WO2007118392A1 - A method and device for transmitting voice data - Google Patents

A method and device for transmitting voice data Download PDF

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Publication number
WO2007118392A1
WO2007118392A1 PCT/CN2007/000283 CN2007000283W WO2007118392A1 WO 2007118392 A1 WO2007118392 A1 WO 2007118392A1 CN 2007000283 W CN2007000283 W CN 2007000283W WO 2007118392 A1 WO2007118392 A1 WO 2007118392A1
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Prior art keywords
voice data
data frame
processing module
received
deletion
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PCT/CN2007/000283
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French (fr)
Chinese (zh)
Inventor
Shoubo Xie
Jie Yao
Tao Yu
Dong Zhang
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Huawei Technologies Co., Ltd.
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Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2007118392A1 publication Critical patent/WO2007118392A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W28/00Network traffic management; Network resource management
    • H04W28/02Traffic management, e.g. flow control or congestion control
    • H04W28/06Optimizing the usage of the radio link, e.g. header compression, information sizing, discarding information

Definitions

  • the present invention relates to the field of mobile communication technologies, and in particular, to a method and apparatus for transmitting voice data. Background of the invention
  • a call is made between user terminals in a code division multiple access (CDMA) mobile communication system, and the voice of the call is encoded by a certain speech coding algorithm to obtain a voice data frame, and the obtained voice data frame passes the voice between the network devices.
  • the interface is transmitted.
  • a voice data frame includes a frame header and a voice payload.
  • the frame header is frame format information defined for realizing transmission of a voice data frame in a physical layer.
  • the voice payload is voice data formed by encoding a voice.
  • CDMA2000 Different CDMA systems may use different speech coding algorithms.
  • the following three variable rate speech codec algorithms are mainly used in CDMA2000:
  • Enhanced Variable Rate Codec (EVRC), Qualcomm Code Excited Linear Prediction 13K (QCELP13K) and High-Through Code Excited Linear Predictive 8K Speech Angle Code coder ( Qualcomm Code Excited Linear Prediction 8K , QCELP8K ).
  • Table 1 Algorithm 4 ⁇ rate Half rate 1/4 rate 1/8 rate
  • the embodiment of the invention provides a method for transmitting voice data, which can improve the transmission efficiency of voice data.
  • the method comprises the following steps:
  • the deleted voice data stream is transmitted through the voice interface.
  • the embodiment of the invention further provides an apparatus for transmitting voice data, which can improve the transmission efficiency of voice data.
  • the device includes the following modules:
  • a rate identifying unit configured to receive a voice data frame and send the received voice data frame to a deletion processing unit; determine whether the received voice data frame is an inactive voice data frame, generate a deletion indication signal according to the result of the determination, and The generated deletion indication signal is sent to the deletion processing unit;
  • a deletion processing unit configured to save a preset deletion processing policy; perform a deletion operation on the currently received voice data frame according to the received deletion indication signal and the deletion processing policy, or output the received voice data frame.
  • the present invention also discloses a base station or a base station controller for transmitting voice data.
  • the base station or the base station controller includes, in addition to the channel processing module and the interface processing module, a compression processing module for identifying the voice from the channel processing module.
  • the inactive voice data frame in the data frame is deleted according to a preset policy, and some or all of the inactive voice data frames are deleted, and the deleted voice data frame is sent to the interface processing module.
  • the inactive voice data frame is identified and deleted according to a certain policy, and the voice interface can transmit the inactive voice data under the condition that the voice quality of the call is hardly affected.
  • the number of frames is reduced, improving the ability of the voice interface to carry voice data.
  • FIG. 1 is a schematic structural view of a prior art CDMA system
  • FIG. 2 is a schematic diagram showing a frame structure of a voice data frame of an Abis interface
  • FIG. 3 is a schematic diagram of a system according to an embodiment of the present invention.
  • FIG. 4 is a schematic diagram of an apparatus according to an embodiment of the present invention
  • FIG. 5 is a flowchart of processing according to an embodiment of the present invention. Mode for carrying out the invention
  • the transmitted voice data frame is identified, the inactive voice data frame is identified, some or all of the inactive voice data frames are deleted according to a certain policy, and the voice data frame is sent to The transmission channel of the voice interface.
  • the solution of the invention further comprises: detecting the silence in the received voice data stream at the receiving end of the voice data frame, and compensating the inactive voice data frame at the muted position.
  • the structure of the CDMA system is shown in Figure 1.
  • the signal sent by the terminal (MS) passes through the base transceiver station (BTS) and the base station controller (BSC)/packet control function node.
  • BTS base transceiver station
  • BSC base station controller
  • PCF Packet Data Serving Node
  • MSC Packet Switching Center
  • the voice interface includes an Abis interface between the BTS and the BSC and an A interface (including an A1 interface, an A2 interface, and an A5 interface) between the BTS and the MSC. Therefore, it is necessary to improve the transmission efficiency of the above interface as much as possible under the condition that the transmission equipment hardware of the Abis interface and the A interface are fixed.
  • the voice interface in this embodiment is an Abis interface between the BTS and the BSC.
  • the protocol stack transmitted by the Abis interface is shown in Figure 2.
  • the voice payload part of the voice data frame is encoded by some kind.
  • the method for processing voice data, the encoding method may be one of the foregoing EVRC, QCELP13K or QCELP8K;
  • the frame header portion includes an Abis frame header and a transmission bearer frame header, wherein the Abis frame header is used to ensure voice data in the BTS and the BSC
  • the Abis interface is reliably transmitted, and the transmission bearer header is used to ensure reliable transmission of voice data in the physical channel.
  • the Abis interface of the CDMA system is not standardized, the content of the Abis frame header defined by each manufacturer is very different.
  • the length of the Abis frame header is 4 to 8 bits; the transmission bearer header is also different according to the physical channel of the bearer, and there is an asynchronous transmission mode. (ATM), IP, Advanced Data Link Control (HDLC) or custom format, etc., typically 4 to 10 bits in length.
  • the transmitted voice data frame is identified, the inactive voice data frame is identified, and some or all of the inactive voice data is deleted according to a certain policy.
  • the frame is then sent to the transmission channel of the voice interface, so that the bandwidth of the voice interface can be saved.
  • the rate of each voice data frame can be determined by the physical layer rate determination, so as to determine whether the rate frame is an activated voice data frame or an inactive voice data frame; and the Abis frame header includes a frame number of the voice data frame, according to the frame number.
  • inactive voice data frame processing strategies such as deleting all inactive voice data frames, or deleting only inactive voice data frames with even frame numbers.
