CN102665242B - Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access - Google Patents
Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access Download PDFInfo
- Publication number
- CN102665242B CN102665242B CN201210169009.6A CN201210169009A CN102665242B CN 102665242 B CN102665242 B CN 102665242B CN 201210169009 A CN201210169009 A CN 201210169009A CN 102665242 B CN102665242 B CN 102665242B
- Authority
- CN
- China
- Prior art keywords
- candidate
- time slot
- encoding rate
- speech encoding
- access
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Expired - Fee Related
Links
- 238000000034 method Methods 0.000 title claims abstract description 27
- 230000003044 adaptive effect Effects 0.000 title claims abstract description 6
- 108010003272 Hyaluronate lyase Proteins 0.000 claims description 7
- 238000011144 upstream manufacturing Methods 0.000 claims description 7
- 230000008859 change Effects 0.000 description 2
- 238000005516 engineering process Methods 0.000 description 2
- 238000006467 substitution reaction Methods 0.000 description 2
- 101100480738 Arabidopsis thaliana TFCC gene Proteins 0.000 description 1
- 230000001174 ascending effect Effects 0.000 description 1
- 230000008901 benefit Effects 0.000 description 1
- 230000006872 improvement Effects 0.000 description 1
- 238000005259 measurement Methods 0.000 description 1
- 230000007246 mechanism Effects 0.000 description 1
- 238000010295 mobile communication Methods 0.000 description 1
- 230000004048 modification Effects 0.000 description 1
- 238000012986 modification Methods 0.000 description 1
- 230000008569 process Effects 0.000 description 1
- 230000011664 signaling Effects 0.000 description 1
Images
Classifications
-
- Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
- Y02—TECHNOLOGIES OR APPLICATIONS FOR MITIGATION OR ADAPTATION AGAINST CLIMATE CHANGE
- Y02D—CLIMATE CHANGE MITIGATION TECHNOLOGIES IN INFORMATION AND COMMUNICATION TECHNOLOGIES [ICT], I.E. INFORMATION AND COMMUNICATION TECHNOLOGIES AIMING AT THE REDUCTION OF THEIR OWN ENERGY USE
- Y02D30/00—Reducing energy consumption in communication networks
- Y02D30/70—Reducing energy consumption in communication networks in wireless communication networks
Landscapes
- Mobile Radio Communication Systems (AREA)
Abstract
The invention discloses a method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access. When UE (user equipment) accesses initially, the method includes: firstly, judging whether the UE can access in at least one alternate time slot at a first speech coding rate or not according to a first parameter, uploading signal interference rate (SIR), uploading interference signal code power (ISCP) of each alternate time slot and emission power of the UE in each alternate time slot, if yes, indicating that the UE can access in at least one alternate time slot by the first speech coding rate, if not, indicating that the UE can access at the second speed coding rate, wherein the first parameter is path loss from the base station to the UE or RSCP (received signal code power) of a PCCPCH (primary common control physical channel), and the first speech coding rate is higher than the second speech coding rate. The invention further provides equipment. By the method and the equipment for selecting AMR speech coding rate during initial access, access success rate can be increased, and network coverage and speech quality can be improved.
Description
Technical field
The present invention relates to mobile communication technology, particularly in a kind of initial when access, is selected the method and apparatus of AMR speech encoding rate.
Background technology
In TD-SCDMA system, adaptive multi-rate (AMR) audio coder & decoder (codec) is supported 8 provenance rate mode: 12.2kbps (GSM-EFR), 10.2kbps, 7.95kbps, 7.40kbps (IS-641), 6.70kbps (PDC-EFR), 5.90kbps, 5.15kbps and 4.75kbps altogether.
At present, system adopts the speed access UE of 12.2Kbps conventionally, at this moment, network side need to be the code channel that 1 spreading factor of each call distribution (SF) is 8, when UE is in the time that the access slot of weak overlay area or UE exists strong jamming, (SIR) is lower for user's signal-to-jamming ratio, easily call drop, access success rate is lower, and voice quality is also poor.
In prior art, there is a kind of method of the AMR of change speech encoding rate, the method is for up, after UE access, by setting up " transmitting power of UE is measured ", in the time that the transmitting power of UE exceedes certain thresholding, by transformat combination control (TFC Control) message, the AMR speech encoding rate of this UE is changed, change to low speech encoding rate pattern.But this method is applicable to the scene after UE access, cannot improve the access success rate of UE, and have the incompatible problem of UE.For example: on market, in existing mainstream chip producer, have several families not support up TFCC mechanism.
