CN102237091B - Frame division multiplexing-based adaptive voice service bearing method and system - Google Patents

Abstract

The invention relates to a frame division multiplexing-based adaptive voice service bearing method and system in a time division-synchronization code division multiple access (TD-SCDMA) system. The method comprises the following steps of: at a radio network controller, selecting voice coding/decoding rate according to a voice service load of a network and distributing radio resources to user equipment based on a radio frame; and at the user equipment, transmitting and/or receiving voice signals based on the voice coding/decoding rate selected by the radio network controller and based on the radio resources distributed by the radio frame. According to the method and the system, when the voice service load in the network is lower, high voice coding/decoding rate can be selected, and the radio resources are distributed to the user equipment in a mode of continuous radio frames; when the voice service load in the network is higher, low voice coding/decoding rate can be selected, and the same radio resources are distributed to at least two pieces of user equipment by using at least one continuous radio frame as a unit; and therefore, frame division multiplexing-based adaptive voice service bearing is realized.

Description

Adaptive Voice Service Bearing Method and System Based on Frame Division Multiplexing

technical field

The present invention relates to voice communication in the TD-SCDMA system, more specifically, to a method and system for carrying adaptive voice services based on frame division multiplexing in the TD-SCDMA system, a radio network controller (RNC) and user equipment ( UE).

Background technique

In the global mobile communication (GSM) system, there are full rate (FR) voice service and half rate (HR) voice service, FR is the first digital voice coding standard used in the GSM digital mobile phone system, codec (codec ) has an average bit rate of 13kbps. HR works at 5.6kbps, which requires half the bandwidth of the full-rate codec, so the network capacity of the voice service can be doubled, of course, the audio quality will be reduced in this case. In the existing GSM network, when the network load is high, HR voice service can be switched to increase the voice service capacity.

For the TD-SCDMA system, with the increase of users, China Mobile Communications Corporation (CMCC) also hopes to adopt low-rate (such as AMR 4.75kbps) voice services to save Orthogonal Variable Spread Spectrum (OVSF) codes.

For example, for the high-rate (AMR 12.2kbps) voice service in the TD-SCDMA system, for each subframe, use 2×SF16 (two consecutive SF16 spreading codes) on the downlink (DL), and 2×SF16 on the uplink On the road (UL), use 1×SF8 (one SF8 spreading code).

In order to achieve low-rate voice services, it is recommended to use 1×SF16 (one SF16 spreading code) for each subframe on DL and UL to double the voice service capacity, which is called Code Division Multiplexing (CDM) method .

With the CDM method, 1×SF16 code is allocated to each user equipment (UE), so Midamble K=16 needs to be used to support up to 16 users in one DL or UL time slot.

Compared with Midamble K=8 (the default setting in existing networks), Midamble K=16 has poor correlation performance and thus affects joint detection. Moreover, when Midamble K=16, because the length of the detection window is reduced from 16 chips to 8 chips, the peak value of the channel impulse response (CIR) is easy to slip out of the detection window in the complex wireless environment, as shown in the figure 1.

In Fig. 1, when the detection window length is 16 chips (K=8), more side peaks can be included in the detection window without making these side peaks change in the wireless environment or wrong SS adjustment , slide out of the window.

Therefore, it is hoped that in the TD-SCDMA system, Midamble K=8 can still be used to realize low-rate voice services.

Contents of the invention

In view of the above-mentioned problems in the prior art, the present invention is proposed, and the present invention is committed to providing a method and system for carrying adaptive voice services based on frame division multiplexing (TDM) in TD-SCDMA, a radio network controller and a user equipment. Especially when carrying low rate (such as AMR4.75kbps) voice services, Midamble K=8 can still be used.

According to the first aspect of the present invention, there is provided a method for carrying adaptive voice traffic based on frame division multiplexing in a TD-SCDMA system, the method comprising the following steps: at the radio network controller, according to the voice traffic load of the network selecting a speech codec rate and allocating radio resources to user equipment based on radio frames; Transmission and/or reception of signals.

According to a second aspect of the present invention, a radio network controller used in a TD-SCDMA system is provided, the radio network controller includes: a voice traffic load monitoring part, which is set to The voice service load is monitored; the voice codec rate selection part, the voice codec rate selection part is set to, according to the monitoring results of the voice service load monitoring part, select the voice codec rate group suitable for the preset voice codec rate group The speech codec rate of the monitoring result, wherein, the preset speech codec rate group includes at least the first high speech codec rate and the second low speech codec rate; the radio resource allocation unit, the radio resource allocation unit is set to Allocating radio resources to one or more user equipments in units of one or more radio frames according to the speech codec rate selected by the speech codec rate selection section; and a notification section configured to Notifying the user equipment of the speech codec rate selected by the speech codec rate selection unit and the result of radio resource allocation performed by the radio resource allocation unit.