  • the BTS and the BSC respectively access the Abis interface through the respective interface processing modules, and respectively add a compression processing module between the original channel processing module and the interface processing module of the BTS and the BSC, where
  • the compression processing module of the BTS device is configured to identify the inactive voice data frame in the service voice data frame, and delete some or all of the inactive voice data frames according to a preset policy; and also to identify the silence in the downlink voice data frame.
  • a non-activated speech data frame is generated and filled in the identified mute position.
  • the compression processing module of the BSC device is used to analyze the Abis frame header in the downlink voice data frame. Feature information, identifying inactive voice data frames, and deleting some or all of the inactive voice data frames according to a pre-specified policy; also for identifying silence in the uplink voice data frame, generating inactive voice data frames and filling in the identified The mute position.
  • a rate identifying unit 401 configured to receive a voice data frame from a channel processing module of the BTS device, and perform physical layer rate determination on the voice data frame, if the rate of the voice data frame meets a non- Activating the determination condition of the speech data frame rate, generating an allow deletion indication signal, and transmitting the generated permission deletion indication signal and the speech data frame to the deletion processing unit 402, otherwise transmitting the speech data frame separately to the deletion processing unit 402.
  • the encoding algorithm used is QCELP13K.
  • the determination condition is that the rate of the voice data frame is 1/8 of the full rate, that is, the air interface rate is 1800 bps or the 20 ms packet length is 20 bits.
  • the deletion processing unit 402 is configured to save a preset deletion processing policy.
  • it is determined whether the permission deletion indication signal is received, and if received, determining whether to delete according to the deletion processing policy. If the voice data frame is deleted, the voice data frame is deleted, and a silence is formed in the time period occupied by the voice data frame; if the permission deletion instruction signal is not received or the voice data frame is not deleted according to the deletion processing policy, The received voice data frame is sent to the interface processing module.
  • the deletion processing policy may be: deleting all the identified inactive voice data frames; or determining whether the frame number of the identified inactive voice data frame satisfies certain conditions, such as whether it is an even number, and deleting the inactive condition that meets the condition. Voice data frame.
  • the deletion processing policy is not limited to the above example, and may be set according to actual needs.
  • the solution of the invention further comprises: at the receiving end of the voice data frame, detecting silence in the data stream composed of the received voice data frame, and compensating the inactive voice data frame in the muted position.
  • the compression processing module also includes:
  • a silence time determining unit 403 configured to receive a voice data frame from the interface processing module and send the received voice data frame to the recovery unit 404; determine whether silence is present in the voice data frame, and if yes, to the recovery unit 404 sends a resume indication signal; the method of determining the silence is: setting a timer for performing silence time determination, the duration of the timer is the same as the voice data frame transmission interval of the Abis interface, for example, 20 milliseconds; When a voice data frame is reached, a reset operation is performed on the timer and the timing is restarted. If the timer expires, it indicates that no voice data frame is received between the current start timing of the timer and the timeout period, that is, a clock appears. The silence interval caused by the 1/8 frame being deleted.
  • the recovery unit 404 is configured to send the received voice data frame to the channel processing module.
  • a 1/8 frame is generated and sent to the channel processing module.
  • the generating 1/8 frame is: generating 1/8 frame of the voice payload coded as FF, and the frame number is obtained by the mute position of the previous voice data frame number modulo 16 plus 1. Since the frame number is 0 to 15 cycles, if the frame number of the previous frame is 15, the frame number of the filled 1/8 frame is 0.
  • the generated 1/8 frame voice payload code can also be a code for other comfort noise.
  • the composition and connection relationship of the compression processing module of the BSC device can be obtained, and therefore will not be described again.
  • the above compression processing device may be an independent device or integrated as a functional module in a base station or a base station controller in a base station or a base station controller.
  • the processing flow of the uplink voice data frame by using the foregoing apparatus is as shown in FIG. 5, and includes the following steps:
  • Step 501 Determine a physical layer rate of the uplink voice data frame output by the BTS channel processing module, and identify an inactive voice data frame.
  • the inactive voice data frame is a 1/8 rate voice data frame.
  • Step 502 Delete some or all of the inactive voice data frames according to the pre-defined policy, and send the deleted uplink voice data stream to the transport channel of the Abis interface through the interface processing module of the BTS.
  • Step 503 The interface processing module of the BSC receives the uplink voice data stream of the Abis interface, and identifies the silence in the received data stream;
  • Step 504 Generate 1/8 frames and fill in the identified mute position, and then send the uplink voice data frame to the subsequent processing module of the BSC.
  • the processing flow of the downlink voice data frame can be obtained by referring to the above process, and therefore will not be described again.
  • the speech effect processed by the embodiment of the present invention is evaluated by using the P.862 method, and the encoding algorithm of the speech data is EVRC.
  • the average opinion score (MOS) of the voice data is 3.739 points, and the score of the voice data after the prior art processing is 3.798 points.
  • the processing of the present invention causes the MOS to drop by only 0.059 points, and it can be considered that the voice quality has not decreased. Therefore, the embodiment of the present invention reduces the waste of bandwidth resources of the Abis interface and greatly improves the voice bearer efficiency of the Abis interface under the premise that the call quality is not substantially affected. If the E1 interface is used in the Abis interface, the number of voice channels carried by each E1 interface can be increased from the normal 180 to 190 voice channels to more than 240 voice channels.
  • the solution of the present invention can also be applied to other voice interfaces in a CDMA system, for example, to an A interface between a BSC and an MSC, including interfaces such as Al, A2, and A5.
  • the inventive solution can also be widely applied to other wireless communication systems, such as CDMA2000 systems.

Abstract

A method for transmitting voice data includes the steps of: identifying the inactivated voice data frame in voice data stream; according to the preset deletion strategy, deleting some or all of the identified inactivated voice data frames; after the deleting process, transmitting the voice data stream through the voice interface. In addition, a device for transmitting voice data is also disclosed. By means of the method of the invention, the amount of the inactivated voice data frame transmitted via voice interface could be reduced, and the capability of voice interface for bearing voice data could be improved, with almost no effect on the voice quality of the communication.

