CN1474581A - Method for realizing voice coding and decoding changeover and intercommunication - Google Patents
Method for realizing voice coding and decoding changeover and intercommunication Download PDFInfo
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- CN1474581A CN1474581A CNA021300321A CN02130032A CN1474581A CN 1474581 A CN1474581 A CN 1474581A CN A021300321 A CNA021300321 A CN A021300321A CN 02130032 A CN02130032 A CN 02130032A CN 1474581 A CN1474581 A CN 1474581A
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Abstract
This invention relates to a method for realizing phonetic code and decode conversion and inter communication, first of all, arranging multiple phonetic code, decode algorithms suitable for different networks in a phonetic process unit. When receiving phonetic signals carried by phonetic code form from a network A, the phonetic decode function corresponding to a form is called and convert it to linear PCM data then to call a phonetic code function corresponding to form b is called to convert PCM data to form b to be sent to network B. The entire process proceedure is finished in a unit, either finish conversion of low bit rate code/decode or conversion from low bit rate to G.711.
Description
Technical field
The present invention relates to a kind of method that realizes encoding and decoding speech conversion and intercommunication, belong to the voice transmission technology field.
Background technology
At public switch telephone network, in (hereinafter to be referred as PSTN) system, voice signal is that the G.711 voice encoding and decoding format with 64kbps transmits on time division multiplexing (hereinafter to be referred as TDM) link.Mobile communcations system that grows up subsequently (as gsm system, WCDM system etc.) and IP telephony system all pass through to use low bit rate encoding and decoding speech technology, to save air interface and transfer resource.Along with the extensive use of mobile communication and IP phone, increasing voice signal will transmit in the mode of low bit rate encoding and decoding.In at present up-to-date 3G mobile communcations system, the packet-based form of core net, voice signal is carried on IP bag or the ATM cell with the form of low bit rate encoding and decoding, converts other encoding and decoding speech (as G.711) and other networks (as PSTN) intercommunication on gateway exchange to.The realization technology of this speech and the IP telephony system of fixed network are closely similar.But different mobile communcations systems uses different encoding and decoding speech standards, and for example, Wideband Code Division Multiple Access (WCDMA) (hereinafter to be referred as WCDMA) system uses adaptive multi-rate (hereinafter to be referred as AMR) encoding and decoding speech; CDMA 2000 (hereinafter to be referred as CDMA2000) system uses enhanced variable rate coding (hereinafter to be referred as EVRC), high pass Qualcomm Code Excited Linear Prediction (QCELP) (hereinafter to be referred as QCELP), SMV encoding and decoding speech.IP telephony system also has its special encoding and decoding speech, generally uses G.723.1 and G.729 encoding and decoding speech of ITU-T.Along with moving and the extensive use of fixed network packet voice, the intercommunication between the different mobile communication system, between mobile communcations system and the fixed network IP phone will become focus.
Traditional implementation method, different networks is built the gateway exchange gateway respectively, and the two couples together by the TDM link.A kind of speech of network converts G.711 encoding and decoding speech at the gateway exchange gateway place of this network, by the gateway exchange gateway place of TDM link transmission to another network, the gateway exchange gateway of another network converts thereof into the voice encoding and decoding format of present networks, as shown in Figure 1.
The shared gateway of perhaps different networks, but intra-gateway is handled separately encoding and decoding speech respectively by different pronounciation processing chips or subsystem, and connect with different network, pass through G.711 encoding and decoding speech of TDM link transmission between speech chip/subsystem, with the conversion between the encoding and decoding speech of realizing different system, the essence of its speech processes is with to build the gateway exchange gateway respectively identical, as shown in Figure 2.
Traditional implementation method need convert the encoding and decoding speech of heterogeneous networks to G.711 encoding and decoding speech, and need by two audio coder ﹠ decoder (codec)s of TDM link link, therefore cause the waste of TDM resource (relay resource or intra-gateway TDM resource) and increased time-delay (the sampling time-delay of TDM).Owing to when the heterogeneous networks intercommunication, need encoding and decoding speech to do the conversion of encoding and decoding speech a-G.711-encoding and decoding speech b, be equivalent to the cascade of three grades of encoding and decoding speechs, therefore also reduced voice quality.
