synchronization is reduced. Even if some of the data bits (non-gray parts) are changed due to transmission errors, due to the large amount of synchronization bit data, it is difficult for the change of some data bits to meet the synchronization requirements, thereby avoiding the occurrence of false synchronization. For the same reason, after out-of-synchronization due to transmission error or handover, the correct synchronization head can be quickly found and synchronization can be restored according to the 37 synchronization bits.
在实际运用中,本实施例中的用户Abis接口抗误码开关与上一实施例中的站点Abis接口抗误码开关可以同时设置,也可以选择设置。当仅仅设置二者之一时,按照前两个实施例进行处理;当选择同时打开上述两个开关时,这两个功能同时处理,也就是:一方面,对于经由此站点的所有的相关业务数据,扩展传输带宽,并增加纠错机制处理;另一方面,对于此用户的所有相关业务数据经过的站点,扩展传输带宽,并增加纠错机制处理。这是运用本发明实施例的应有之义,没有超出本发明的保护范围。In practical application, the user Abis interface anti-error switch in this embodiment and the site Abis interface anti-error switch in the previous embodiment can be set at the same time, or can be set selectively. When only one of the two is set, it is processed according to the first two embodiments; when the two switches are selected to be turned on at the same time, these two functions are processed at the same time, that is: on the one hand, for all relevant business data via this site , expand the transmission bandwidth, and increase the error correction mechanism processing; on the other hand, for the stations through which all relevant business data of this user pass, expand the transmission bandwidth, and increase the error correction mechanism processing. This is the proper meaning of using the embodiments of the present invention, and does not exceed the protection scope of the present invention.
利用本实施例,实现了对于需要提供高质量语音服务的用户,对其在业务数据扩展传输带宽后,通过增加TRAU与BTS之间的传输校验机制,以及采用类似全速率TRAU帧同步格式传送半速率语音帧,有效解决传输误码、切换等因素导致的丢帧、错帧、伪同步帧对语音质量的损伤,提高半速率语音的通话质量。而且,根据实际运用的需要和传输条件的情况,通过增加站点Abis接口抗误码开关和用户抗误码开关,或者选择在Abis接口微波传输的站点,或者对语音质量有较高要求的高端用户,开启此功能,能够有效的实现系统容量和话音质量之间的均衡。Utilizing this embodiment, it is realized that for users who need to provide high-quality voice services, after expanding the transmission bandwidth of business data, by adding a transmission verification mechanism between TRAU and BTS, and using a similar full-rate TRAU frame synchronization format to transmit Half-rate voice frames can effectively solve the damage to voice quality caused by frame loss, wrong frames, and pseudo-synchronized frames caused by factors such as transmission errors and switching, and improve the call quality of half-rate voice. Moreover, according to actual application needs and transmission conditions, by adding the site Abis interface anti-error switch and user anti-error switch, or choose a site with microwave transmission at the Abis interface, or a high-end user who has high requirements for voice quality , enabling this function can effectively achieve the balance between system capacity and voice quality.
而且,通过上述二个实施例,克服了在半速率语音编解码情况下出现伪同步的概率很高的技术问题。这是由于半速率语音编解码中同步比特非常少,在传输误码或者切换情况下,出现伪同步的概率很高。而半速率语音编解码压缩比高,如果出现错帧、丢帧、伪同步帧,对语音质量的破坏程度非常高。现有技术中的半速率TRAU帧抗传输误码能力差,导致有误码或者切换发生时,语音质量下降明显。据测算,一个伪同步帧将对语音质量MOS分(MeanOpinion Score,平均意见得分)产生0.05~0.3分的影响。(MOS分的满分为5分,得分越高,语音质量约好)。而在上述两个实施例中,由于同步比特数据量大,出现伪同步的概率大大降低,很好地保障了语音质量。Moreover, through the above two embodiments, the technical problem of high probability of pseudo-synchronization in the case of half-rate speech codec is overcome. This is because there are very few synchronization bits in the half-rate speech codec, and there is a high probability of false synchronization in the case of transmission errors or switching. However, the half-rate speech codec has a high compression ratio. If there are frame errors, frame loss, or pseudo-synchronous frames, the damage to the speech quality will be very high. The half-rate TRAU frame in the prior art has poor ability to resist transmission errors, resulting in a significant drop in voice quality when there is a bit error or handover occurs. According to calculations, a pseudo-synchronous frame will have an impact of 0.05 to 0.3 points on the voice quality MOS score (Mean Opinion Score, average opinion score). (The full score of MOS is 5 points, the higher the score, the better the voice quality). However, in the above two embodiments, due to the large amount of synchronization bit data, the probability of false synchronization is greatly reduced, and the voice quality is well guaranteed.
