CN100353352C - Method of reducing data transmission delay in coding decoding process and its device - Google Patents

Method of reducing data transmission delay in coding decoding process and its device Download PDF

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Publication number
CN100353352C
CN100353352C CNB03109757XA CN03109757A CN100353352C CN 100353352 C CN100353352 C CN 100353352C CN B03109757X A CNB03109757X A CN B03109757XA CN 03109757 A CN03109757 A CN 03109757A CN 100353352 C CN100353352 C CN 100353352C
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China
Prior art keywords
voice packet
data
voice
module
buffer zone
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CNB03109757XA
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CN1538315A (en
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邹立建
曾思南
张军
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Huawei Technologies Co Ltd
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Huawei Technologies Co Ltd
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Abstract

The present invention relates to a method for reducing data transmission delays in the processes of decoding and encoding voice, which is suitable for the transmission process of the conversion of more than one encoding format of IP voice. A decoding module first decodes a source code voice packet in one format into the voice packet in a standard format, and then an encoding module encodes the voice packet in the standard format into the voice packet in another format so as to continue transmission. The method also comprises the following steps: a. after the decoding module decodes the input source code voice packet into voice packet data in the standard format, the voice packet data in the standard format is stored in a voice data buffer zone; b. the encoding module directly reads the voice packet data decoded in step A from the voice data buffer zone, and the voice packet data is encoded into the voice packet in a needed format so as to be output. The transmission delays caused by the processes of encoding and decoding the voice packet can be reduced by the present invention.

Description

The method and the device thereof of data transmission time-delay in the cataloged procedure separated in a kind of minimizing
Technical field
The present invention relates to data transmission technology, be meant a kind of method and device thereof that data transmission time-delay in the cataloged procedure separated in voice that reduce especially.
Background technology
Along with Voice over IP (VoIP, Voice over Internet Protocol) development of technology and the interconnected requirement of diverse network, in the process of voice coding transmission, sometimes need voice to be transformed into another kind of coded system and continue transmission again, sometimes even can need repeatedly the conversion of coded system from a kind of coded system.As adaptive multi-rate (AMR, Adaptive Multi-Rate) with G.729, G.723 wait conversion between the coded system.
Coding and decoding is the algorithm that a kind of software is realized.At present, the method that realizes the voice coding modes conversion is finished with two processor cascades.As shown in Figure 1, a processor A is as demoder, and another processor B is as scrambler.Processor A is finished the decode operation of the source code voice packet of input, and decoder module is decoded the voice packet of input, is sent to the serial data interface module with standard data format and exports; Processor B receives data by the serial data interface module, sends data to coding module and encodes, the purpose encoded voice bag of the output form that requires.
In the voice coding modes transfer process, this cascade system will be transmitted two data between processor through the serial data interface module, the suitable converter of the effect of serial data interface module, the data stream transformat and the external data serial transfer form of processor inside are changed, therefore in this cascade, Data transmission is introduced bigger transmission delay through the dual serial data-interface between CPU.As the VoP of 20ms size,, need the time of 20ms at least if by the serial ports transmission.Serial data interface module between CPU becomes the speed bottleneck of voice packet coded system transfer process.
Summary of the invention
In view of this, fundamental purpose of the present invention has been to provide a kind of device that voice are separated the method for data transmission time-delay in the cataloged procedure and realized the method that reduces.
Realize the present invention, need following method:
At first, the voice packet input interface module is converted into the source code voice packet data stream of internal format, be transferred to decoder module, decoder module is read corresponding algorithm from algoritic module data stream is decoded, again decoded data are directly write the speech data buffer zone, and represent that by zone bit is set these data are new data;
Then, coding module directly reads these data from the speech data buffer zone, and removing corresponding marker bit, according to the respective algorithms of reading data stream is encoded again from algoritic module, stream data transmission behind the coding is given the voice packet output interface module, and the voice packet output interface module is converted into the output of purpose encoded voice packet format with data stream.
The device of realizing the method comprises decoder module, speech data buffer zone, coding module and algoritic module at least.
By said method as can be seen, the present invention is incorporated into demoder and scrambler in the processor, by shared drive communicate voice data stream, has reduced the transmission delay of data in separating cataloged procedure.
Description of drawings
Fig. 1 for separate by the serial ports cascade, the synoptic diagram of scrambler.
Fig. 2 separates the composition and the principle schematic of data transmission time-delay in the cataloged procedure for the present invention realizes reducing.
Embodiment
Below by specific embodiment with reference to accompanying drawing 2, the present invention is described in more detail.
Fig. 2 separates the composition and the principle schematic of data transmission time-delay in the cataloged procedure for the present invention realizes reducing.As shown in Figure 2, comprise decoder module and coding module in the processor, decoder module adopts shared drive to be connected as the speech data buffer zone with coding module, wherein also comprises algoritic module, and storage is used for the algorithm of Code And Decode.Concrete voice packet coded format switch process is as follows:
At first, the voice packet input interface module is converted into the source code voice packet data stream of internal format, be transferred to decoder module, decoder module is read corresponding algorithm from algoritic module data stream is decoded, again decoded data are directly write the speech data buffer zone, and represent that by zone bit is set these data are new data;
Then, coding module directly reads these data from the speech data buffer zone, and removing corresponding marker bit, according to the respective algorithms of reading data stream is encoded again from algoritic module, stream data transmission behind the coding is given the voice packet output interface module, and the voice packet output interface module is converted into the output of purpose encoded voice packet format with data stream.
Under this scheme, the time quantum that reduces is about the frame length of source code voice packet input voice usually.For example: to the G.729 conversion of voice coding modes, source code voice packet form is AMR from AMR, frame length 20ms, and purpose adopts the encoded voice packet format for G.729, frame length 10ms; Adopt this method can reduce the time-delay of about 20ms than serial data interface method.
Wherein, when the change of format of source code voice packet or purpose encoded voice bag, can replace the original decoding or the algorithm of coding by the voice packet input interface module to algoritic module input new algorithm software package or instruction.
The above only is preferred embodiment of the present invention, and is in order to restriction the present invention, within the spirit and principles in the present invention not all, any modification of being done, is equal to replacement, improvement etc., all should be included within protection scope of the present invention.