Description

一种传输语音数据的方法和装置  Method and device for transmitting voice data
技术领域 Technical field
本发明涉及移动通信技术领域, 特别涉及一种传输语音数据的方法 和装置。 发明背景  The present invention relates to the field of mobile communication technologies, and in particular, to a method and apparatus for transmitting voice data. Background of the invention
码分多址复用( CDMA )移动通信系统中的用户终端之间进行通话, 通话的语音要通过一定的语音编码算法进行编码得到语音数据帧, 所得 语音数据帧通过各个网络设备之间的语音接口进行传输。 一个语音数据 帧包括帧头和语音净荷两部分, 其中帧头是为了实现语音数据帧在物理 层中传输而定义的帧格式信息, 语音净荷是对语音进行编码后形成的语 音数据。  A call is made between user terminals in a code division multiple access (CDMA) mobile communication system, and the voice of the call is encoded by a certain speech coding algorithm to obtain a voice data frame, and the obtained voice data frame passes the voice between the network devices. The interface is transmitted. A voice data frame includes a frame header and a voice payload. The frame header is frame format information defined for realizing transmission of a voice data frame in a physical layer. The voice payload is voice data formed by encoding a voice.
不同的 CDMA 系统可能使用不同的语音编码算法。 例如, 在 CDMA2000中主要使用以下三种可变速率的语音编解码算法:  Different CDMA systems may use different speech coding algorithms. For example, the following three variable rate speech codec algorithms are mainly used in CDMA2000:
增强型变速率语音编解码器 ( Enhanced Variable Rate Codec , EVRC )、 高通码激励线性预测 13K语音编解码器 (Qualcomm Code Excited Linear Prediction 13K, QCELP13K )和高通码激励线性预测 8K 语音编角竽码器 ( Qualcomm Code Excited Linear Prediction 8K , QCELP8K )。 不同编解码算法在各种速率下的语音数据帧长度和空口速 率如表 1所示: 算法 4^率 半速率 1/4速率 1/8速率 Enhanced Variable Rate Codec (EVRC), Qualcomm Code Excited Linear Prediction 13K (QCELP13K) and High-Through Code Excited Linear Predictive 8K Speech Angle Code coder ( Qualcomm Code Excited Linear Prediction 8K , QCELP8K ). The speech data frame length and air interface rate of different codecs at various rates are shown in Table 1: Algorithm 4^ rate Half rate 1/4 rate 1/8 rate
EVRC 空口速率 ( bps ) 9600 4800 无 1200 EVRC air interface rate ( bps ) 9600 4800 none 1200
20ms包长 ( bit ) 171 80 无 16 QCELP 8K 空口速率(bps ) 9600 4800 2400 160020ms packet length (bit) 171 80 no 16 QCELP 8K air interface rate (bps) 9600 4800 2400 1600
20ms包长 ( bit ) 171 80 40 1620ms packet length ( bit ) 171 80 40 16
QCELP 空口速率(bps ) 14400 7200 3600 1800 13K 20ms包长 ( bit ) 266 124 54 20 表 1 由于通话中总是一方在说, 另一方在听, 并且说话语音中有部分空隙, 因此可以把通话过程中传输的声音分为有效语音和背景噪声两部分。 在 上述编码算法中,全速率的语音净荷为对有效语音的编码,半速率和 1/4 速率的语音净荷为对有效语音和背景噪声过渡带的编码, 上述这些速率 的语音数据帧又被称为激活语音数据帧; 1/8 帧为对背景噪声的编码, 又被称为非激活语音数据帧。 QCELP air interface rate (bps) 14400 7200 3600 1800 13K 20ms packet length (bit) 266 124 54 20 Table 1 Since the party is always talking, the other party is listening, and there is a gap in the voice, so the call can be made. The sound transmitted in is divided into two parts: effective speech and background noise. In the above encoding algorithm, the full rate speech payload is the encoding of the effective speech, and the half rate and 1/4 rate speech payload are the encoding of the effective speech and background noise transition bands, and the speech data frames of the above rates are It is called an active speech data frame; 1/8 frame is the encoding of background noise, also known as inactive speech data frame.
在现有技术中, 语音数据帧在语音接口中传输时, 非激活语音数据 帧是和激活语音数据帧一起被传输的。 由于非激活语音数据帧实际上并 不传输有效语音, 因此非激活语音数据帧所占用的这一部分传输带宽, 实际是对语音接口宝贵带宽资源的浪费。 发明内容  In the prior art, when a voice data frame is transmitted in a voice interface, the inactive voice data frame is transmitted along with the activated voice data frame. Since the inactive voice data frame does not actually transmit valid voice, the portion of the transmission bandwidth occupied by the inactive voice data frame is actually a waste of valuable bandwidth resources of the voice interface. Summary of the invention
本发明实施例提出一种传输语音数据的方法, 能够提高语音数据的 传输效率。  The embodiment of the invention provides a method for transmitting voice data, which can improve the transmission efficiency of voice data.
该方法包括如下步驟:  The method comprises the following steps:
识别出语音数据流中的非激活语音数据帧;  Identifying inactive voice data frames in the voice data stream;
根据预先设置的删除策略, 删除部分或全部被识别出的非激活语音 数据帧;  Deleting some or all of the identified inactive voice data frames according to a preset deletion policy;
通过语音接口传输删除处理后的语音数据流。 本发明实施例还提出一种传输语音数据的装置, 能够提高语音数据 的传输效率。 The deleted voice data stream is transmitted through the voice interface. The embodiment of the invention further provides an apparatus for transmitting voice data, which can improve the transmission efficiency of voice data.
该装置包括如下模块:  The device includes the following modules:
速率识别单元, 用于接收语音数据帧并将所接收的语音数据帧发送 到删除处理单元; 判断所接收的语音数据帧是否是非激活语音数据帧, 根据判断的结果生成删除指示信号, 并将所生成的删除指示信号发送到 删除处理单元;  a rate identifying unit, configured to receive a voice data frame and send the received voice data frame to a deletion processing unit; determine whether the received voice data frame is an inactive voice data frame, generate a deletion indication signal according to the result of the determination, and The generated deletion indication signal is sent to the deletion processing unit;
删除处理单元, 用于保存预先设置的删除处理策略; 根据所接收的 删除指示信号和删除处理策略对当前接收的语音数据帧进行删除操作, 或者输出所接收的语音数据帧。  And a deletion processing unit, configured to save a preset deletion processing policy; perform a deletion operation on the currently received voice data frame according to the received deletion indication signal and the deletion processing policy, or output the received voice data frame.
本发明还公开了传输语音数据的基站或基站控制器, 所述基站或基 站控制器除了包括信道处理模块和接口处理模块之外, 还包括压缩处理 模块, 用于识别出来自信道处理模块的语音数据帧中的非激活语音数据 帧, 并按照预先设置的策略删除部分或全部非激活语音数据帧, 并将删 除处理后的语音数据帧发送至接口处理模块。  The present invention also discloses a base station or a base station controller for transmitting voice data. The base station or the base station controller includes, in addition to the channel processing module and the interface processing module, a compression processing module for identifying the voice from the channel processing module. The inactive voice data frame in the data frame is deleted according to a preset policy, and some or all of the inactive voice data frames are deleted, and the deleted voice data frame is sent to the interface processing module.
