CN105812097A - Self-adaptive AMR code rate adjusting method based on network states - Google Patents
Self-adaptive AMR code rate adjusting method based on network states Download PDFInfo
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Abstract
The invention discloses a self-adaptive AMR code rate adjusting method based on network states. The method comprises following steps of 1, determining an initial code mode for a call sender; 2, writing a rate corresponding to the initial code mode of the sender in an RTP packet; 3, transmitting the RTP packet to a receiver; calculating the packet lost and delay parameter of the sender by the receiver; 4, judging whether the current initial code rate of the sender is applicable to the current network state of the receiver or not by the receiver according to the packet lost and the delay parameter; if the current initial code rate of the sender is applicable to the current network state of the receiver, keeping the current code rate; otherwise, changing a CMR (Codec Mode Request) value according to a rate adjusting strategy; 5, writing the rate corresponding to the adjusted CMR value in the RTP packet by the receiver; transmitting to the sender; calculating the packet lost and delay parameter by the sender in a similar way; judging according to the adjusting strategy; changing the CMR value; and adjusting the AMR code rate in real time. The method has the advantages that through real time adjustment of the code rate, constant changes of the network conditions are adapted, and therefore, the good call experience of a user is ensured.
Description
Technical field
The invention belongs to mobile Internet speech processes field, describe the AMR code rate self-adapting regulation method of a kind of state Network Based.
Background technology
Along with the life of development and the fusion of mobile communication technology and Internet technology, mobile information and people creates increasingly closer contacting, it is desirable to access the Internet whenever and wherever possible easily and obtain information and service.
By in by the end of August, 2015, the mobile interchange network users at existing 9.46 hundred million families of China, the sum of surfing Internet with cell phone, more than 900,000,000, accounts for the ratio of mobile phone user up to 69.5%.A large amount of application based on mobile Internet and business extend to the every aspect social life from amusement.Sustainable growth along with the development of network technology and mobile data traffic, voice calling service breaks through the restriction of conventional telecommunications telephone service, the direction of the networking telephone (VoiceoverInternetProtocol, VoIP) high towards call tone quality, rate are low is developed.But when network condition is unstable, the networking telephone based on mobile terminal still suffers from problem, and voice call quality can not well be ensured.
Self-adapting multi-rate narrowband voice coding (AdaptiveMultiRate-NarrowBandSpeechCodec, AMR-NB) algorithm is based on Code Excited Linear Prediction (CodeExcitedLinearPrediction, CELP) the speech coder standard of algorithm, it is mainly used in 3G (Third Generation) Moblie WCDMA (WidebandCodeDivisionMultipleAccess, W-CDMA) system.
The multiple coding mode that AMR coding provides can solve the rate allocation of source and channel coding better, certain coding mode can be selected adaptively to be transmitted according to channel conditions, make the distribution to Radio Resource and utilize more rationally, effectively, flexibly, meanwhile, experience of better conversing can also be provided the user.
The frame structure of AMR is as it is shown in figure 1, include 3 parts: AMR frame head, and AMR assists information and AMR core frames data；Wherein, AMR frame head includes the frame type of 4bits and the frame feature indicating bit of 1bit.AMR assists information for mode adaptive and error detection, specifically includes: the mode flags of 3bits, the mode request of 3bits and the coding cycle redundancy check of 8bits.AMR core frames data include voice or noise data, and the data of each of which frame are divided into: A/B/C tri-class field, and A class field is most important data in a frame, once A class field is damaged, whole frame just cannot decode.So, to use the method for various redundancy check that A class field portions data are protected by when being typically in being wirelessly transferred.Compared with A class field, B class field is relatively less important data；C class field is the data that entirety decoding impact is minimum.
AMR speech coder can be effectively improved voice quality, strengthens the ability of the anti-channel errors of system, adds power system capacity simultaneously, have the irreplaceable advantage of other voice coding modes.Meanwhile, AMR according to network condition, can select different coding modes, it is possible to better adapts to the change of network.