According to a third aspect of the present invention, a user equipment used in a TD-SCDMA system is provided, the user equipment includes a speech codec unit, a buffer and rate matching unit, and a transceiver unit, wherein the speech codec unit is controlled by It is set to codec the voice signal according to the voice codec rate selected by the system; the buffer and rate matching part is set to buffer the voice signal encoded by the voice codec part and according to the voice transmission rate of the network Perform rate matching processing, then send the voice signal processed by the rate matching process to the transceiver part, and perform rate matching processing on the voice signal received from the transceiver part and send the voice signal after the rate matching process to the speech codec part; the transceiver part is set to send the speech signal processed by the buffer and rate matching part within the frame offset period determined based on the frame offset value assigned by the system, and/or The voice signal sent from the network is received within the frame offset period determined based on the frame offset knowledge allocated by the system, and the voice signal is sent to the buffering and rate matching part.

According to the fourth aspect of the present invention, there is provided an adaptive voice service bearer system based on frame division multiplexing in a TD-SCDMA system, which includes a radio network controller and user equipment, wherein the radio network controller is controlled by It is set to select the voice codec rate according to the voice traffic load of the network and allocate radio resources to the user equipment based on the radio frame, and the user equipment is set to be based on the voice codec rate selected by the radio network controller and the radio frame-based The allocated wireless resources are used to send and/or receive voice signals.

According to the present invention, when the voice service load in the network is low, a high voice codec rate can be selected, and wireless resources can be allocated to the user equipment in the form of continuous wireless frames; and when the voice service load in the network is high, it can A low speech codec rate is selected, and the same radio resource is allocated to at least two user equipments in units of at least one continuous radio frame, so as to realize adaptive speech service bearer based on frame division multiplexing.

According to the present invention, low-rate voice services can be carried based on frame division multiplexing. Compared with the traditional CDM method, the low-rate voice service bearer method based on frame division multiplexing can still use Midamble K=8, so that good detection performance can be obtained due to its longer detection window, and at the same time it can be used in a multi-path wireless environment for better performance.

Moreover, by adopting the low-rate voice service bearing method based on frame division multiplexing, the user equipment can use discontinuous transmission (DTX) and discontinuous reception (DRX) more frequently. In this way, the user equipment only needs to transmit or receive on some radio frames instead of all radio frames. Therefore, the user equipment can save more power.

Description of drawings

The accompanying drawings are included to provide a further understanding of the invention, constitute a part of the specification, illustrate embodiments of the invention, and together with the text description, serve to explain the principle of the invention, wherein the same elements are used throughout are indicated with the same reference numerals. In the attached picture:

Fig. 1 is a comparison diagram of the peak value of the channel impulse response when the detection window length is 8 chips and 16 chips respectively;

FIG. 2 is a diagram illustrating an implementation example of a user plane according to one embodiment of the present invention;

FIG. 3 is a diagram illustrating an implementation example of a control plane according to one embodiment of the present invention;

FIG. 4 is a diagram illustrating a transmission channel configuration according to one embodiment of the present invention;

FIG. 5 is a diagram illustrating a physical channel configuration according to one embodiment of the present invention;

FIG. 6 is a comparison diagram illustrating the physical channel configuration when the traditional CDM method is used to carry low-rate voice services and the TDM method is used to carry low-rate voice services according to an embodiment of the present invention;

FIG. 7 is a diagram illustrating a network structure of a TD-SCDMA system related to the present invention;

FIG. 8 is a functional block diagram illustrating the composition of a user equipment (UE) according to one embodiment of the present invention;

9 is a functional block diagram illustrating the composition of a radio network controller (RNC) according to one embodiment of the present invention;

FIG. 10 is an overall flowchart illustrating an adaptive voice service bearing method according to an embodiment of the present invention;

FIG. 11 is a flow chart illustrating an example of the processing procedure of the adaptive voice service bearing method at the RNC according to an embodiment of the present invention; and

FIG. 12A and FIG. 12B are flow charts respectively illustrating an example of the processing procedure of the adaptive voice service bearing method at the UE according to an embodiment of the present invention.

Detailed ways

These and other aspects of the invention will become apparent with reference to the following description and drawings. In these descriptions and drawings, specific embodiments of the present invention are specifically disclosed to show some ways of implementing the principles of the present invention, but it should be understood that the scope of the present invention is not limited thereto. On the contrary, the invention includes all changes, modifications and equivalents coming within the spirit and scope of the appended claims.

Features described and/or exemplified for one embodiment can be used in the same or similar manner in one or more other embodiments, and/or in combination with or instead of features of other embodiments use.

It should be emphasized that the term "comprising" when used in this specification is used to refer to the presence of stated features, elements, steps or components, but does not exclude one or more other features, elements, steps, components or The presence or increase of their combinations.