Summary of the invention
The method and apparatus of selecting AMR speech encoding rate while the invention provides a kind of initial access, to improve the access success rate of UE, and improves voice quality.
When one provided by the invention initially accesses, select the method for adaptive multi-rate (AMR) speech encoding rate, in the time that subscriber equipment (UE) initially accesses, the method comprises:
A, according to the uplink interference of the first parameter, upward signal interference ratio (SIR), each candidate's time slot and UE in the transmitting power of each candidate's time slot, judge whether UE can adopt the first speech encoding rate to access at least one candidate's time slot; Described the first parameter is that the path loss of UE or the received signal code power (RSCP) of Primary Common Control Physical Channel (PCCPCH) are arrived in base station;
If the result that B judges is yes, indicate described UE to adopt the first speech encoding rate to access at described at least one candidate's time slot, otherwise, indicate described UE to adopt the second speech encoding rate to access; Wherein, the first speech encoding rate is higher than the second speech encoding rate.
Preferably, described A comprises:
According to the measured value of the uplink interference signal code power (ISCP) of the measured value of the first parameter, the up target SIR setting in advance and each candidate's time slot, estimation UE is in the expection transmitting power of each candidate's time slot;
If UE all exceedes default power threshold in the expection transmitting power of all candidate's time slots, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default power threshold in expection transmitting power adopts the first speech encoding rate to access.
Preferably, the method further comprises: set in advance the expection transmitting power interval that each speech encoding rate is corresponding, if UE all exceedes default thresholding in the expection transmitting power of all candidate's time slots,, according to the residing interval of expection transmitting power of UE, adopt corresponding speech encoding rate as described the second speech encoding rate.
Preferably, described A comprises:
Transmitting power according to the measured value of the first parameter, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the up ISCP of each candidate's time slot;
If the up ISCP of all candidate's time slots all exceedes default ISCP thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default ISCP thresholding at up ISCP adopts the first speech encoding rate to access.
Preferably, described A comprises:
Transmitting power according to the measured value of the up ISCP of the measured value of the first parameter, each candidate's time slot and the UE that sets in advance at each candidate's time slot, estimates the up SIR of each candidate's time slot;
If the up SIR of all candidate's time slots all fails to reach default SIR thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can reach default SIR thresholding at up SIR adopts the first speech encoding rate to access.
Preferably, described A comprises:
In the transmitting power of each candidate's time slot, estimate that the base station of each candidate's time slot is to the path loss of UE according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance; If the path loss of all candidate's time slots all exceedes default line loss threshold, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default line loss threshold in path loss adopts the first speech encoding rate to access;
Or the transmitting power according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the PCCPCHRSCP of each candidate's time slot; If the PCCPCH RSCP of all candidate's time slots does not all reach default thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can exceed default thresholding at PCCPCH RSCP adopts the first speech encoding rate to access.
Preferably, carry out described judgement according to formula (1) or formula (2):
Power_of_UE
n＝L
PCCPCH+SIR+ISCP
n+C1 （1）
Power_of_UE
n=SIR+ISCP
n-PCCPCH RSCP+C2 （2）
Wherein:
Power_of_UE
nrepresent the transmitting power of UE at candidate's time slot n;
L
pCCPCHthe path loss that represents PCCPCH channel, unit is dB;
SIR represents up SIR;
ISCP
nrepresent the uplink interference of candidate's time slot n;
PCCPCH RSCP represents the RSCP of PCCPCH;
C1 is constant, comprises uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor;
C2 is constant, the transmitting power, uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor that comprise PCCPCH.
A kind of equipment provided by the invention, comprising:
Judge module, for according to the uplink interference of the first parameter, upward signal interference ratio (SIR), each candidate's time slot and subscriber equipment (UE) in the transmitting power of each candidate's time slot, judge whether UE can adopt the first speech encoding rate to access at least one candidate's time slot; Described the first parameter is that the path loss of UE or the received signal code power (RSCP) of Primary Common Control Physical Channel (PCCPCH) are arrived in base station;
Speed is selected module, be used for according to the judged result of judge module, in the case of the result judging be, indicate described UE to adopt the first speech encoding rate to access at described at least one candidate's time slot, in judged result be no in the situation that, indicate described UE to adopt the second speech encoding rate to access, wherein, the first speech encoding rate is higher than the second speech encoding rate.