从以上技术方案可以看出, 在语音数据帧的发送端, 识别并按照一 定的策略删除非激活语音数据帧, 可以在通话语音质量几乎不受影响的 条件下,使得语音接口传输非激活语音数据帧的数目减少,提高语音接口 承载语音数据的能力。 附图简要说明  It can be seen from the above technical solution that, at the transmitting end of the voice data frame, the inactive voice data frame is identified and deleted according to a certain policy, and the voice interface can transmit the inactive voice data under the condition that the voice quality of the call is hardly affected. The number of frames is reduced, improving the ability of the voice interface to carry voice data. BRIEF DESCRIPTION OF THE DRAWINGS
图 1为现有技术 CDMA系统的结构示意图;  1 is a schematic structural view of a prior art CDMA system;
图 2为 Abis接口语音数据帧的帧结构示意图;  2 is a schematic diagram showing a frame structure of a voice data frame of an Abis interface;
图 3为本发明实施例系统示意图;  3 is a schematic diagram of a system according to an embodiment of the present invention;
图 4为本发明实施例装置示意图; 图 5为本发明实施例处理流程图。 实施本发明的方式 4 is a schematic diagram of an apparatus according to an embodiment of the present invention; FIG. 5 is a flowchart of processing according to an embodiment of the present invention. Mode for carrying out the invention
本发明实施例包括以下基本内容:  The embodiment of the invention includes the following basic contents:
在语音数据帧的发送端 , 对所发送的语音数据帧进行识别, 识别出 其中的非激活语音数据帧, 按照一定的策略删除部分或全部的非激活语 音数据帧, 再将语音数据帧发送至语音接口的传输信道。 为了保证语音 数据帧所组成的语音数据流是连续的, 本发明方案还包括在语音数据帧 的接收端, 检测出所接收的语音数据流中的静音, 在静音的位置补偿非 激活语音数据帧。  At the transmitting end of the voice data frame, the transmitted voice data frame is identified, the inactive voice data frame is identified, some or all of the inactive voice data frames are deleted according to a certain policy, and the voice data frame is sent to The transmission channel of the voice interface. In order to ensure that the voice data stream composed of the voice data frames is continuous, the solution of the invention further comprises: detecting the silence in the received voice data stream at the receiving end of the voice data frame, and compensating the inactive voice data frame at the muted position.
为使本发明的目的、技术方案和优点更加清楚, 下面以 CDMA系统 为例, 对本发明内容作进一步的详细阐述。  In order to make the objects, technical solutions and advantages of the present invention more clear, the content of the present invention will be further described in detail below by taking a CDMA system as an example.
CDMA系统结构如图 1所示, 在上行方向, 终端 (MS )发出的信 号通过基站收发信机(BTS )和基站控制器(BSC ) /分组控制功能节点 The structure of the CDMA system is shown in Figure 1. In the uplink direction, the signal sent by the terminal (MS) passes through the base transceiver station (BTS) and the base station controller (BSC)/packet control function node.
( PCF ), 连接分组数据服务节点(PDSN )或分组交换中心(MSC ), 再 由 PDSN 或 MSC 转发至互联网络 (Internet )、 公用陆地移动网络(PCF), connected to a Packet Data Serving Node (PDSN) or Packet Switching Center (MSC), which is then forwarded by the PDSN or MSC to the Internet (Internet), public land mobile network
( PLMN )、 公用电话交换网 (PSTN )或综合业务数字网 (ISDN ); 在 下行方向信号则按与上述相反的方向发送。 对于整个 CDMA系统而言, 希望能尽可能地提高语音接口的传输带宽, 即接口所能承载语音数据的 容量。所述语音接口包括 BTS和 BSC之间的 Abis接口以及 BTS与 MSC 之间的 A接口 (包括 A1接口、 A2接口和 A5接口)。 这就需要在 载 Abis接口以及 A接口的传输设备硬件条件一定的情况下,尽可能地提高 上述接口的传输效率。 (PLMN), Public Switched Telephone Network (PSTN) or Integrated Services Digital Network (ISDN); signals in the downstream direction are transmitted in the opposite direction to the above. For the entire CDMA system, it is desirable to increase the transmission bandwidth of the voice interface as much as possible, that is, the capacity of the interface to carry voice data. The voice interface includes an Abis interface between the BTS and the BSC and an A interface (including an A1 interface, an A2 interface, and an A5 interface) between the BTS and the MSC. Therefore, it is necessary to improve the transmission efficiency of the above interface as much as possible under the condition that the transmission equipment hardware of the Abis interface and the A interface are fixed.
本实施例所述语音接口为 BTS和 BSC之间的 Abis接口。 Abis接口 传输的协议栈如图 2所示, 语音数据帧的语音净荷部分是通过某种编码 方法处理后的语音数据, 所述编码方法可以是上述 EVRC、 QCELP13K 或 QCELP8K中的一种; 帧头部分包括 Abis帧头和传输承载帧头, 其中 Abis帧头用于保证语音数据在 BTS和 BSC之间的 Abis接口可靠传输, 传输承载帧头用于保证语音数据在物理信道中可靠传输。 The voice interface in this embodiment is an Abis interface between the BTS and the BSC. The protocol stack transmitted by the Abis interface is shown in Figure 2. The voice payload part of the voice data frame is encoded by some kind. The method for processing voice data, the encoding method may be one of the foregoing EVRC, QCELP13K or QCELP8K; the frame header portion includes an Abis frame header and a transmission bearer frame header, wherein the Abis frame header is used to ensure voice data in the BTS and the BSC The Abis interface is reliably transmitted, and the transmission bearer header is used to ensure reliable transmission of voice data in the physical channel.
由于 CDMA系统的 Abis接口没有标准化,各个厂家定义的 Abis帧 头内容有很大不同, 通常 Abis帧头的长度为 4 ~ 8比特; 传输承载帧头 也根据承载的物理信道不同, 有异步传输模式 (ATM )、 IP、 高级数据 链路控制 ( HDLC )或自定义的格式等等, 长度一般为 4 ~ 10比特。  Since the Abis interface of the CDMA system is not standardized, the content of the Abis frame header defined by each manufacturer is very different. Generally, the length of the Abis frame header is 4 to 8 bits; the transmission bearer header is also different according to the physical channel of the bearer, and there is an asynchronous transmission mode. (ATM), IP, Advanced Data Link Control (HDLC) or custom format, etc., typically 4 to 10 bits in length.