Many aspects of the invention will be better understood with reference to the following drawings. Components and features shown in one figure or embodiment of the invention may be combined with components and features shown in one or more other figures or embodiments. Furthermore, in the drawings, like reference numerals designate corresponding parts throughout the figures, and may be used to designate the same or similar parts in more than one embodiment.

First, the principle of implementing low-rate (eg AMR 4.75kbps) voice services in a TD-SCDMA system by using a frame division multiplexing (TDM) method according to an embodiment of the present invention will be described with reference to FIGS. 2-6.

In the TD-SCDMA system, at least two voice services such as AMR 12.2kbps (high) and AMR 4.75kbps (low) can be realized.

When realizing the voice service of AMR 4.75kbps, it is hoped to still adopt the same code resources as the voice service of AMR 12.2kbps, that is, for the downlink (DL), use two consecutive SF16 codes, and for the uplink (UL ), use a SF8 yard, can continue to use the same Midamble K=8 like this.

The calculation shows that the transmission bandwidth of AMR 4.75kbps voice service allows wireless resource allocation in units of two virtual resource units (VRU). Multiple (for example, two) user equipments (UEs) share the same radio resource in different radio frames, that is, the same carrier frequency, the same time slot, and the same channel code. In this way, the system can accommodate more voice service users.

Fig. 2 illustrates an implementation example of a user plane (UP) according to one embodiment of the present invention.

At the Iu UP layer, no matter what voice service bearer method is adopted, the protocol data unit (PDU) will be transmitted according to the Iu timing interval (ITI) of 20ms. A PDU is composed of a RAB Subflow Combination Indicator (RFCI) and a Service Data Unit (SDU), ie, PDU=RFCI+SDU.

For voice traffic, the Radio Link Control (RLC) layer is transparent. However, buffering and rate matching functions need to be completed at the media access control (MAC) layer.

For example, for the sub-flow #1 and sub-flow #2 of the AMR 4.75kbps voice service, the MAC entity needs to complete the following buffering and rate matching functions:

When a packet is received from the lower layer (Frame Protocol (FP) layer), deliver SDUs to the upper layer at intervals of 20ms (this is equal to the ITI of the Iu UP layer);

• When sending a packet to the FP layer within its own time slice (repetition length (RL)), double the amount of data is sent.

In this way, the change from the conventional CDM method to the TDM method according to the invention is transparent to the Iu UP layer. The same physical resource can be allocated to two UEs with a radio frame of 10 ms as a unit, and the two UEs can be switched between each other to realize TDM-based low-rate voice service bearer.

FIG. 3 is a diagram illustrating an implementation example of a control plane according to one embodiment of the present invention.

The implementation of the control plane is similar to the implementation of the user plane, the difference is that 4 signaling radio bearers (SRBs) are multiplexed at the MAC layer, of which two SRBs are used for the radio resource control layer (RRC), and the other two SRBs are used for For the Network Adaptation Sublayer (NAS), these SRBs and their functions are well known in the art and will not be described in detail here. Considering the real-time nature of the voice service, it is necessary to shorten the TDM cycle (Repeat Period (RP)). For this reason, the transmission time interval (TTI) of the SRB of the dedicated channel (DCH) should be as small as possible, but the bandwidth of the SRB should still be maintained, for example, 1.7kbps or 3.4kbps.

In this way, keeping Midamble K = 8 and the length of the detection window at 16 chips, through 1:2 TDM, the CS domain 4.75kbps voice + 3.4kbps SRB bearer is realized, so that the capacity of the system voice service can be doubled .

FIG. 4 is a diagram illustrating a transmission channel configuration according to one embodiment of the present invention.

FIG. 5 is a diagram illustrating a physical channel configuration according to one embodiment of the present invention.

FIG. 6 is a diagram illustrating the comparison of physical channel configurations when the traditional CDM method is used to carry low-rate voice services and the TDM method is used to carry low-rate voice services according to an embodiment of the present invention. It can be seen from the figure that according to the present invention, code resources can be saved.

Considering the real-time characteristics of the voice service, the RP should be set as small as possible, for example, it can be set to 40ms. Specifically, transmission and physical channel parameters may be defined as follows, for example:

●RP=40ms, repetition length (RL)=20ms;

●Voice substream #1: TTI=10ms, TB size (transport block size)=42 or 39 bits;

●Voice substream #2: TTI=10ms, TB size (transport block size)=53 bits;

● SRB: TTI = 20ms, TB size = 148 bits.

For the MAC-d entity of the SRB, 1×148 bits will be sent within its own 20ms RL, so the average bit rate of the SRB is 3.4kbps. Using a 3.4kbps SRB instead of a 1.7kbps SRB can reduce delay in the control plane. In most cases, no data is transmitted in the SRB, so the voice radio bearer can use all the radio bandwidth.

Table 1 shows the configuration of physical channel parameters adopted for low-rate voice services according to an embodiment of the present invention.