2015, the domestic network telephone software having had more than hundred sections, along with people's increase to the degree of dependence of smart mobile phone, the networking telephone also develops towards cell phone end, current more popular network telephone software, such as: wechat phone directory, micro-words, 360 freephones, cloud such as exhale at the version being also all proposed mobile terminal, to adapt to the communication requirement of people.Although the speech quality of the most mobile terminal networking telephone substantially and black phone suitable, but when network condition is unstable, still suffer from the problem that speech quality is unstable, it is impossible to provide, for people, experience of well conversing.
Summary of the invention
The present invention is directed to the problems referred to above and be analyzed research, audio communication system based on mobile Internet, adopt AMR coded system, different code rates is selected according to network condition, solve the impact that network condition instability is brought preferably, propose the AMR code rate self-adapting regulation method of a kind of state Network Based for this, specifically comprise the following steps that
Step one, for call sender, determine initial code pattern according to oneself residing current network type；
Current network type residing for sender includes: 2G, 3G, more than 3G or WiFi；
Code mode request CMR (CodecModeRequest), has 8 values, is followed successively by 0,1,2,3,4,5,6 and 7；
If what current sender connected is that 2G type network is conversed, then initial code pattern is CMR=0；
If the network type that current sender connects is 3G, more than 3G or WiFi, then initial code pattern is CMR=7；
Step 2, by corresponding for the initial code pattern of sender speed write RTP bag；
Speed corresponding to initial code pattern is followed successively by: 0 time speed of pattern is 4.75kbit/s；1 time speed of pattern is 5.15kbit/s；2 times speed of pattern are 5.90kbit/s；3 times speed of pattern are 6.70kbit/s；4 times speed of pattern are 7.40kbit/s；5 times speed of pattern are 7.95kbit/s；6 times speed of pattern are 10.2kbit/s；Under mode 7, speed is 12.2kbit/s.
Initial CMR has two states, and during CMR=0, AMR code rate is arranged on 4.75kbps；During CMR=7, AMR code rate is arranged on 12.2kbps.
Step 3, RTP bag are transferred to the receiving terminal of call, and receiving terminal calculates packet loss and the delay parameter of sender；
The packet loss cal_lost of RTP bag is calculated as follows:
Cal_lost=(packetlost/good*100)
Packetlost is packet loss quantity；Good is the RTP bag quantity of transmission.
Accumulative instantaneous time delay value cal_delay:
Cal_delay+=(jitter-250*mu)/8/mu；
Jitter is delay variation；Mu is sample rate, for the multiple of 8K；
Average delay avg_delay, arranges every about 1 second and judges once:
Avg_delay=cal_delay/timeCount
TimeCount is frequency of reckoning by time, and every 1 second is 1 time.；
According to delay variation statistical value, calculate final time delay Dtr:
Dtr(n+1)=α Dtr(n)+(1-α)Delay
Wherein DtrN () is the value of last Delay Variation, Delay is the instantaneous time delay value calculating gained.
Step 4, receiving terminal utilize packet loss and delay parameter, it is judged that the current initial code rate of sender, if appropriate for the current network state of receiving terminal, if be suitable for, keeps current code rate；Otherwise, CMR value is changed according to speed adjustable strategies；
Specifically comprise the following steps that
Step 401, according to time delay level, two delay threshold: Dth1 and Dth2 are set；
Step 402, judge whether time delay meets: Dtr< Dth1, if it is, enter step 403；Otherwise enter step 407；
Step 403, judge whether packet loss meets lost≤1%, if it is, enter step 404；Otherwise, step 405 is entered；
Step 404, judge whether CMR value is 7；If it is, keep CMR value constant；Otherwise, current CMR value is from increasing 1；
Step 405, determining whether whether packet loss meets: 1% < lost≤5%, if it is, enter step 406, otherwise current CMR value is set to 0；
Step 406, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 407, determine whether whether time delay meets Dth1≤Dtr≤ Dth2, if it is, enter step 408；Otherwise, current CMR value is set to 0；
Step 408, judge whether packet loss meets lost≤1%, if it is, enter step 409；Otherwise, step 410 is entered；
Step 409, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 410, judge that whether packet loss meets 1% < lost≤5%, if it is, enter step 411；Otherwise current CMR value is set to 0；
Step 411, judge that whether CMR value is be more than or equal to 2；If it is, current CMR value is from subtracting 2；Otherwise, current CMR value is set to 0；
The speed write RTP bag of the CMR value correspondence after adjustment is exported to sender by step 5, receiving terminal, and sender in like manner calculates packet loss and delay parameter, and judges by the adjustable strategies of step 4, changes CMR value and then real-time adjustment AMR code rate.