为了提高接口的传输效率, 在语音数据帧的发送端设备上, 对所发 送的语音数据帧进行识别, 识别出其中的非激活语音数据帧, 按照一定 的策略删除部分或全部的非激活语音数据帧, 再将语音数据帧发送至语 音接口的传输信道, 这样就可以实现节省语音接口带宽的目的。  In order to improve the transmission efficiency of the interface, on the transmitting device of the voice data frame, the transmitted voice data frame is identified, the inactive voice data frame is identified, and some or all of the inactive voice data is deleted according to a certain policy. The frame is then sent to the transmission channel of the voice interface, so that the bandwidth of the voice interface can be saved.
可以通过物理层速率判定得到各个语音数据帧的速率, 从而判断出 该速率帧是激活语音数据帧还是非激活语音数据帧;并且 Abis帧头中包 含了语音数据帧的帧号, 根据帧号可以制定不同的非激活语音数据帧处 理策略, 如全部删除非激活语音数据帧, 或仅删除帧号为偶数的非激活 语音数据帧等等。  The rate of each voice data frame can be determined by the physical layer rate determination, so as to determine whether the rate frame is an activated voice data frame or an inactive voice data frame; and the Abis frame header includes a frame number of the voice data frame, according to the frame number. Develop different inactive voice data frame processing strategies, such as deleting all inactive voice data frames, or deleting only inactive voice data frames with even frame numbers.
本实施例系统如图 3所示, BTS和 BSC分别通过各自的接口处理模 块接入 Abis接口, 在 BTS和 BSC原有的信道处理模块和接口处理模块 之间, 分别加入压缩处理模块, 其中,  As shown in FIG. 3, the BTS and the BSC respectively access the Abis interface through the respective interface processing modules, and respectively add a compression processing module between the original channel processing module and the interface processing module of the BTS and the BSC, where
BTS设备的压缩处理模块用于识别出业务语音数据帧中的非激活语 音数据帧, 并按照预先设置的策略删除部分或全部非激活语音数据帧; 还用于识别下行语音数据帧中的静音, 生成非激活语音数据帧并填入所 识别出的静音位置。  The compression processing module of the BTS device is configured to identify the inactive voice data frame in the service voice data frame, and delete some or all of the inactive voice data frames according to a preset policy; and also to identify the silence in the downlink voice data frame. A non-activated speech data frame is generated and filled in the identified mute position.
BSC设备的压缩处理模块用于分析下行语音数据帧中 Abis帧头的 特征信息, 识别出非激活语音数据帧, 并按照预先指定的策略删除部分 或全部非激活语音数据帧; 还用于识别上行语音数据帧中的静音, 生成 非激活语音数据帧并填入所识别出的静音位置。 The compression processing module of the BSC device is used to analyze the Abis frame header in the downlink voice data frame. Feature information, identifying inactive voice data frames, and deleting some or all of the inactive voice data frames according to a pre-specified policy; also for identifying silence in the uplink voice data frame, generating inactive voice data frames and filling in the identified The mute position.
从以上描述可以看出, 在 BTS增加的压缩处理模块和在 BSC增加 的压缩处理模块的功能是相同的, 以 BTS设备为例,压缩处理模块的内 部单元的组成、 连接关系以及与外部其他模块的连接关系如图 4所示: 速率识别单元 401, 用于接收来自 BTS设备的信道处理模块的语音 数据帧, 并对该语音数据帧进行物理层速率判定, 如果该语音数据帧的 速率符合非激活语音数据帧速率的判定条件, 则生成一个允许删除指示 信号, 并将所生成的允许删除指示信号和该语音数据帧发送到删除处理 单元 402, 否则将该语音数据帧单独发送到删除处理单元 402。 本例中 假设采用的编码算法为 QCELP13K, 所述判定条件就是语音数据帧的速 率为全速率的 1/8, 即空口速率为 1800bps或 20ms包长为 20比特。  As can be seen from the above description, the functions of the compression processing module added in the BTS and the compression processing module added in the BSC are the same. Taking the BTS device as an example, the composition, connection relationship, and other modules of the internal unit of the compression processing module are the same. The connection relationship is as shown in FIG. 4: a rate identifying unit 401, configured to receive a voice data frame from a channel processing module of the BTS device, and perform physical layer rate determination on the voice data frame, if the rate of the voice data frame meets a non- Activating the determination condition of the speech data frame rate, generating an allow deletion indication signal, and transmitting the generated permission deletion indication signal and the speech data frame to the deletion processing unit 402, otherwise transmitting the speech data frame separately to the deletion processing unit 402. In this example, it is assumed that the encoding algorithm used is QCELP13K. The determination condition is that the rate of the voice data frame is 1/8 of the full rate, that is, the air interface rate is 1800 bps or the 20 ms packet length is 20 bits.
删除处理单元 402, 用于保存预先设置的删除处理策略; 当收到来 自速率识别单元 401的语音数据帧时,判断是否收到允许删除指示信号, 若收到, 则根据删除处理策略决定是否删除该语音数据帧, 若删除该语 音数据帧, 删除后在该语音数据帧所占用的时间段形成一段静音; 如果 没有收到允许删除指示信号或根据删除处理策略决定不删除该语音数 据帧, 则将所收到的语音数据帧发送到接口处理模块。  The deletion processing unit 402 is configured to save a preset deletion processing policy. When receiving the voice data frame from the rate identification unit 401, it is determined whether the permission deletion indication signal is received, and if received, determining whether to delete according to the deletion processing policy. If the voice data frame is deleted, the voice data frame is deleted, and a silence is formed in the time period occupied by the voice data frame; if the permission deletion instruction signal is not received or the voice data frame is not deleted according to the deletion processing policy, The received voice data frame is sent to the interface processing module.
所述删除处理策略可以是删除所有被识别出来的非激活语音数据 帧; 或者判断识别出的非激活语音数据帧的帧号是否满足一定的条件, 例如是否为偶数, 并删除满足条件的非激活语音数据帧。 所述删除处理 策略不局限于上面所举的例子, 可以根据实际需要设置。  The deletion processing policy may be: deleting all the identified inactive voice data frames; or determining whether the frame number of the identified inactive voice data frame satisfies certain conditions, such as whether it is an even number, and deleting the inactive condition that meets the condition. Voice data frame. The deletion processing policy is not limited to the above example, and may be set according to actual needs.