Summary of the invention
The objective of the invention is to overcome existing encoding and decoding speech technology waste TDM resource, increase the shortcoming of delaying time and voice quality being reduced, a kind of method that realizes encoding and decoding speech conversion and intercommunication is proposed, to save the TDM resource, reduce the sampling time-delay, and avoid low bit rate encoding and decoding speech and the G.711 cascade of encoding and decoding speech, improve voice quality.
The realization encoding and decoding speech conversion that the present invention proposes and the method for intercommunication comprise following each step:
1, in Audio Processing Unit (chip or subsystem), disposes multiple voice coder, the decoding algorithm that is applicable to heterogeneous networks;
2, Audio Processing Unit receives the voice signal with the carrying of speech coding a form from network A;
3, call and the corresponding tone decoding algorithm of speech coding a form, convert speech coding a to linear impulsive coded modulation (hereinafter to be referred as PCM) data;
4, call and the corresponding speech coding algorithm of speech coding b form, the linear PCM data transaction is become speech coding b form, send to network B.
Voice coder in the said method, decoding algorithm are respectively any one standard coding and decoding algorithm.
The realization encoding and decoding speech conversion that the present invention proposes and the method for intercommunication, the entire process process is finished in a processing unit fully, do not need the link of TDM link, do not need middle by being transformed into the G.711 conversion of encoding and decoding speech yet, therefore adopt method of the present invention, both can finish the conversion between the low bit rate encoding and decoding speech, and can realize equally from the low bit rate encoding and decoding speech to the G.711 conversion of encoding and decoding speech, and not increase any burden.
Description of drawings
Fig. 1 is respectively the network structure of existing two kinds of different encoding and decoding speech conversion methods with Fig. 2.
Fig. 3 is a network structure of realizing the inventive method.
Fig. 4 is the flow chart of the inventive method.
Embodiment
The realization encoding and decoding speech conversion that the present invention proposes and the method for intercommunication at first dispose multiple voice coder, the decoding algorithm that is applicable to heterogeneous networks in same Audio Processing Unit (chip or subsystem); Audio Processing Unit receives the voice signal with the carrying of speech coding a form from network A; Call and the corresponding tone decoding algorithm of speech coding a form, convert speech coding a to the linear PCM data; Call and the corresponding speech coding algorithm of speech coding b form, the linear PCM data transaction is become speech coding b form, send to network B.
The network configuration of the inventive method as shown in Figure 3, the entire process process is finished in a processing apparatus fully, does not need the TDM link, does not change by encoding and decoding speech G.711 in the middle of also not needing, its handling process as shown in Figure 4.Should be pointed out that and use method of the present invention that both can finish the conversion between the low bit rate encoding and decoding speech, the low bit rate encoding and decoding speech that can realize equally arrives the G.711 conversion of encoding and decoding speech, and does not increase any burden.