Preferably, described judge module, for according to the measured value of the uplink interference signal code power (ISCP) of the measured value of the first parameter, the up target SIR setting in advance and each candidate's time slot, is estimated the expection transmitting power of UE at each candidate's time slot; If UE all exceedes default power threshold in the expection transmitting power of all candidate's time slots, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default power threshold in expection transmitting power adopts the first speech encoding rate to access.
Preferably, in described equipment, can also comprise memory module;
Described memory module, for storing the speech encoding rate setting in advance and the corresponding relation of expecting transmitting power interval;
Described speed is selected module, for the corresponding relation of storing according to the residing interval of expection transmitting power and the memory module of UE, adopts corresponding speech encoding rate as described the second speech encoding rate.
Preferably, described judge module, for the transmitting power at each candidate's time slot according to the measured value of the first parameter, the up target SIR setting in advance and the UE that sets in advance, is estimated the up ISCP of each candidate's time slot; If the up ISCP of all candidate's time slots all exceedes default ISCP thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default ISCP thresholding at up ISCP adopts the first speech encoding rate to access.
Preferably, described judge module, for the transmitting power at each candidate's time slot according to the measured value of the up ISCP of the measured value of the first parameter, each candidate's time slot and the UE that sets in advance, is estimated the up SIR of each candidate's time slot; If the up SIR of all candidate's time slots all fails to reach default SIR thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can reach default SIR thresholding at up SIR adopts the first speech encoding rate to access.
Preferably, described judge module for according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance in the transmitting power of each candidate's time slot, estimate that the base station of each candidate's time slot is to the path loss of UE; If the path loss of all candidate's time slots all exceedes default line loss threshold, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default line loss threshold in path loss adopts the first speech encoding rate to access;
Or described judge module, for the transmitting power at each candidate's time slot according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance, is estimated the PCCPCH RSCP of each candidate's time slot; If the PCCPCH RSCP of all candidate's time slots does not all reach default thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can exceed default thresholding at PCCPCH RSCP adopts the first speech encoding rate to access.
As seen from the above technical solution, the present invention is by the time that UE initially accesses, path loss or PCCPCH RSCP according to base station to UE, and the up ISCP of up SIR, each candidate's time slot and UE are in the transmitting power of each candidate's time slot, judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot; Only in judged result for be in the situation that, just indicate this UE to adopt higher speech encoding rate to access at corresponding candidate's time slot, otherwise, indicate this UE to adopt lower speech encoding rate to access, thereby can improve SIR or obtain higher coding gain, promote access success rate, improve the network coverage and voice quality.
Technical scheme provided by the invention relates to four different parameters (wherein, path loss and PCCPCHRSCP be two can phase trans-substitution parameter, therefore, both are considered as to a parameter), in the time of practical application, can fix wherein any three, a remaining parameter of estimation accordingly, and whether meet default thresholding according to this parameter and judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot.For example: the measured value of path loss and the up ISCP of candidate's time slot that can arrive according to base station UE, be required to meet under the prerequisite of certain target SIR, estimate the expection transmitting power of UE at each candidate's time slot, and be that user selects suitable speech encoding rate to access according to the result of estimation.Visible, technical scheme provided by the invention is very flexible.
Accompanying drawing explanation
Fig. 1 selects the schematic flow sheet of the method for AMR speech encoding rate while being the better initial access of the present invention one;
Fig. 2 is the composition structural representation of network equipment of the present invention.
Embodiment
For making object of the present invention, technical scheme and advantage clearer, referring to the accompanying drawing embodiment that develops simultaneously, the present invention is described in further detail.
Prior art is not distinguished the coverage condition of user present position and the disturbed condition of access slot conventionally, selects regularly a certain speed to access UE.In the time that UE selects high speech encoding rate access, may cause the transmitting power of UE to be not enough to meet the SIR of system requirements, thereby have the risk of access failure or call drop, or after access, voice quality is poor.
Main thought of the present invention is: in the time that UE initially accesses, arrive the path loss of UE or the received signal code power (RSCP) of Primary Common Control Physical Channel (PCCPCH) according to base station, and the uplink interference signal code power (ISCP) of up SIR, each candidate's time slot and UE are in the transmitting power of each candidate's time slot, judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot; If the result judging is yes, indicate this UE to adopt higher speech encoding rate to access at described at least one candidate's time slot, otherwise, indicate this UE to adopt lower speech encoding rate to access.Because base station is to there being following relation between the path loss of UE and PCCPCH RSCP:
Base station is to transmitting power-PCCPCH RSCP of path loss=PCCPCH of UE
Therefore, in estimation when other parameters, path loss and PCCPCH RSCP be two can phase trans-substitution parameter, if no special instructions, the follow-up use of the present invention " the first parameter " representative " path loss or the PCCPCH RSCP of UE arrived in base station ".