Physical channel parameters DPCH uplink DPCH downlink modulation QPSK QPSK code and slot/radio frame SF8×1 code×2 time slots SF16×2 codes×2 time slots Maximum number of data bits/wireless frame 328 bits 328 bits TFCI code word/wireless frame 16 bits 16 bits TPC/wireless frame 2×2 bits 2×2 bits SS/wireless frame 2×2 bits 2×2 bits RP 4 4 RL 2 2 Punch limit 0.44 0.44

If the segmentation function of MAC-d is supported, a smaller TB can be used for signaling, for example, 148 bits/2=74 bits, or 148 bits/4=37 bits, so that a smaller TTI can be used for SRB, and can be used Smaller RL. If the TTI of SRB and RB is 10ms, RP is 20ms, and RL is 10ms. Then the voice traffic will have no delay.

The implementation manner of the present invention will be further described below in conjunction with an implementation example in a TD-SCDMA network.

Figure 7 shows a typical configuration example of a TD-SCDMA network. As shown in the figure, a typical TD-SCDMA network includes a User Equipment (UE) domain 100 , a Radio Network Subsystem (RNS) domain 200 and a Core Network (CN) domain 300 . The UE domain 100 includes at least one UE. The figure schematically shows two user equipments UE1 and UE2, but it should be understood that the number of UEs that can be accommodated in the system is not limited to two. RNS domain 200 includes NodeB 210 and radio network controller (RNC) 220. NodeB 210 communicates with UE domain 100 through an air interface, and connects with RNC 220 through an Iub interface. The connection between the RNS domain 200 and the CN domain 300 and the composition of the CN domain 300 are not directly related to the present invention, and will not be described in detail here.

The present invention will be described in more detail below focusing on the configuration examples of UE 100 and RNC 220 and the processing procedures executed in them.

According to one embodiment of the present invention, RNC 220 can monitor the voice traffic load in the network (for example, the number of users who initiate voice calls at the same time in the cell), and select from a preset voice codec rate group (for example, including AMR 12.2kbps and AMR4.75kbps), select a suitable speech codec rate, and allocate radio resources (frequency, code channel, time slot) for UE 100 based on the radio frame according to the selected speech codec rate, so that it can accommodate as much as possible There may be as many users as possible for voice communication. The RNC 220 can notify the UE 100 of the selected voice codec rate and the corresponding radio resource allocation result, and the UE 100 can perform voice service communication accordingly. Note that the notation "100" is used here to refer to both the UE domain and the UE.

FIG. 8 is a schematic block diagram illustrating a configuration example of UE 100 according to one embodiment of the present invention, and FIG. 9 is a schematic block diagram illustrating a configuration example of RNC 220 according to one embodiment of the present invention.

As shown in FIG. 8, UE 100 may include a speech codec unit 110, a buffer and rate matching unit 120, and a transceiver unit 130.

As shown in FIG. 9 , the RNC 220 may include a voice traffic load monitoring unit 2210, a voice codec rate selection unit 2220, a radio resource allocation unit 2230, and a notification unit 2240.

In the RNC 220, when the voice traffic load monitoring section 2210 detects that the voice traffic load in the network is low, for example, the number of users simultaneously initiating a voice call does not exceed the predetermined number of users, the voice codec rate selection section 2220 As a result, a high speech codec rate can be selected from a preset set of speech codec rates, eg AMR 12.2kbps, which corresponds to the conventional case. And when the voice traffic load monitoring section 2210 detects that the voice traffic load in the network is relatively high, for example, the number of users who initiate voice calls at the same time exceeds a predetermined number of users, the voice codec rate selection section 2220 can select from Select a low voice codec rate in the preset voice codec rate group, such as AMR 4.75kbps, which corresponds to the low-rate voice service bearer described above.

The radio resource allocation unit 2230 allocates radio resources to one or more UEs in units of one or more radio frames according to the speech codec rate selected by the speech codec rate selection unit 2220 . For example, when the voice codec rate selection unit 2220 selects, for example, AMR12.2kbps, the radio resource allocation unit 2230 allocates radio resources to each UE in the form of continuous radio frames, that is, in each radio frame, allocates to each UE The wireless resources are all different. This corresponds to the conventional case, but can also be considered a special case of the TDM method of the present invention. When the voice codec rate selection unit 2220 has selected, for example, AMR 4.75kbps, the radio resource allocation unit 2230 can, for example, set the RP and RL of the code channel to 40 ms and 20 ms respectively, to set two continuous radio frames as units of two UEs are allocated the same radio resources, but the frame offset values (offset) of the two UEs are different, one is 0 and the other is 2. In this way, the two UEs can alternately use the same radio resource to send and receive voice services in units of two radio frames. The RP, RL, and offset values listed here are just examples. For example, as mentioned above, you can also set RP and RL to 20ms and 10ms respectively, and allocate the same offset to two or more UEs in units of one radio frame. Wireless resources, so as to realize voice service without delay.