Introduce embodiments of the invention below:
The portable terminal of supposing 1 WCDMA makes a call to 1 fixed network IP phone user, and the WCDMA terminal is used the AMR encoding and decoding speech, packets of voice of every 20ms transmission; Fixed network IP phone user uses G.729 encoding and decoding speech, every 20ms transmission packets of voice (being equivalent to comprise 2 frame 10msG.729 speech frames).Traditional implementation procedure, AMR encoding and decoding speech function is to realize in two different Audio Processing Units with encoding and decoding speech function G.729.For example, use the TMS320C5410 dsp chip of a slice TI company to realize AMR encoding and decoding speech function, use another sheet TMS320C5410 chip to realize G.729 encoding and decoding speech function, between two chips by the TDM link bearer G.711 the speech coding form come intercommunication.The WCDMA terminal to fixed network IP phone user's one-way communication path as shown in Figure 5.At first, the TMS320C5410 dsp chip of realizing AMR encoding and decoding speech function receives AMR speech frame (comprising the 20ms voice messaging) from the HPI interface, call AMR tone decoding algorithm routine, convert the AMR speech frame to linear PCM signal (comprising the 20ms voice messaging), calling G.711 then, the speech coding algorithm program becomes G.711 code stream (comprising the 20ms voice messaging) with the linear PCM conversion of signals.By the TDM link will be G.711 code stream be sent to and realize the G.729 TMS320C5410 dsp chip of encoding and decoding speech function, this chip receives the G.711 code stream of 20ms from the TDM interface, call then G.711 the decoding algorithm program G.711 code stream convert the linear PCM signal to, then call twice G.729 the encryption algorithm program linear PCM conversion of signals is become G.729 speech frame (comprising the 20ms voice messaging) of two frames, see off by the HPI interface then.Wherein the time-delay that causes of each several part is:
AMR encoder: 45ms
AMR decoder: 20ms
G.729 encoder: 45ms
G.729 decoder: 20ms
G.711 Code And Decode device: can ignore
Summation: 130ms
The method applied in the present invention, realizing the AMR voice coder simultaneously with a slice TMS320C5410 dsp chip, decoding algorithm and voice coder G.729, decoding algorithm, its handling process is: at first, the TMS320C5410 dsp chip receives AMR speech frame (comprising the 20ms voice messaging) from the HPI interface, call AMR tone decoding algorithm routine, convert the AMR speech frame to linear PCM signal (comprising the 20ms voice messaging), then call twice G.729 the encryption algorithm program linear PCM conversion of signals is become G.729 speech frame (comprising the 20ms voice messaging) of two frames, see off by the HPI interface then.
Wherein the time-delay that causes of each several part is:
AMR encoder: 45ms
AMR decoder: 20ms
G.729 encoder: 40ms (having saved the TDM sampling time)
G.729 decoder: 20ms
Summation: 125ms
Therefore adopt method of the present invention, one-way delay can be saved 5ms.
Claims (4)
1, a kind of method that realizes encoding and decoding speech conversion and intercommunication comprises following each step:
(1) in Audio Processing Unit, disposes multiple voice coder, the decoding algorithm that is applicable to heterogeneous networks;
(2) Audio Processing Unit receives the voice signal with the carrying of speech coding a form from network A;
(3) call and the corresponding tone decoding algorithm of speech coding a form, convert speech coding a to the linear impulsive coding modulation data;
(4) call the corresponding speech coding algorithm of speech coding b form, convert the linear impulsive coding modulation data to speech coding b form, send to network B.
2, the method for claim 1 is characterized in that wherein said voice coder, decoding algorithm are standard coding and decoding algorithm.
3, method as claimed in claim 2 is characterized in that wherein said speech coding algorithm is speech coding algorithm G.729.
4, method as claimed in claim 2 is characterized in that wherein said tone decoding algorithm is an AMR tone decoding algorithm.
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| CNA021300321A CN1474581A (en) | 2002-08-10 | 2002-08-10 | Method for realizing voice coding and decoding changeover and intercommunication |
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2008138261A1 (en) * | 2007-05-11 | 2008-11-20 | Huawei Technologies Co., Ltd. | Ip multimedia subsystem, coding and decoding conversion control method and device thereof |
| CN101568043B (en) * | 2009-04-30 | 2011-02-02 | 尹海盛 | In-band signaling interconnection device among PCM devices |
| CN101656735B (en) * | 2009-09-11 | 2013-02-13 | 中兴通讯股份有限公司 | Code flow processing method and device in communication link |
-
2002
- 2002-08-10 CN CNA021300321A patent/CN1474581A/en active Pending
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2008138261A1 (en) * | 2007-05-11 | 2008-11-20 | Huawei Technologies Co., Ltd. | Ip multimedia subsystem, coding and decoding conversion control method and device thereof |
| CN101568043B (en) * | 2009-04-30 | 2011-02-02 | 尹海盛 | In-band signaling interconnection device among PCM devices |
| CN101656735B (en) * | 2009-09-11 | 2013-02-13 | 中兴通讯股份有限公司 | Code flow processing method and device in communication link |
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