The present invention, in the time judging whether UE can adopt higher speech encoding rate to access, relates to four different parameters: the ISCP of the first parameter, up SIR, each candidate's time slot and UE are in the transmitting power of each candidate's time slot.In the time of specific implementation, can fix wherein any three, a remaining parameter of estimation accordingly, and whether meet default thresholding according to this parameter and judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot.For example:
The first situation:
In the time that AMR user initially accesses, if network side (comprises core net, RNC etc.) and UE all support multiple AMR speech encoding rates (can expand to future and comprise above-mentioned 8 kinds of speed in other interior speed), so, network side can for example, according to the uplink interference situation of the measured value of the first parameter and candidate's time slot (: the up ISCP of this time slot), be required to meet under the prerequisite of certain target SIR, calculate UE needed expection transmitting power on each candidate's time slot, if UE all exceedes default power threshold in the expection transmitting power of all candidate's time slots, think that this access exists potential risk, network side indication UE is with lower speech encoding rate access, thereby obtain higher SIR or coding gain, improve access success rate to reach, strengthen the object that covers and improve voice quality.
The second situation:
Transmitting power according to the measured value of the first parameter, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the up ISCP of each candidate's time slot;
If the up ISCP of all candidate's time slots all exceedes default ISCP thresholding, judge that UE can not adopt higher speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default ISCP thresholding at up ISCP adopts higher speech encoding rate to access.
The third situation:
Transmitting power according to the measured value of the up ISCP of the measured value of the first parameter, each candidate's time slot and the UE that sets in advance at each candidate's time slot, estimates the up SIR of each candidate's time slot;
If the up SIR of all candidate's time slots all fails to reach default SIR thresholding, judge that UE can not adopt higher speech encoding rate to access, otherwise, judge that candidate's time slot that UE can reach default SIR thresholding at up SIR adopts higher speech encoding rate to access.
The 4th kind of situation:
In the transmitting power of each candidate's time slot, estimate that the base station of each candidate's time slot is to the path loss of UE according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance; If the path loss of all candidate's time slots all exceedes default line loss threshold, judge that UE can not adopt higher speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default line loss threshold in path loss adopts higher speech encoding rate to access;
Or the transmitting power according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the PCCPCHRSCP of each candidate's time slot; If the PCCPCH RSCP of all candidate's time slots does not all reach default thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can exceed default thresholding at PCCPCH RSCP adopts the first speech encoding rate to access.
Relation between above-mentioned four parameters can use formula (1) or formula (2) to represent:
Power_of_UE
n＝L
PCCPCH+SIR+ISCP
n+C1 （1）
Power_of_UE
n=SIR+ISCP
n-PCCPCH RSCP+C2 （2）
Wherein:
Power_of_UE
nrepresent the transmitting power of UE at candidate's time slot n;
L
pCCPCHthe path loss that represents PCCPCH channel, unit is dB;
SIR represents up SIR;
ISCP
nrepresent the uplink interference of candidate's time slot n;
PCCPCH RSCP represents the RSCP of PCCPCH;
C1 is constant, comprises uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor;
C2 is constant, the transmitting power, uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor that comprise PCCPCH.
For different situations, formula (1) or (2) are out of shape, can estimate corresponding parameter.
Take the first situation as example, referring to Fig. 1, when initially being accessed, the present invention select the flow process of the method for AMR speech encoding rate to describe below.Shown in Fig. 1, method comprises the following steps:
Step 101: according to formula (3), according to the ISCP value of candidate's time slot of predefined target SIR, measurement and path loss situation, the expection transmitting power of estimation UE in the time of each candidate's time slot access.