The notification unit 2240 may notify the UE 100 of the speech codec rate selected by the speech codec rate selection unit 2220 and the result of radio resource allocation performed by the radio resource allocation unit 2230.

In the UE 100, the speech codec unit 110 can codec the speech signal according to the speech codec rate notified by the notification unit 2240 in the RNC 220.

The buffering and rate matching unit 120 can buffer the voice signal coded by the voice codec unit 110 , perform rate matching processing according to the voice transmission rate of the network, and then send the rate matching voice signal to the transceiver unit 130 . The buffering and rate matching section 120 can also perform rate-removing processing on the voice signal received from the transceiver section 130 and send the voice signal after the rate-matching processing to the voice codec section 110, so that the voice codec section 110 can perform the rate-matching process according to the The corresponding speech codec rate is decoded.

For the case where the speech codec rate is, for example, AMR 12.2kbps, the buffering and rate matching processing of the buffering and rate matching unit 120 is the same as the conventional case, and will not be described in detail.

For example, when the voice codec rate is AMR 4.75kbps, when sending a signal, for example, the buffering and rate matching part 120 can first buffer the voice signal of 40ms (corresponding to the RP allocated by the system), and perform the rate according to the voice transmission rate of the network. The matching process enables the 40ms voice signal to be sent out within the 20ms RL (that is, two radio frames). When receiving a signal, the buffering and rate matching unit 120 performs a de-rate matching process opposite to the above-mentioned rate matching process on the signal received by the transceiver unit 130 within 20 ms of RL (that is, two wireless frames), and restores it to The speech signal of 40ms is then sent to the speech codec unit 110 for decoding.

The transceiver unit 130 may transmit the voice signal processed by the buffering and rate matching unit 120 within a frame offset period determined based on a frame offset value (for example, 0 or 2) assigned by the system (that is, the RNC 220), and receive The voice signal sent from the network is sent to the buffering and rate matching unit 120 for de-rate matching processing.

It should be noted that the speech codec rates listed above are AMR 12.2kbps or AMR 4.75kbps and RP and RL are 40ms and 20ms respectively are just examples. Other appropriate speech codec rates and RP, RL can also be selected according to the situation. RL value.

The following describes an adaptive voice service bearing method according to an embodiment of the present invention with reference to FIG. 10 to FIG. 12B .

Fig. 10 is an overall flow chart, at step S1000 place, RNC 220 selects suitable speech codec rate (for example, AMR 12.2kbps and AMR4.75kbps) according to the speech traffic load in the network, and allocate radio resource for UE based on radio frame . At step S2000. UE 100 transmits and receives voice services according to the voice codec rate selected by RNC 220 and the allocated radio resources.

FIG. 11 further illustrates the processing procedure of S1000 in the figure in detail.

At step S1010, the RNC 220 monitors the voice traffic load in the network, such as the number of users simultaneously initiating voice calls.

At step S1020, the RNC 220 judges whether the current voice traffic load is higher than a predetermined threshold, for example, judges whether the number of users simultaneously initiating voice calls exceeds a preset number of users. When the judgment result is "yes", the process goes to step S1030, and the low-rate voice service bearer process branch is executed; and when the judgment result is "no", the process goes to step S1030', and the high-rate voice service bearer process branch is executed.

In step S1030, because the current voice traffic load is relatively high, the RNC 220 selects a low voice codec rate, such as AMR 4.75kbps, from the preset voice codec rate group. Next, in step S1040, for example, the RP and RL of the code channel can be set to 40ms and 20ms respectively, and the same radio resource is allocated to the two UEs 100 in units of two consecutive radio frames, and the offset of the two UEs 100 The values are set to 0 and 2, respectively. The RP, RL, and offset values listed here are only examples. For example, as mentioned above, RP and RL can also be set to 20ms and 10ms respectively, and two or more UEs 100 can be assigned the same Wireless resources, so as to achieve voice services without delay.

In step S1030', because the current voice traffic load is low, the RNC 220 selects a high voice codec rate, such as AMR 12.2kbps, from the preset voice codec rate group. Next, in step S1040', radio resources are allocated to each UE 100 in the form of continuous radio frames. That is, in each radio frame, the radio resources allocated to each UE 100 are different. This corresponds to the conventional case.

Subsequently, in step S1050, RNC 220 notifies UE 100 of the selected speech codec rate and radio resource allocation result.

12A and 12B are flow charts showing the process of sending and receiving voice services performed at UE 100, respectively.

As shown in FIG. 12A, when sending a voice service, firstly, in step S2110, the voice signal is sampled and coded based on the received voice codec rate.

Next, in step S2120, the speech samples are buffered, for example, according to a certain period. If the speech codec rate is a high speech codec rate such as AMR 12.2kbps, for example, the speech samples may be buffered at a cycle of 20ms. If the speech codec rate is a low speech codec rate such as AMR 4.75kbps, for example, the speech samples may be buffered at a period of 40ms (corresponding to the RP allocated by the system).