通过以上两个单元, 可以实现在语音接口发送语音数据帧时, 对非 激活语音数据帧进行删除处理。 在删除非激活语音数据帧的处理前, 语音数据帧所组成的语音数据 流是连续的, 没有静音出现; 而删除处理后, 在非激活语音数据帧被删 除的地方就形成了静音。 为了保证语音数据流是连续的, 本发明方案还 包括在语音数据帧的接收端, 检测出所接收的语音数据帧组成的数据流 中的静音, 在静音的位置补偿非激活语音数据帧。 则压缩处理模块还要 包括: Through the above two units, it is possible to delete the inactive voice data frame when the voice interface sends a voice data frame. Before the process of deleting the inactive voice data frame, the voice data stream composed of the voice data frame is continuous without muting; after the deletion process, the silence is formed where the inactive voice data frame is deleted. In order to ensure that the voice data stream is continuous, the solution of the invention further comprises: at the receiving end of the voice data frame, detecting silence in the data stream composed of the received voice data frame, and compensating the inactive voice data frame in the muted position. The compression processing module also includes:
静音时间判定单元 403 , 用于接收来自接口处理模块的语音数据帧 并将所接收的语音数据帧发送到恢复单元 404; 判断所述语音数据帧中 是否出现了静音, 如果是, 则向恢复单元 404发送一个恢复指示信号; 所述判断出现静音的做法为: 设置一个用于进行静音时间判定的定时 器, 该定时器时长与 Abis接口的语音数据帧发送间隔相同, 例如为 20 毫秒; 当接收到一个语音数据帧则对该定时器执行复位操作并重新开始 计时; 如果定时器超时, 则说明在定时器本次开始定时到超时之间这段 时间没有收到语音数据帧, 即出现了一个因为 1/8帧被删除造成的静音 间隔。  a silence time determining unit 403, configured to receive a voice data frame from the interface processing module and send the received voice data frame to the recovery unit 404; determine whether silence is present in the voice data frame, and if yes, to the recovery unit 404 sends a resume indication signal; the method of determining the silence is: setting a timer for performing silence time determination, the duration of the timer is the same as the voice data frame transmission interval of the Abis interface, for example, 20 milliseconds; When a voice data frame is reached, a reset operation is performed on the timer and the timing is restarted. If the timer expires, it indicates that no voice data frame is received between the current start timing of the timer and the timeout period, that is, a clock appears. The silence interval caused by the 1/8 frame being deleted.
恢复单元 404, 用于将所接收的语音数据帧发送到信道处理模块; 当收到恢复指示信号时, 则生成一个 1/8帧发送到信道处理模块。 所述 生成 1/8帧为: 生成语音净荷编码为 FF的 1/8帧, 并且帧号由静音位置 前一语音数据帧号模 16加 1得到。 因为帧号是 0 ~ 15循环的, 如果前 一帧的帧号为 15, 则所填入的 1/8帧的帧号为 0。 所生成的 1/8帧的语 音净荷编码还可以是其他舒适噪音的编码。  The recovery unit 404 is configured to send the received voice data frame to the channel processing module. When the resume indication signal is received, a 1/8 frame is generated and sent to the channel processing module. The generating 1/8 frame is: generating 1/8 frame of the voice payload coded as FF, and the frame number is obtained by the mute position of the previous voice data frame number modulo 16 plus 1. Since the frame number is 0 to 15 cycles, if the frame number of the previous frame is 15, the frame number of the filled 1/8 frame is 0. The generated 1/8 frame voice payload code can also be a code for other comfort noise.
参照上述对 BTS设备的压缩处理模块的描述, 可以得到 BSC设备 的压缩处理模块的組成结构和连接关系, 故不再赘述。  Referring to the description of the compression processing module of the BTS device, the composition and connection relationship of the compression processing module of the BSC device can be obtained, and therefore will not be described again.
上述压缩处理装置可为独立装置, 或者作为基站或基站控制器中的 功能模块, 集成在基站或基站控制器中。 采用上述装置对上行语音数据帧的处理流程如图 5所示, 包括如下 步骤: The above compression processing device may be an independent device or integrated as a functional module in a base station or a base station controller in a base station or a base station controller. The processing flow of the uplink voice data frame by using the foregoing apparatus is as shown in FIG. 5, and includes the following steps:
步骤 501 : 对 BTS信道处理模块输出的上行语音数据帧的物理层速 率进行判定, 识别出非激活语音数据帧。 在本实施例中, 所述非激活语 音数据帧就是 1/8速率的语音数据帧。  Step 501: Determine a physical layer rate of the uplink voice data frame output by the BTS channel processing module, and identify an inactive voice data frame. In this embodiment, the inactive voice data frame is a 1/8 rate voice data frame.
步骤 502:按照预先制定的策略删除部分或全部非激活语音数据帧, 并将删除处理后的上行语音数据流通过 BTS 的接口处理模块发送至 Abis接口的传输信道。  Step 502: Delete some or all of the inactive voice data frames according to the pre-defined policy, and send the deleted uplink voice data stream to the transport channel of the Abis interface through the interface processing module of the BTS.
步骤 503: BSC的接口处理模块接收 Abis接口的上行语音数据流, 识别出所接收凄 t据流中的静音;  Step 503: The interface processing module of the BSC receives the uplink voice data stream of the Abis interface, and identifies the silence in the received data stream;
步骤 504: 生成 1/8帧并填入所识别出的静音位置, 然后将上行语音 数据帧发送到 BSC的后续处理模块。  Step 504: Generate 1/8 frames and fill in the identified mute position, and then send the uplink voice data frame to the subsequent processing module of the BSC.
对下行语音数据帧的处理流程可以参照上述流程得出,故不再赘述。 使用 P.862方法对本发明实施例处理后的语音效果进行评估, 语音 数据的编码算法为 EVRC。 本发明实施例处理后, 语音数据的平均意见 得分( MOS )为 3.739分, 而现有技术处理后语音数据得分为 3.798分。 本发明处理造成 MOS仅下降了 0.059分, 可以认为话音质量没有下降。 因此, 本发明实施例在基本没有影响通话质量的前提下, 减少了 Abis 接口带宽资源的浪费, 极大提高 Abis接口语音承载效率。 若 Abis接口 中采用 E1接口, 每路 E1接口承载的话路数可以从通常的 180 ~ 190话 路提高到 240话路以上。  The processing flow of the downlink voice data frame can be obtained by referring to the above process, and therefore will not be described again. The speech effect processed by the embodiment of the present invention is evaluated by using the P.862 method, and the encoding algorithm of the speech data is EVRC. After processing in the embodiment of the present invention, the average opinion score (MOS) of the voice data is 3.739 points, and the score of the voice data after the prior art processing is 3.798 points. The processing of the present invention causes the MOS to drop by only 0.059 points, and it can be considered that the voice quality has not decreased. Therefore, the embodiment of the present invention reduces the waste of bandwidth resources of the Abis interface and greatly improves the voice bearer efficiency of the Abis interface under the premise that the call quality is not substantially affected. If the E1 interface is used in the Abis interface, the number of voice channels carried by each E1 interface can be increased from the normal 180 to 190 voice channels to more than 240 voice channels.