Power_of_UE_Expect
(time slot n)=SIR
init_Target+ ISCP
(time slot n)+ Margin
uL_Interference(3)
+ Gain
cUBF+ L
pCCPCH+ 10log (the request RU number of U on time slot n)-12
In formula (3):
SIR
init_Target: being the initial value of up target SIR, is configurable parameter;
ISCP
(time slot n): the measured value that is the Interference Signal Code Power (ISCP) of ascending time slot n;
Margin
uL_Interference: being uplink interference surplus, is configurable parameter;
Gain
cUBF: being upstream multi-antenna receiving gain offset, is the configurable parameter in backstage, for this parameter difference corresponding to different Path numbers;
RU: be Resource Unit;
12: be the coding gain of SF-16; Here, only for illustrating;
L
pCCPCH: be the path loss of Primary Common Control Physical Channel (PCCPCH) channel that UE measures, unit is dB, its computing formula as shown in Equation (4):
L
PCCPCH=PCCPCH
Power-PCCPCH
RSCP （4）
PCCPCH RSCP: for UE measures and the received signal code power value to the PCCPCH of RNC by signaling message report;
PCCPCH Power: be the transmitting power of the PCCPCH channel of backstage configuration.
Step 102: judge according to the result of calculation of step 101, in the time that the expection transmitting power of the UE calculating on all candidate's time slots all exceedes certain thresholding, think that UE exists strong jamming in weak overlay area or target time slot, there is the risk of access failure in UE, execution step 103, network side indication UE is with lower speech encoding rate access; Otherwise, execution step 104:, network side indication UE does not exceed candidate's time slot of default thresholding with higher speech encoding rate access in expection transmitting power.
In this step, in the time that indication UE adopts lower speech encoding rate access, network side can also specify this calling to be carried by single-code channel, also can adopt the carrying of dicode road, and the covering gain meeting that carrying code channel number difference causes is slightly variant.
Can set in advance the expection transmitting power interval that each speech encoding rate is corresponding, in step 103, in the time that judgement UE accesses with lower speech encoding rate, according to the residing interval of expection transmitting power of UE, adopt corresponding speech encoding rate.
Corresponding to said method, the invention provides a kind of network equipment as shown in Figure 2, this equipment comprises:
Speed is selected module 220, be used for according to the judged result of judge module 210, in the case of the result judging be, indicate this UE to adopt the first speech encoding rate to access at described at least one candidate's time slot, in judged result be no in the situation that, indicate this UE to adopt the second speech encoding rate to access, wherein, the first speech encoding rate is higher than the second speech encoding rate.
Wherein, judge module 210, in the time judging, also can carry out according to above-mentioned four kinds of situations.
For the first situation, shown in Fig. 2, in equipment, can further include memory module 230, for storing the speech encoding rate setting in advance and the corresponding relation of expecting transmitting power interval;
Now, speed is selected module 220, for the corresponding relation of storing according to the residing interval of expection transmitting power and the memory module of UE, adopts corresponding speech encoding rate as described the second speech encoding rate.
The present invention is described in detail below to lift two preferred embodiments:
Embodiment mono-:
Suppose that network side allows AMR to call out and adopts 12.2Kbps and two kinds of speed of 4.75Kbps (network side can be specified one or more speed), the target SIR of the initial access of backstage configuration AMR and the expection transmitting power thresholding of UE.
When calculating according to formula (3) while accessing on all candidate's time slots in community, when the expection transmitting power of UE all exceedes the expection transmitting power thresholding of UE, so, network side specifies UE to adopt the speed of 4.75Kbps to access; Otherwise network side still specifies UE to adopt the speed of 12.2Kbps to access.
Embodiment bis-:
Due to the Power_of_UE_Expect in formula (3)
(time slot n), SIR
init_Target, ISCP
(time slot n)and L
pCCPCHbe variable, setting certain several variable is wherein a fixed value, and the expected results of a variable that just can be left by judgement is used as the choice criteria of UE speech encoding rate.For example, by above-mentioned formula distortion, can obtain formula (5):
ISCP
(time slot n)=Power_of_UE_Expect
(time slot n)-SIR
init_Target-Margin
uL_Interference(5)
-Gain
cUBF-L
pCCPCH-10log (the request RU number of UE on time slot n)+12
Thereby can be equivalent to: for example, in the transmitting power certain (being set as 24dbm) of UE, and target SIR is set under 8 prerequisite, require the ISCP value of up target time slot lower than certain thresholding, if all up candidate's time slots all can not meet this requirement, UE just accesses with lower speech encoding rate so.