Next, in step S2130, rate matching processing is performed on the buffered voice samples according to the voice transmission rate of the network. Regardless of the speech codec rate, after the rate matching process, the buffered speech samples can be sent out within a period of 20ms (two wireless frames).

Next, in step S2140, send rate-matched processed signal.

As shown in FIG. 12B, when receiving a voice service, first in step S2210, the UE 100 receives a voice signal within the frame offset period allocated to itself determined according to the frame offset value.

Next, in step S2220, the voice signal is buffered for eg 20 ms, and the de-rate matching process is performed according to the notified voice codec rate. For example, when the voice codec rate is such as the high voice codec rate of AMR 12.2kbps, after the rate matching process, the voice signal of 20ms is still obtained; and for the low voice codec rate such as AMR 4.75kbps In the case of the codec rate, after de-rate matching processing, a 40ms voice signal can be obtained.

Next, in step S2230, the voice signal that has undergone rate-matching processing is decoded according to the corresponding voice codec rate, so as to finally play it to the user through the playback device.

It should be noted that the speech codec rates listed above are AMR 12.2kbps or AMR 4.75kbps and RP and RL are 40ms and 20ms respectively are just examples. Other appropriate speech codec rates and RP, RL can also be selected according to the situation. RL value.

According to the present invention, when carrying low-rate voice services, the TDM method is used to replace the traditional CDM method, and Midamble K=8 can be used, thereby obtaining good detection performance due to its longer detection window, and at the same time in a multipath wireless environment for better performance.

Moreover, by adopting the TDM method, the UE can use DTX and DRX more frequently, and the UE only needs to transmit or receive in some radio frames instead of all radio frames. Therefore, when carrying low-rate voice services, the UE can save more power.

Although only the preferred embodiments have been chosen to illustrate the present invention, those skilled in the art can easily make various changes and modifications based on the disclosure herein without departing from the scope of the invention defined by the appended claims. The descriptions of the above embodiments are illustrative only, and do not constitute limitations on the invention defined by the appended claims and their equivalents.

It should be understood that each part of the present invention can be realized by hardware, software, firmware or a combination thereof. In the embodiments described above, various steps or methods may be implemented by software or firmware stored in memory and executed by a suitable instruction execution system. For example, if implemented in hardware, as in another embodiment, it can be implemented by any one or combination of the following techniques known in the art: Logic gates with logic functions for data signals Discrete logic circuits for circuits, ASICs with suitable combinational logic gates, programmable gate arrays (PGAs), field programmable gate arrays (FPGAs), etc.

Any process or method descriptions or blocks in flowcharts or otherwise described herein may be understood to represent a module comprising code of one or more executable instructions for implementing specific logical functions or steps in a process, fragments or portions, and the scope of the preferred embodiment of the invention includes additional implementations in which execution may be performed out of the order shown or discussed, including substantially concurrently or in reverse order depending on the functions involved functions, which should be understood by those skilled in the art to which the present invention pertains.

The logic and/or steps represented in the flowcharts or otherwise described herein, for example, can be considered as a sequenced listing of executable instructions for implementing the logical functions, can be embodied in any computer-readable medium, For use with instruction execution systems, devices, or devices (such as computer-based systems, systems including processors, or other systems that can fetch instructions from instruction execution systems, devices, or devices and execute instructions), or in conjunction with these instruction execution systems, devices or equipment used. For the purposes of this specification, a "computer-readable medium" may be any device that can contain, store, communicate, propagate or transmit a program for use in or in conjunction with an instruction execution system, device or device. A computer readable medium can be, for example, but not limited to, an electronic, magnetic, optical, electromagnetic, infrared, or semiconductor system, apparatus, device, or propagation medium. More specific examples (non-exhaustive list) of computer-readable media include the following: electrical connection with one or more wires (electronic device), portable computer disk case (magnetic device), random access memory (RAM) (electronic devices), read-only memory (ROM) (electronic devices), erasable programmable read-only memory (EPROM or flash memory) (electronic devices), optical fiber (optical devices), and portable compact disc read-only memory (CDROM) ) (optical device). In addition, the computer-readable medium may even be paper or other suitable medium on which the program can be printed, as it may be possible, for example, by optically scanning the paper or other medium, followed by editing, interpreting, or other suitable processing if necessary. The program is processed electronically and stored in computer memory.

The foregoing description and drawings illustrate various features of the invention. It should be understood that those skilled in the art can prepare appropriate computer codes to implement the various steps and processes described above and illustrated in the accompanying drawings. It should also be understood that the various terminals, computers, servers, networks, etc. described above may be of any type, and the computer codes may be prepared according to the disclosed content to realize the present invention by using the devices.