如图5所示,是本发明的增强语音质量的基站的第一个实施例,包括:As shown in Figure 5, it is the first embodiment of the base station for enhancing voice quality of the present invention, including:
传输带宽扩展单元501,用于:扩展BTS与BSC之间的传输带宽;A transmission bandwidth extension unit 501, configured to: extend the transmission bandwidth between the BTS and the BSC;
TRAU传输纠错单元502,用于:在TRAU和所述BTS之间的TRAU帧中增加纠错机制。The TRAU transmission error correction unit 502 is configured to: add an error correction mechanism in the TRAU frame between the TRAU and the BTS.
通过本发明的上述实施例,通过扩展BTS与BSC之间的传输带宽,在TRAU和BTS之间的TRAU帧中增加纠错机制,能够降低数据传输的误码率,提升语音通信质量。Through the above-mentioned embodiments of the present invention, by expanding the transmission bandwidth between BTS and BSC and adding an error correction mechanism in the TRAU frame between TRAU and BTS, the bit error rate of data transmission can be reduced and the quality of voice communication can be improved.
其中,在上述实施例中,所述BTS与所述BSC之间的数据可以是半速率编码格式。Wherein, in the above embodiment, the data between the BTS and the BSC may be in a half-rate encoding format.
其中,在上述实施例中,为了通过开关来指示传输带宽扩展单元和纠错机制编码单元工作,还可以包括:Wherein, in the above embodiment, in order to instruct the transmission bandwidth extension unit and the error correction mechanism encoding unit to work through the switch, it may also include:
抗误码开关,所述抗误码开关为站点抗误码开关或用户抗误码开关,用于:打开或关闭预设的抗误码开关并发送打开信号,所述打开信号用于指示所述传输带宽扩展单元和所述TRAU传输纠错单元工作。An anti-error switch, the anti-error switch is a site anti-error switch or a user anti-error switch, used to: turn on or off a preset anti-error switch and send an open signal, and the open signal is used to indicate the The transmission bandwidth extension unit and the TRAU transmission error correction unit work.
其中,在上述实施例中,在BTS与BSC之间采用微波传输或其他原因导致的传输误码率较高时,为了提高传输质量,还可以包括:Wherein, in the above embodiment, when microwave transmission is used between the BTS and the BSC or the transmission bit error rate is high due to other reasons, in order to improve the transmission quality, it may also include:
传输误码检查单元,用于:如果所述BTS与所述BSC之间采用微波传输,或所述BTS与所述BSC之间存在传输误码,则生成启动指令,所述启动指令用于指示所述传输带宽扩展单元和所述TRAU传输纠错单元工作。a transmission error checking unit, configured to generate a start instruction if microwave transmission is used between the BTS and the BSC, or a transmission error exists between the BTS and the BSC, and the start instruction is used to indicate The transmission bandwidth extension unit and the TRAU transmission error correction unit work.
其中,在上述实施例中,如果打开的所述抗误码开关为所述站点抗误码开关,则Wherein, in the above-mentioned embodiment, if the anti-error switch that is turned on is the site anti-error switch, then
所述传输带宽扩展单元将所述站点的所述BTS与所述BSC之间的传输带宽由8Kbps扩展为16Kbps;The transmission bandwidth extension unit expands the transmission bandwidth between the BTS of the site and the BSC from 8Kbps to 16Kbps;
所述TRAU传输纠错单元在所述TRAU和所述BTS之间的TRAU帧中增加纠错机制。The TRAU transmission error correction unit adds an error correction mechanism in the TRAU frame between the TRAU and the BTS.
其中,在上述实施例中,如果打开的所述抗误码开关为所述用户抗误码开关,则Wherein, in the above embodiment, if the anti-error switch that is turned on is the user anti-error switch, then
所述传输带宽扩展单元将所述用户数据经过的所有站点的所述BTS与所述BSC之间的传输带宽由8Kbps扩展为16Kbps;The transmission bandwidth expansion unit expands the transmission bandwidth between the BTS and the BSC of all stations through which the user data passes from 8Kbps to 16Kbps;
所述TRAU传输纠错单元在所述用户数据经过的所有站点的所述TRAU和所述BTS之间的TRAU帧中增加纠错机制。The TRAU transmission error correction unit adds an error correction mechanism in TRAU frames between the TRAU and the BTS of all stations through which the user data passes.
针对现有的半速率传输方式,在站点抗误码开关或用户抗误码开关打开时,将传输带宽扩展为16Kbps,并相应增加纠错机制,有效地降低了因传输误码导致的语音损伤。For the existing half-rate transmission mode, when the site anti-error switch or user anti-error switch is turned on, the transmission bandwidth is expanded to 16Kbps, and an error correction mechanism is added accordingly, which effectively reduces the voice damage caused by transmission errors .