As seen from the above-described embodiment, the present invention is in the time that UE initially accesses, path loss or PCCPCH RSCP according to base station to UE, and the up ISCP of up SIR, each candidate's time slot and UE are in the transmitting power of each candidate's time slot, judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot; Only in judged result for be in the situation that, just indicate this UE to adopt higher speech encoding rate to access at corresponding candidate's time slot, otherwise, indicate this UE to adopt lower speech encoding rate to access, thereby can improve SIR or obtain higher coding gain, promote access success rate, improve the network coverage and voice quality.
And, the present invention can fix any three in four parameters, a remaining parameter of estimation accordingly, and whether meet default thresholding according to this parameter and judge whether UE can adopt higher speech encoding rate to access at least one candidate's time slot, therefore, technical scheme provided by the invention is very flexible.
The foregoing is only preferred embodiment of the present invention, in order to limit the present invention, within the spirit and principles in the present invention not all, any modification of making, be equal to replacement, improvement etc., within all should being included in the scope of protection of the invention.
Claims (13)
1. a method of selecting adaptive multi-rate (AMR) speech encoding rate while initially access, is characterized in that, in the time that subscriber equipment (UE) initially accesses, the method comprises:
A, the uplink interference of fixing the first parameter, upward signal interference ratio (SIR), each candidate's time slot and UE any three parameters in the transmitting power of each candidate's time slot, the parameter that estimation is remaining, and whether meet default thresholding according to the parameter of estimation and judge whether UE can adopt the first speech encoding rate to access at least one candidate's time slot; Described the first parameter is that the path loss of UE or the received signal code power (RSCP) of Primary Common Control Physical Channel (PCCPCH) are arrived in base station;
If the result that B judges is yes, indicate described UE to adopt the first speech encoding rate to access at described at least one candidate's time slot, otherwise, indicate described UE to adopt the second speech encoding rate to access; Wherein, the first speech encoding rate is higher than the second speech encoding rate.
2. method according to claim 1, is characterized in that, described A comprises:
According to the measured value of the uplink interference signal code power (ISCP) of the measured value of the first parameter, the up target SIR setting in advance and each candidate's time slot, estimation UE is in the expection transmitting power of each candidate's time slot;
If UE all exceedes default power threshold in the expection transmitting power of all candidate's time slots, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default power threshold in expection transmitting power adopts the first speech encoding rate to access.
3. method according to claim 2, is characterized in that:
The method further comprises: set in advance the expection transmitting power interval that each speech encoding rate is corresponding, if UE all exceedes default thresholding in the expection transmitting power of all candidate's time slots,, according to the residing interval of expection transmitting power of UE, adopt corresponding speech encoding rate as described the second speech encoding rate.
4. method according to claim 1, is characterized in that, described A comprises:
Transmitting power according to the measured value of the first parameter, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the up ISCP of each candidate's time slot;
If the up ISCP of all candidate's time slots all exceedes default ISCP thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default ISCP thresholding at up ISCP adopts the first speech encoding rate to access.
5. method according to claim 1, is characterized in that, described A comprises:
Transmitting power according to the measured value of the up ISCP of the measured value of the first parameter, each candidate's time slot and the UE that sets in advance at each candidate's time slot, estimates the up SIR of each candidate's time slot;
If the up SIR of all candidate's time slots all fails to reach default SIR thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can reach default SIR thresholding at up SIR adopts the first speech encoding rate to access.
6. method according to claim 1, is characterized in that, described A comprises:
In the transmitting power of each candidate's time slot, estimate that the base station of each candidate's time slot is to the path loss of UE according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance; If the path loss of all candidate's time slots all exceedes default line loss threshold, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default line loss threshold in path loss adopts the first speech encoding rate to access;
Or the transmitting power according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance at each candidate's time slot, estimates the PCCPCHRSCP of each candidate's time slot; If the PCCPCH RSCP of all candidate's time slots does not all reach default thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can exceed default thresholding at PCCPCH RSCP adopts the first speech encoding rate to access.
7. according to the method described in claim 1 to 6 any one, it is characterized in that, carry out described judgement according to formula (1) or formula (2):
Power_of_UE
n=L
PCCPCH+SIR+ISCP
n+C1 （1）
Power_of_UE
n=SIR+ISCP
n-PCCPCHRSCP+C2 （2）
Wherein:
Power_of_UE
nrepresent the transmitting power of UE at candidate's time slot n;
L
pCCPCHthe path loss that represents PCCPCH channel, unit is dB;
SIR represents up SIR;
ISCP
nrepresent the uplink interference of candidate's time slot n;
PCCPCHRSCP represents the RSCP of PCCPCH;
C1 is constant, comprises uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor;
C2 is constant, the transmitting power, uplink interference surplus, upstream multi-antenna receiving gain offset and the coding gain corresponding to corresponding spreading factor that comprise PCCPCH.