本发明方案还可以应用于 CDMA系统中其他语音接口 ,例如应用于 BSC与 MSC之间的 A接口, 包括 Al、 A2、 A5等接口。 本发明方案还 可广泛应用于其他无线通讯系统, 如 CDMA2000 系统。 本领域技术人 员应当认识到, 只要是连续传输可变速率的语音数据帧的场合, 都可通 过应用本发明方案达到提高传输效率的有益效果。 The solution of the present invention can also be applied to other voice interfaces in a CDMA system, for example, to an A interface between a BSC and an MSC, including interfaces such as Al, A2, and A5. The inventive solution can also be widely applied to other wireless communication systems, such as CDMA2000 systems. Those skilled in the art should recognize that as long as the variable rate speech data frame is continuously transmitted, The beneficial effects of improving the transmission efficiency are achieved by applying the solution of the present invention.
以上所述仅为本发明的较佳实施例而已, 并不用以限制本发明, 凡 在本发明的精神和原则之内所作的任何修改、 等同替换和改进等, 均应 包含在本发明的保护范围之内。  The above is only the preferred embodiment of the present invention, and is not intended to limit the present invention. Any modifications, equivalent substitutions and improvements made within the spirit and principles of the present invention should be included in the protection of the present invention. Within the scope.

Claims

权利要求书 Claim
1、 一种传输语音数据的方法, 其特征在于, 该方法包括如下步骤: 识别出语音数据流中的非激活语音数据帧; A method for transmitting voice data, the method comprising the steps of: identifying an inactive voice data frame in a voice data stream;
根据预先设置的删除策略, 删除部分或全部被识别出的非激活语音 数据帧;  Deleting some or all of the identified inactive voice data frames according to a preset deletion policy;
通过语音接口传输删除处理后的语音数据流。  The deleted voice data stream is transmitted through the voice interface.
2、根据权利要求 1所述的方法, 其特征在于, 所述识别出语音数据 流中的非激活语音数据帧包括: 对当前所要发送的语音数据帧进行物理 层速率判定, 若所述语音数据帧的速率符合非激活语音数据帧的判定条 件, 则该语音速率帧为非激活语音数据帧。  The method according to claim 1, wherein the identifying the inactive voice data frame in the voice data stream comprises: performing physical layer rate determination on the currently transmitted voice data frame, if the voice data The rate of the frame conforms to the decision condition of the inactive voice data frame, and the voice rate frame is the inactive voice data frame.
3、根据权利要求 2所述的方法, 其特征在于, 所述语音数据帧的速 率符合非激活语音数据帧的判定条件为: 该语音数据帧的速率为 1/8速 率。  The method according to claim 2, wherein the rate of the voice data frame conforming to the inactive voice data frame is: the rate of the voice data frame is 1/8 rate.
4、根据权利要求 1所述的方法, 其特征在于, 所述删除部分被识别 出的非激活语音数据帧为: 判定所述非激活语音数据帧的帧号是否满足 预先设置的删除条件, 若是则删除该非激活语音数据帧。  The method according to claim 1, wherein the deleted part of the inactivated voice data frame is: determining whether the frame number of the inactive voice data frame satisfies a preset deletion condition, if Then delete the inactive voice data frame.
5、根据权利要求 1所述的方法, 其特征在于, 所述通过语音接口传 输删除处理后的语音数据流之后, 进一步包括:  The method according to claim 1, wherein after the deleting the processed voice data stream by using the voice interface, the method further includes:
接收所述删除处理后的语音数据流, 识别出所接收的语音数据流中 的静音;  Receiving the deleted voice data stream, and identifying the silence in the received voice data stream;
生成非激活语音数据帧并填充在所识别出的静音的位置。  A non-activated speech data frame is generated and populated at the identified muted position.
6、根据权利要求 5所述的方法, 其特征在于,设置时长为语音数据 帧发送间隔的定时器, 则所述识别出语音数据流中的静音包括:  The method according to claim 5, wherein the setting of the duration of the voice data frame transmission interval, the identifying the silence in the voice data stream comprises:
当接收到一个语音数据帧时, 所述定时器开始计时, 如果在超时之 前收到下一个语音数据帧, 则将该定时器复位并继续执行本步骤; 如果 所述定时器超时, 则该定时器本次开始计时到超时之间这段时间为静 音。 When a voice data frame is received, the timer starts timing, if it is timed out If the next voice data frame is received before, the timer is reset and the step is continued; if the timer expires, the time between the timer starts to timeout and the timeout is muted.
7、根据权利要求 5所述的方法, 其特征在于, 所述生成非激活语音 数据帧为:  The method according to claim 5, wherein the generating the inactive voice data frame is:
生成语音净荷编码为 FF, 语音数据帧号为上一个语音数据帧号模 16加 1的 1/8速率帧。  The generated voice payload code is FF, and the voice data frame number is the 1/8 rate frame of the previous voice data frame number modulo 16 plus 1.
8、根据权利要求 1至 6任一项所述的方法, 其特征在于, 所述语音 接口为基站与基站控制器之间的语音接口, 或者基站控制器与移动交换 中心之间的语音接口。  The method according to any one of claims 1 to 6, wherein the voice interface is a voice interface between a base station and a base station controller, or a voice interface between the base station controller and the mobile switching center.
9、 一种传输语音数据的装置, 其特征在于, 该装置包括: 速率识别单元, 用于接收语音数据帧并将所接收的语音数据帧发送 到删除处理单元; 判断所接收的语音数据帧是否是非激活语音数据帧, 根据判断的结果生成删除指示信号, 并将所生成的删除指示信号发送到 删除处理单元;  A device for transmitting voice data, the device comprising: a rate identifying unit, configured to receive a voice data frame and send the received voice data frame to a deletion processing unit; and determine whether the received voice data frame is Is a non-activated voice data frame, generating a deletion indication signal according to the result of the determination, and transmitting the generated deletion indication signal to the deletion processing unit;
删除处理单元, 用于保存预先设置的删除处理策略; ^居所接收的 删除指示信号和删除处理策略对当前接收的语音数据帧进行删除操作, 或者输出所接收的语音数据帧。  The deletion processing unit is configured to save the preset deletion processing policy; the deletion indication signal and the deletion processing policy received by the residence delete the currently received voice data frame, or output the received voice data frame.