Claims (5)

1, the method that data transmission is delayed time in the cataloged procedure separated in a kind of voice that reduce, be applicable to that ip voice carries out in the transmission course of more than one coded format conversions, decoder module is decoded as standard format with a kind of source code voice packet of form earlier, coding module is encoded to another kind of required voice packet form continuation transmission with the voice packet of standard format again, it is characterized in that this method is further comprising the steps of:
A, decoder module deposit the speech data buffer zone in after the source code voice packet of importing is decoded as the voice packet data of standard format, and the zone bit of sign new data is set in the data of speech data buffer zone;
B, coding module are removed the zone bit of institute's sense data directly from the decoded voice packet data of speech data buffer zone read step A, are encoded to the voice packet output of required form.
2, method according to claim 1 is characterized in that: described decoder module and coding module are finished decoding and coding according to corresponding algorithm respectively.
3, method according to claim 2 is characterized in that this method further comprises: the described algorithm that is used to decode or encode upgrades by input new algorithm or instruction.
4, the device that data transmission is delayed time between encode processor separated in a kind of voice that reduce, and it is characterized in that this device comprises:
Decoder module, the data stream that the source code voice packet that is used for importing is decoded as standard format deposits the speech data buffer zone in;
The speech data buffer zone is used to store the data stream between decoder module and the coding module, the data stream of storage is provided with the zone bit of sign new data;
Coding module is used for from the data stream of speech data buffer zone readout sign quasiconfiguaration, removes the zone bit of institute's sense data, is encoded to the voice packet of required form.
5, device according to claim 4 is characterized in that this device further comprises: the algoritic module of used algorithm when being used for memory encoding and decoding.
CNB03109757XA 2003-04-15 2003-04-15 Method of reducing data transmission delay in coding decoding process and its device Expired - Fee Related CN100353352C (en)

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CN100353352C true CN100353352C (en) 2007-12-05

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Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102890934B (en) * 2012-10-11 2014-06-11 广州海格通信集团股份有限公司 Multi-channel duplex vocoder and multi-channel duplex vocoder system
CN116032422A (en) * 2019-05-15 2023-04-28 华为技术有限公司 Data transmission method and device
CN111200692B (en) * 2019-12-24 2021-10-26 广州市高科通信技术股份有限公司 Voice equipment, processing method, device and storage medium for network telephone

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6160915A (en) * 1996-11-07 2000-12-12 Sony Corporation Coded signal transmission method and apparatus
US6195358B1 (en) * 1997-03-15 2001-02-27 International Business Machines Corp. Internet telephony signal conversion
US6253293B1 (en) * 1997-11-14 2001-06-26 Cirrus Logic, Inc. Methods for processing audio information in a multiple processor audio decoder

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6160915A (en) * 1996-11-07 2000-12-12 Sony Corporation Coded signal transmission method and apparatus
US6195358B1 (en) * 1997-03-15 2001-02-27 International Business Machines Corp. Internet telephony signal conversion
US6253293B1 (en) * 1997-11-14 2001-06-26 Cirrus Logic, Inc. Methods for processing audio information in a multiple processor audio decoder

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