WO2007118392A1 - A method and device for transmitting voice data - Google Patents
A method and device for transmitting voice data Download PDFInfo
- 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
- Authority
- WO
- WIPO (PCT)
- Prior art keywords
- voice data
- data frame
- processing module
- received
- deletion
- Prior art date
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W28/00—Network traffic management; Network resource management
- H04W28/02—Traffic management, e.g. flow control or congestion control
- H04W28/06—Optimizing 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
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Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
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CN200610072478.0 | 2006-04-17 | ||
CNB2006100724780A CN100461906C (en) | 2006-04-17 | 2006-04-17 | Method and device for improving CDMA system voice interface transmission efficiency |
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WO2007118392A1 true WO2007118392A1 (en) | 2007-10-25 |
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PCT/CN2007/000283 WO2007118392A1 (en) | 2006-04-17 | 2007-01-25 | A method and device for transmitting voice data |
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WO (1) | WO2007118392A1 (en) |
Families Citing this family (3)
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CN102137440A (en) * | 2010-12-13 | 2011-07-27 | 华为软件技术有限公司 | Method and device for realizing voice communication |
WO2018205080A1 (en) * | 2017-05-08 | 2018-11-15 | 海能达通信股份有限公司 | Method, device and apparatus for carrying channel associated signaling during voice calling process |
CN110650167B (en) | 2018-06-26 | 2021-02-23 | 华为技术有限公司 | Communication method and device |
Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
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US5870397A (en) * | 1995-07-24 | 1999-02-09 | International Business Machines Corporation | Method and a system for silence removal in a voice signal transported through a communication network |
CN1237304A (en) * | 1996-09-13 | 1999-12-01 | 夸尔柯姆股份有限公司 | Conference calling system and method for wireless communications channel |
US6324188B1 (en) * | 1997-06-12 | 2001-11-27 | Sharp Kabushiki Kaisha | Voice and data multiplexing system and recording medium having a voice and data multiplexing program recorded thereon |
Family Cites Families (3)
Publication number | Priority date | Publication date | Assignee | Title |
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JP3556419B2 (en) * | 1996-12-09 | 2004-08-18 | 株式会社東芝 | Portable wireless telephone |
US6526128B1 (en) * | 1999-03-08 | 2003-02-25 | Agere Systems Inc. | Partial voice message deletion |
JP4296154B2 (en) * | 2003-04-04 | 2009-07-15 | 三菱電機株式会社 | Audio transmission system |
-
2006
- 2006-04-17 CN CNB2006100724780A patent/CN100461906C/en not_active Expired - Fee Related
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2007
- 2007-01-25 WO PCT/CN2007/000283 patent/WO2007118392A1/en active Application Filing
- 2007-01-25 CN CNA2007800003421A patent/CN101317480A/en active Pending
Patent Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
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US5870397A (en) * | 1995-07-24 | 1999-02-09 | International Business Machines Corporation | Method and a system for silence removal in a voice signal transported through a communication network |
CN1237304A (en) * | 1996-09-13 | 1999-12-01 | 夸尔柯姆股份有限公司 | Conference calling system and method for wireless communications channel |
US6324188B1 (en) * | 1997-06-12 | 2001-11-27 | Sharp Kabushiki Kaisha | Voice and data multiplexing system and recording medium having a voice and data multiplexing program recorded thereon |
Also Published As
Publication number | Publication date |
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CN101317480A (en) | 2008-12-03 |
CN100461906C (en) | 2009-02-11 |
CN1984398A (en) | 2007-06-20 |
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