Specific embodiments of the invention are disclosed herein. Those of ordinary skill in the art will readily recognize that the present invention has other applications in other environments. In fact, many implementations and implementations exist. The appended claims are in no way intended to limit the scope of the invention to the specific embodiments described above. In addition, any reference to "means for" is intended to delineate elements and explain the means-plus-function of the claims, and any referenced elements that do not specifically use "means for" are not intended to be used A means-plus-function element is understood even if the claim includes the word "means".

While the invention has been shown and described with respect to a particular preferred embodiment or embodiments, it is obvious that equivalent modifications and alterations will occur to those skilled in the art upon the reading and understanding of the specification and drawings. Especially with respect to the various functions performed by the aforementioned elements (parts, components, means, compositions, etc.), unless otherwise indicated, terms used to describe these elements (including references to "means") are intended to correspond to the specific Any element that is functional (ie, functionally equivalent), even if that element is structurally different from the disclosed structure that performs that function in an illustrated example embodiment or embodiments of the invention. Additionally, while specific features of the invention have been described above with respect to only one or more of several illustrated embodiments, such features may be incorporated as needed and considered advantageous to any given or particular application. Combined with one or more other features of other embodiments.

Claims (15)

1. one kind is divided multiplexing adaptive voice service bearer method based on frame in the TD-SCDMA system, and this method may further comprise the steps:

At the radio network controller place; From preset encoding and decoding speech rate set, select encoding and decoding speech speed and be the user equipment allocation Radio Resource according to the speech business load of network based on radio frames; Wherein, Should preset encoding and decoding speech rate set comprise the first high encoding and decoding speech speed and the second low voice encoding and decoding speed at least; And under the situation of having selected the second low voice encoding and decoding speed, be that unit is respectively at least two Radio Resources that user equipment allocation is same with at least one continuous radio frames; With

At said subscriber equipment place,, carry out the transmission and/or the reception of voice signal based on the selected encoding and decoding speech speed of said radio network controller with based on radio frames institute assigned radio resource.

2. the method for claim 1, wherein at the radio network controller place, further comprise according to the speech business load selection encoding and decoding speech speed of network and based on the step of radio frames distributing radio resource:

Speech business load according to network; From preset encoding and decoding speech rate set, select to be suitable for the encoding and decoding speech speed of this speech business load; Wherein, this preset encoding and decoding speech rate set comprises the first high encoding and decoding speech speed and the second low voice encoding and decoding speed at least;

Coming with one or more radio frames according to selected encoding and decoding speech speed is that unit is one or more user equipment allocation Radio Resource; And

Give said one or more subscriber equipment with selected encoding and decoding speech speed and allocation of radio resources result notification.

3. method as claimed in claim 2; Wherein, At the subscriber equipment place,, carry out the transmission of voice signal and/or the step of reception and further comprise based on the selected encoding and decoding speech speed of said radio network controller with based on radio frames institute assigned radio resource:

Encoding and decoding speech speed based on received is sampled, is encoded voice signal; The speech sample that is obtained is cushioned and carries out rate-matched according to the voice transfer speed of network, and the frame offset value that in based on received allocation of radio resources result, comprises and sending in definite vertical shift period cushioned and through the speech sample of rate-matched; And/or

The frame offset value that in based on received allocation of radio resources result, comprises and receive the voice signal that sends from network in definite vertical shift period, and by received encoding and decoding speech speed this voice signal is decoded.

4. method as claimed in claim 3, wherein, at said subscriber equipment place, with the period that equates with the repetition period of the code channel that comprises among the received allocation of radio resources result be that unit cushions speech sample.

5. method as claimed in claim 4, wherein, the said step of encoding and decoding speech speed of selecting at the radio network controller place comprises:

If the speech business load of network is lower than the threshold value of regulation, then select the said first high encoding and decoding speech speed; With

If the speech business load of network is higher than the threshold value of regulation, then select the said second low voice encoding and decoding speed.

6. method as claimed in claim 5, wherein, said step of joining Radio Resource in radio network controller punishment comprises:

If in the step of said selection encoding and decoding speech speed, selected the said first high encoding and decoding speech speed, then the mode with continuous radio frames is each said user equipment allocation Radio Resource; With

If in the step of said selection encoding and decoding speech speed, selected the said second low voice encoding and decoding speed; Then repetition period and repeat length and the frame offset value through code channel is set suitably is that unit is respectively at least two Radio Resources that user equipment allocation is same with at least one continuous radio frames.

7. like each described method in the claim 2 to 6, wherein, the said first high encoding and decoding speech speed is AMR 12.2kbps, and the said second low voice encoding and decoding speed is AMR 4.75kbps.