其中,在上述实施例中,所述TRAU传输纠错单元具体为:Wherein, in the above embodiment, the TRAU transmission error correction unit is specifically:
纠错编解码子单元,用于:对业务数据进行分组纠错编解码。The error correction codec subunit is used for: performing packet error correction codec on the service data.
其中,在上述实施例中,所述业务数据具体为:Wherein, in the above-mentioned embodiment, the business data is specifically:
控制比特、CRC校验比特或语音参数。Control bits, CRC check bits or speech parameters.
其中,在上述实施例中,所述分组纠错编解码采用的是:Wherein, in the above embodiment, the packet error correction codec adopts:
艾布朗森码、循环码或汉明码。Abronson, cyclic, or Hamming codes.
利用艾布朗森码或其他编解码方法对重要的业务数据,如控制比特、CRC校验比特或重要的语音参数进行纠错编码后传输,可以降低伪同步的概率,减少传输误码导致的语音损伤,提升语音质量。Use Abronson code or other encoding and decoding methods to transmit important business data, such as control bits, CRC check bits or important voice parameters, after error correction coding, which can reduce the probability of false synchronization and reduce the voice loss caused by transmission errors. Impairment, improve voice quality.
如图6所示,是本发明中利用BTS602和TRAU601对语音参数纠错编解码过程示意图。上述过程也适用于控制比特、CRC校验比特及其他需要进行纠错编解码的业务数据。站点抗误码开关和用户抗误码开关在BTS侧,当然在有的实施例中,可以包括述两种抗误码开关中的一种。As shown in FIG. 6 , it is a schematic diagram of the error correction encoding and decoding process of speech parameters by using BTS602 and TRAU601 in the present invention. The above process is also applicable to control bits, CRC check bits and other service data that need to be encoded and decoded for error correction. The station anti-error switch and the user anti-error switch are on the BTS side. Of course, in some embodiments, one of the above two anti-error switches may be included.
本发明的方法和装置的应用范围很广,对于固定电话以及其他有关数据传输等领域,也可以采用有选择的拓展传输带宽,然后增加纠错机制的方法,来保障对于重要信息的正确传输,提升数据传输业务的质量。也可以应用于全速率语音编码等其他方式,具体的技术效果取决于技术方案所具体应用的环境。对于所属领域的技术人员而言,这些变化没有超出本发明的保护范围。The method and device of the present invention have a wide range of applications. For fixed telephones and other related data transmission fields, the method of selectively expanding the transmission bandwidth and adding an error correction mechanism can also be used to ensure the correct transmission of important information. Improve the quality of data transmission services. It can also be applied to other methods such as full-rate speech coding, and the specific technical effect depends on the specific application environment of the technical solution. For those skilled in the art, these changes are within the protection scope of the present invention.
通过以上的实施方式的描述,所属领域的技术人员可以清楚地了解到本发明可借助软件加必需的通用硬件平台的方式来实现,当然也可以通过硬件,但很多情况下前者是更佳的实施方式。基于这样的理解,本发明的技术方案本质上或者说对现有技术做出贡献的部分可以以软件产品的形式体现出来,该计算机软件产品存储在一个存储介质中,包括若干指令用以使得一台计算机设备(可以是个人计算机,服务器,或者网络设备等)执行本发明各个实施例所述的方法。Through the description of the above embodiments, those skilled in the art can clearly understand that the present invention can be implemented by means of software plus a necessary general-purpose hardware platform, and of course also by hardware, but in many cases the former is a better implementation Way. Based on this understanding, the essence of the technical solution of the present invention or the part that contributes to the prior art can be embodied in the form of a software product. The computer software product is stored in a storage medium and includes several instructions to make a A computer device (which may be a personal computer, a server, or a network device, etc.) executes the methods described in various embodiments of the present invention.
以上所述的本发明实施方式,并不构成对本发明保护范围的限定。任何在本发明的精神和原则之内所作的修改、等同替换和改进等,均应包含在本发明的保护范围之内。The embodiments of the present invention described above are not intended to limit the protection scope of the present invention. Any modifications, equivalent replacements and improvements made within the spirit and principles of the present invention shall be included within the protection scope of the present invention.
| Application Number | Priority Date | Filing Date | Title |
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| CN2007100477491ACN101159442B (en) | 2007-10-30 | 2007-10-30 | Method and base station of enhancing voice quality |
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| CN2007100477491ACN101159442B (en) | 2007-10-30 | 2007-10-30 | Method and base station of enhancing voice quality |
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