8. an equipment of selecting adaptive multi-rate (AMR) speech encoding rate while initially access, is characterized in that, comprising:
Judge module, be used for fixing the uplink interference of the first parameter, upward signal interference ratio (SIR), each candidate's time slot and subscriber equipment (UE) any three parameters in the transmitting power of each candidate's time slot, the parameter that estimation is remaining, and judge according to the default thresholding that whether meets of estimating whether UE can adopt the first speech encoding rate to access at least one candidate's time slot; Described the first parameter is that the path loss of UE or the received signal code power (RSCP) of Primary Common Control Physical Channel (PCCPCH) are arrived in base station;
Speed is selected module, be used for according to the judged result of judge module, in the case of the result judging be, indicate described UE to adopt the first speech encoding rate to access at described at least one candidate's time slot, in judged result be no in the situation that, indicate described UE to adopt the second speech encoding rate to access, wherein, the first speech encoding rate is higher than the second speech encoding rate.
9. equipment according to claim 8, is characterized in that:
Described judge module, for according to the measured value of the uplink interference signal code power (ISCP) of the measured value of the first parameter, the up target SIR setting in advance and each candidate's time slot, is estimated the expection transmitting power of UE at each candidate's time slot; If UE all exceedes default power threshold in the expection transmitting power of all candidate's time slots, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default power threshold in expection transmitting power adopts the first speech encoding rate to access.
10. equipment according to claim 9, is characterized in that, also comprises memory module in described equipment;
Described memory module, for storing the speech encoding rate setting in advance and the corresponding relation of expecting transmitting power interval;
Described speed is selected module, for the corresponding relation of storing according to the residing interval of expection transmitting power and the memory module of UE, adopts corresponding speech encoding rate as described the second speech encoding rate.
11. equipment according to claim 8, is characterized in that:
Described judge module, for the transmitting power at each candidate's time slot according to the measured value of the first parameter, the up target SIR setting in advance and the UE that sets in advance, is estimated the up ISCP of each candidate's time slot; If the up ISCP of all candidate's time slots all exceedes default ISCP thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default ISCP thresholding at up ISCP adopts the first speech encoding rate to access.
12. equipment according to claim 8, is characterized in that:
Described judge module, for the transmitting power at each candidate's time slot according to the measured value of the up ISCP of the measured value of the first parameter, each candidate's time slot and the UE that sets in advance, is estimated the up SIR of each candidate's time slot; If the up SIR of all candidate's time slots all fails to reach default SIR thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can reach default SIR thresholding at up SIR adopts the first speech encoding rate to access.
13. equipment according to claim 8, is characterized in that:
Described judge module for according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance in the transmitting power of each candidate's time slot, estimate that the base station of each candidate's time slot is to the path loss of UE; If the path loss of all candidate's time slots all exceedes default line loss threshold, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can not exceed default line loss threshold in path loss adopts the first speech encoding rate to access;
Or described judge module, for the transmitting power at each candidate's time slot according to the measured value of the up ISCP of each candidate's time slot, the up target SIR setting in advance and the UE that sets in advance, is estimated the PCCPCH RSCP of each candidate's time slot; If the PCCPCH RSCP of all candidate's time slots does not all reach default thresholding, judge that UE can not adopt the first speech encoding rate to access, otherwise, judge that candidate's time slot that UE can exceed default thresholding at PCCPCH RSCP adopts the first speech encoding rate to access.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201210169009.6A CN102665242B (en) | 2012-05-28 | 2012-05-28 | Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201210169009.6A CN102665242B (en) | 2012-05-28 | 2012-05-28 | Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN102665242A CN102665242A (en) | 2012-09-12 |
| CN102665242B true CN102665242B (en) | 2014-07-09 |
Family
ID=46774630