10、根据权利要求 9所述的装置, 其特征在于, 该装置进一步包括: 静音时间判定单元, 用于接收语音数据帧并将所接收的语音数据帧 发送到恢复单元; 并判断所接收的语音数据帧中是否出现了静音,若是, 则向恢复单元发送恢复指示信号;  10. The apparatus according to claim 9, wherein the apparatus further comprises: a silence time determining unit, configured to receive a voice data frame and transmit the received voice data frame to a recovery unit; and determine the received voice Whether silence is present in the data frame, and if so, sending a recovery indication signal to the recovery unit;
恢复单元,用于输出所接收的语音数据帧; 当收到恢复指示信号时, 则生成非激活语音数据帧并输出所述非激活语音数据帧。  And a recovery unit, configured to output the received voice data frame; when receiving the resume indication signal, generate an inactive voice data frame and output the inactive voice data frame.
11、一种传输语音数据的基站, 包括信道处理模块和接口处理模块, 其特征在于, 该基站还包括压缩处理模块, 用于识别出来自信道处理模 块的语音数据帧中的非激活语音数据帧, 并按照预先设置的策略删除部 分或全部非激活语音数据帧, 并将删除处理后的语音数据帧发送至接口 处理模块。 11. A base station for transmitting voice data, comprising a channel processing module and an interface processing module, The base station further includes a compression processing module, configured to identify a non-activated voice data frame in a voice data frame from the channel processing module, and delete some or all of the inactive voice data frames according to a preset policy, and The deleted voice data frame is sent to the interface processing module.
12、根据权利要求 11所述的基站, 其特征在于, 所述压缩处理模块 包括:  The base station according to claim 11, wherein the compression processing module comprises:
速率识别单元, 用于接收来自信道处理模块的语音数据帧并将所接 收的语音数据帧发送到删除处理单元; 判断所接收的语音数据帧是否是 非激活语音数据帧, 根据判断地结果生成删除指示信号, 并将所生成的 删除指示信号发送到删除处理单元;  a rate identifying unit, configured to receive a voice data frame from the channel processing module and send the received voice data frame to the deletion processing unit; determine whether the received voice data frame is an inactive voice data frame, and generate a deletion indication according to the determined result Signaling, and transmitting the generated deletion indication signal to the deletion processing unit;
删除处理单元, 用于保存预先设置的删除处理策略; 根据所接收的 删除指示信号和删除处理策略对当前接收的语音数据帧进行删除操作; 或者将所接收的语音数据帧发送到接口处理模块。  a deletion processing unit, configured to save a preset deletion processing policy; perform a deletion operation on the currently received voice data frame according to the received deletion indication signal and the deletion processing policy; or send the received voice data frame to the interface processing module.
13、根据权利要求 12所述的基站, 其特征在于, 所述压缩处理模块 进一步包括:  The base station according to claim 12, wherein the compression processing module further comprises:
静音时间判定单元, 用于接收来自接口处理模块的语音数据帧并将 所接收的语音数据帧发送到恢复单元; 并判断所接收的语音数据帧中是 否出现了静音, 若是则向恢复单元发送恢复指示信号;  a silence time determining unit, configured to receive a voice data frame from the interface processing module and send the received voice data frame to the recovery unit; and determine whether silence is present in the received voice data frame, and if yes, send and recover to the recovery unit Indication signal
恢复单元, 用于将所接收的语音数据帧发送到信道处理模块; 当收 到恢复指示信号时, 则生成非激活语音数据帧并将所述非激活语音数据 帧发送到信道处理模块。  And a recovery unit, configured to send the received voice data frame to the channel processing module; when the recovery indication signal is received, generate an inactive voice data frame and send the inactive voice data frame to the channel processing module.
14、 一种传输语音数据的基站控制器, 包括信道处理模块和接口处 理模块, 其特征在于, 该基站还包括压缩处理模块, 用于识别出来自信 道处理模块的语音数据帧中的非激活语音数据帧, 并按照预先设置的策 略删除部分或全部非激活语音数据帧, 并将删除处理后的语音数据帧发 送至接口处理模块。 A base station controller for transmitting voice data, comprising a channel processing module and an interface processing module, wherein the base station further comprises a compression processing module, configured to identify inactive voice in a voice data frame from the channel processing module Data frame, and delete some or all of the inactive voice data frames according to a preset policy, and delete the processed voice data frames Send to the interface processing module.
15、根据权利要求 14所述的基站控制器, 其特征在于, 所述压缩处 理模块包括:  The base station controller according to claim 14, wherein the compression processing module comprises:
速率识别单元, 用于接收来自信道处理模块的语音数据帧并将所接 收的语音数据帧发送到删除处理单元; 判断所接收的语音数据帧是否是 非激活语音数据帧, 4艮据判断地结果生成删除指示信号, 并将所生成的 删除指示信号发送到删除处理单元;  a rate identifying unit, configured to receive a voice data frame from the channel processing module and send the received voice data frame to a deletion processing unit; determine whether the received voice data frame is an inactive voice data frame, and generate a result according to the judgment result Deleting the indication signal, and transmitting the generated deletion indication signal to the deletion processing unit;
删除处理单元, 用于保存预先设置的删除处理策略; 根据所接收的 删除指示信号和删除处理策略对当前接收的语音数据帧进行删除操作; 或者将所接收的语音数据帧发送到接口处理模块。  a deletion processing unit, configured to save a preset deletion processing policy; perform a deletion operation on the currently received voice data frame according to the received deletion indication signal and the deletion processing policy; or send the received voice data frame to the interface processing module.
16、根据权利要求 15所述的基站控制器, 其特征在于, 所述压缩处 理模块进一步包括:  The base station controller according to claim 15, wherein the compression processing module further comprises:
静音时间判定单元, 用于接收来自接口处理模块的语音数据帧并将 所接收的语音数据帧发送到恢复单元; 并判断所接收的语音数据帧中是 否出现了静音, 若是则向恢复单元发送恢复指示信号;  a silence time determining unit, configured to receive a voice data frame from the interface processing module and send the received voice data frame to the recovery unit; and determine whether silence is present in the received voice data frame, and if yes, send and recover to the recovery unit Indication signal
恢复单元, 用于将所接收的语音数据帧发送到信道处理模块; 当收 到恢复指示信号时, 则生成非激活语音数据帧并将所述非激活语音数据 帧发送到信道处理模块。  And a recovery unit, configured to send the received voice data frame to the channel processing module; when the recovery indication signal is received, generate an inactive voice data frame and send the inactive voice data frame to the channel processing module.
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