It is an advantage of the current invention that:
1, the AMR code rate self-adapting regulation method of a kind of state Network Based, by setting threshold value, makes full use of algorithm parameter few and realize simple characteristic, it is possible to fine must the change of network condition be reacted.
2, the AMR code rate self-adapting regulation method of a kind of state Network Based, by adjusting code rate in real time, thus ensureing the call experience that user is good.
Accompanying drawing explanation
Fig. 1 is the frame structure schematic diagram of AMR of the present invention；
Fig. 2 is AMR speech coding mode schematic diagram of the present invention；
Fig. 3 is the GSM of the present invention graph of a relation being operated under full-rate mode between MOS value and C/I value；
Fig. 4 is the GSM of the present invention graph of a relation being operated under half-rate mode between MOS value and C/I value；
Fig. 5 is AMR adaptive coding code check of the present invention switching schematic diagram；
Fig. 6 is the AMR code rate self-adapting regulation method flow chart of a kind of state Network Based of the present invention；
Fig. 7 is RTP bag form of the present invention and PayloadHeader form schematic diagram；
Fig. 8 is that the present invention judges the initial code rate method flow diagram if appropriate for current network state；
Fig. 9 is the interface display figure in communication process of the present invention.
Detailed description of the invention
Below in conjunction with accompanying drawing, the present invention is described in further detail.
In the existing mobile communication system of AMR, the principle of adaptation mechanism is substantially the same.It is analyzed for global system for mobile communications GSM (GlobalSystemforMobileCommunication)；
The coding mode adopted utilizes the AMR speech coding mode adopted in GSM as in figure 2 it is shown, divide into full rate (FullRate, FR) and two kinds of situations of half rate (HalfRate, HR).GSM first selects coding mode according to the value of carrier/interface ratio (Interference/Carrier, C/I) when selecting coded system；In the good situation of carrier/interface ratio, the coding mode of selection is AMR-HR, it is provided that higher wireless capacity.And when C/I is poor, then work in AMR-FR, it is provided that more better speech quality than EFR；
When full-rate mode and when half-rate mode, the MOS Distribution value figure that each code rate measures when different C/I values is as shown in Figure 3 and Figure 4, can be seen that, reduction along with carrier interference ratio C/I value, the MOS value of each code check all shows as the trend of reduction, because along with the enhancing of interference and noise, communication effect and speech quality that human ear is heard will be deteriorated.
AMR adaptive coding code check switching schematic diagram is as shown in Figure 5, known in gsm system AMR adaptation mechanism, the relief area arranged when switching between adjacent encoder speed, change along with C/I value, code rate will not be immediately switched to adjacent code check, but just carries out the switching of code check after waiting until the threshold value beyond relief area.Utilize relief area, it is possible to prevent the frequent saltus step of speed.
A kind of AMR code rate self-adapting regulation method of state Network Based, adopting the AMR code rate adaptive re-configuration police based on threshold method in the SIP software Sipdroid that increases income is using time delay and packet loss as parameter, and introduce the consideration of time delay trend prediction, formulate rate adaptation adjustable strategies, adapt to the real-time change of network.
Overall flow figure as shown in Figure 6, specifically comprises the following steps that
Step one, for call sender, according to the current network type that oneself is residing, it is determined that initial code pattern；
Current network type residing for each sender includes: 2G, 3G, more than 3G, or WiFi；
Code mode request CMR (CodecModeRequest), has 8 values, is followed successively by 0,1,2,3,4,5,6 and 7；
Utilize monitoring without the algorithm of RTP passage, utilize the network type receiving terminal to determine the code rate that each sender is initial.If what current sender connected is that 2G type network is conversed, then initial code pattern is CMR=0；Initial rate will be arranged on relatively low grade.If the time delay of recipient, packet loss are all smaller, then the code rate of sender can be adjusted upward.Whereas if the network type that current sender connects is 3G, more than 3G or WiFi network, then initial code pattern is CMR=7；Initial code rate is arranged on higher grade, if the time delay of recipient, packet loss are relatively larger, then adjusts downwards code rate.Wherein adjustment downwardly and upwardly is reversible, and the situation with specific reference to time delay and packet loss is determined.