8. radio network controller that in the TD-SCDMA system, uses, this radio network controller comprises:

Speech business load monitoring portion, this speech business load monitoring portion is set to the speech business load in the network is monitored;

Encoding and decoding speech rate selection portion; This encoding and decoding speech rate selection portion is set to the monitoring result according to said speech business load monitoring portion; From preset encoding and decoding speech rate set, select to be suitable for the encoding and decoding speech speed of this monitoring result; Wherein, this preset encoding and decoding speech rate set comprises the first high encoding and decoding speech speed and the second low voice encoding and decoding speed at least;

Allocation of radio resources portion; This allocation of radio resources portion is set to the encoding and decoding speech speed selected according to by said encoding and decoding speech rate selection portion; Coming with one or more radio frames is that unit is one or more user equipment allocation Radio Resource; Wherein, Selected in encoding and decoding speech rate selection portion under the situation of the second low voice encoding and decoding speed, it is that unit is respectively at least two Radio Resources that user equipment allocation is same that allocation of radio resources portion is set to at least one continuous radio frames; And

Notice portion, this notice portion are set to the said selected encoding and decoding speech speed of encoding and decoding speech rate selection portion and said allocation of radio resources portion are carried out the result notification of allocation of radio resources and give said subscriber equipment.

9. radio network controller as claimed in claim 8; Wherein, When said speech business load monitoring portion monitors speech business load in the network when being lower than the threshold value of regulation; The said first high encoding and decoding speech speed is selected by said encoding and decoding speech rate selection portion, and said allocation of radio resources portion is each said user equipment allocation Radio Resource with the mode of continuous radio frames

And monitor speech business load in the network when being higher than the threshold value of regulation when said speech business load monitoring portion; The said second low voice encoding and decoding speed is selected by said encoding and decoding speech rate selection portion; And repetition period and repeat length and the frame offset value of said allocation of radio resources portion through code channel is set suitably is that unit is at least two Radio Resources that user equipment allocation is same with at least one continuous radio frames.

10. like claim 8 or 9 described radio network controllers, wherein, the said first high encoding and decoding speech speed is AMR 12.2kbps, and the said second low voice encoding and decoding speed is AMR 4.75kbps.

11. a subscriber equipment that in the TD-SCDMA system, uses, this subscriber equipment comprise encoding and decoding speech portion, buffering and rate-matched portion and receiving and transmitting part, wherein,

Said encoding and decoding speech portion is set to according to the selected encoding and decoding speech speed of system voice signal carried out encoding and decoding,

Said buffering and rate-matched portion are set to cushioning through the coded voice signal of said encoding and decoding speech portion and carrying out rate-matched according to the voice transfer speed of network and handle; The voice signal that will pass through the rate-matched processing subsequently sends to said receiving and transmitting part; With the voice signal that receives from said receiving and transmitting part is gone rate-matched handle and will send to said encoding and decoding speech portion through removing voice signal after rate-matched is handled

Said receiving and transmitting part is set to sending in definite vertical shift period based on frame offset value that system distributed through the voice signal of said buffering with the processing of rate-matched portion; And/or receiving the voice signal that sends from network in definite vertical shift period based on frame offset value that system distributed; And this voice signal sent to said buffering and rate-matched portion

Wherein, said encoding and decoding speech speed is from the group that comprises the first high encoding and decoding speech speed and the second low voice encoding and decoding speed at least, to select,

And wherein, when the said first high encoding and decoding speech speed had been selected by system, said subscriber equipment was by the mode distributing radio resource with continuous radio frames; And when system has selected said second to hang down voice encoding and decoding speed; Said subscriber equipment has been assigned with repetition period and the repeat length and the frame offset value of the code channel of suitable setting, and is unit and the multiplexing same Radio Resource of at least one other subscriber equipment with at least one continuous radio frames.

12. subscriber equipment as claimed in claim 11, wherein, said buffering and rate-matched portion further are set to, and the period that equates with repetition period of the code channel that distributed with system is that unit cushions speech sample.

13. like each described subscriber equipment in the claim 11 to 12, wherein, the said first high encoding and decoding speech speed is that the AMR 12.2kbps and the said second low voice encoding and decoding speed are AMR 4.75kbps.

14. one kind is divided multiplexing adaptive voice service bearer system based on frame in the TD-SCDMA system, it comprises radio network controller and subscriber equipment,

Wherein, Said radio network controller is set to from preset encoding and decoding speech rate set, select encoding and decoding speech speed and be said user equipment allocation Radio Resource based on radio frames according to the speech business load of network; Wherein, Should preset encoding and decoding speech rate set comprise the first high encoding and decoding speech speed and the second low voice encoding and decoding speed at least; And said radio network controller is set under the situation of having selected the second low voice encoding and decoding speed, is that unit is respectively at least two Radio Resources that user equipment allocation is same with at least one continuous radio frames

And wherein, said subscriber equipment is set to carry out the transmission and/or the reception of voice signal based on the selected encoding and decoding speech speed of said radio network controller with based on radio frames institute assigned radio resource.

15. adaptive voice service bearer as claimed in claim 14 system; Wherein, Said radio network controller is each described radio network controller in the claim 8 to 10, and said subscriber equipment is each described subscriber equipment in the claim 11 to 13.

Images