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201210169009.6A Expired - Fee Related CN102665242B (en) | 2012-05-28 | 2012-05-28 | Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN102665242B (en) |
Families Citing this family (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2015139182A1 (en) * | 2014-03-17 | 2015-09-24 | 华为技术有限公司 | Ue, network side device, power adjustment method, and sg determination method |
| CN107241762A (en) * | 2016-03-29 | 2017-10-10 | 中兴通讯股份有限公司 | Coded system collocation method and device |
| CN113573233B (en) * | 2021-07-23 | 2022-06-28 | 荣耀终端有限公司 | Voice communication method, electronic device and readable medium |
Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2002054622A1 (en) * | 2001-01-02 | 2002-07-11 | Nokia Corporation | Method for open loop power control and bit rate selection |
| CN1728598A (en) * | 2004-07-31 | 2006-02-01 | 西门子（中国）有限公司 | Method for adjusting coding speed in procedure of voice mobile communication |
| CN101119145A (en) * | 2006-08-03 | 2008-02-06 | 普天信息技术研究院 | Auxiliary frequency point opened loop power control method in time division synchronous CDMA system |
| EP2139273A1 (en) * | 2006-04-12 | 2009-12-30 | InterDigital Technology Corporation | Apparatuses for radio resource control requested Codec Rate Control for Voip |
| CN102255686A (en) * | 2010-05-20 | 2011-11-23 | 中兴通讯股份有限公司 | Adaptive multi-rate codec (AMR) initial rate regulation method, device and system |
-
2012
- 2012-05-28 CN CN201210169009.6A patent/CN102665242B/en not_active Expired - Fee Related
Patent Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2002054622A1 (en) * | 2001-01-02 | 2002-07-11 | Nokia Corporation | Method for open loop power control and bit rate selection |
| CN1728598A (en) * | 2004-07-31 | 2006-02-01 | 西门子（中国）有限公司 | Method for adjusting coding speed in procedure of voice mobile communication |
| EP2139273A1 (en) * | 2006-04-12 | 2009-12-30 | InterDigital Technology Corporation | Apparatuses for radio resource control requested Codec Rate Control for Voip |
| CN101119145A (en) * | 2006-08-03 | 2008-02-06 | 普天信息技术研究院 | Auxiliary frequency point opened loop power control method in time division synchronous CDMA system |
| CN102255686A (en) * | 2010-05-20 | 2011-11-23 | 中兴通讯股份有限公司 | Adaptive multi-rate codec (AMR) initial rate regulation method, device and system |
Also Published As
| Publication number | Publication date |
|---|---|
| CN102665242A (en) | 2012-09-12 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| KR100265527B1 (en) | Power control method in wireless networks for communicating multiple information classes | |
| RU2486708C2 (en) | Channel adaptation method, base station and terminal in lte system | |
| US6535723B1 (en) | Method of power control for a wireless communication system having multiple information rates | |
| GB2462063A (en) | Method and apparatus for setting an uplink transmit power level for a wireless communication unit | |
| CN104023382B (en) | Poewr control method and base station in co-channel full duplex system | |
| CN103037488A (en) | Control method for long term evolution (LTE) uplink power and associated equipment thereof | |
| CN104469831B (en) | Fault detection method and device are shunk in the network coverage | |
| US20060293075A1 (en) | Method to improve outer loop power control | |
| CN102665242B (en) | Method and equipment for selecting AMR (adaptive multi-rate) speech coding rate during initial access | |
| CN101827386B (en) | Method, system and related device for adjusting transmitting power | |
| CN104811983A (en) | Adaptive modulation coding method and device | |
| CN100574134C (en) | The method and apparatus that in mobile radio system, is used for power control | |
| CN1505301A (en) | A speed control method in mobile communication system | |
| CN101094015B (en) | Method for controlling initial emission power in open loop in mobile communication system | |
| JP4163675B2 (en) | Apparatus and method for data rate transmission of a terminal in a mobile communication system | |
| WO2021171103A1 (en) | Cellular network indoor traffic auto-detection | |
| CN103379521B (en) | Trigger minimum road test and measure the method for configuration | |
| CN102196547B (en) | Special burst transmission method for time division-synchronous code division multiple access (TD-SCDMA) system | |
| CN113747556B (en) | Indoor distribution system energy saving method, equipment and computer readable storage medium | |
| CN103905173A (en) | Method and device for PDCCH self-adaptive transmission based on DTX and SINR | |
| CN104684034A (en) | Communication terminal management method and communication system | |
| CN104969632A (en) | Transmit power control systems, devices, and methods | |
| CN101990286B (en) | Status judgment method of transport format combination in dual-carrier high speed uplink packet access (HSUPA) system and device thereof | |
| CN1507191A (en) | Rate control method in mobile communication system | |
| CN100440749C (en) | Method for controlling self-adaptive down going multiple speed tate modes |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20140709 |