Step 2, by corresponding for the initial code pattern of sender speed write RTP bag；
Speed corresponding to initial code pattern is followed successively by: 0 time speed of pattern is 4.75kbit/s；1 time speed of pattern is 5.15kbit/s；2 times speed of pattern are 5.90kbit/s；3 times speed of pattern are 6.70kbit/s；4 times speed of pattern are 7.40kbit/s；5 times speed of pattern are 7.95kbit/s；6 times speed of pattern are 10.2kbit/s；Under mode 7, speed is 12.2kbit/s.
Concrete transmitting procedure is as follows: terminal 1 and terminal 2 are conversed, and terminal 1, as sender, sends initial code rate to terminal 2, and the encode function of the incoming terminal 1 of initial Mode, Mode writes FT field in RTP bag；CMR value is write on the first character joint of payload and sends by terminal 1, and terminal 2 is as receiving terminal, and the first character joint of payload in the RTP bag being previously received is taken out by decode function, obtains CMR value, obtains Mode according to CMR；Terminal 2 is added up terminal 1 and is sent time delay and the packet loss of bag, judge the terminal 1 present rate pattern network if appropriate for terminal 2, if it is improper, sending through terminal 2 after adjusting CMR, the payloadheader in RTP bag notifies terminal 1, now, terminal 2 is as sender, in like manner, terminal 1 receives the RTP bag that terminal 2 sends and unpacks, after be encoded the adjustment of speed according to CMR value.
Demand according to transmission plan, it is necessary to the network type at place is judged by call terminal, and provide initial CMR value.
Sending side terminal, according to oneself residing network type, carries out the determination of initial CMR, and initial CMR has two states, CMR=0 under 2G network, namely AMR code rate is arranged on 4.75kbps, at 3G, CMR=7 when more than 3G or WiFi network, namely AMR code rate is arranged on 12.2kbps.
Step 3, RTP bag are transferred to receiving terminal, and receiving terminal calculates packet loss and the delay parameter of sender.
Specify about the RTP transformat of AMR according in RFC3267 and RFC4867, RTP bag form as it is shown in fig. 7, and notify receiving terminal by packet header of RTP bag, receiving terminal is unpacked after receiving RTP bag, extraction CMR value,
Wherein RTP header has 12 bytes, and PayloadHeader has a byte, and employing is octet-align pattern, and TableofContent contains the head of AMR frame, and part below is effective speech frame.And 12 bytes of RTPHeader are followed by payload payload in Sipdroid, this part payload is do not comprise coded modulation speed (CodecModeRequest, CMR) field.It is thus desirable to the RTP bag form in Sipdroid is done certain process: add a byte making a start, byte that receiving end correspondence subtracts, thus can comprise the judgement information of code rate.PayloadHeader is designed as octet-align pattern, and namely Gao Siwei represents CMR value, and low four are left 0.
The time delay transmitting terminal given out a contract for a project according to receiving terminal, packet drop, devise threshold method adjustable strategies.
Packet loss is the principal element considered, time delay is another Consideration.When first time adjusts code rate, judging according to initial CMR, in process afterwards, the judgement of CMR is to adjust according to the previous CMR value judging the moment.
First, it is utilized respectively code statement and calculates the packet loss cal_lost of RTP VoP:
Cal_lost=(packetlost/good*100)
Packetlost is packet loss quantity；Good is the data packet number of transmission.
Accumulated delay value cal_delay:
Cal_delay+=(jitter-250*mu)/8/mu；
Jitter is delay variation；Mu is sample rate, is the multiple of 8K；If sample rate is 16k, then mu is 2；
Average delay avg_delay, arranges every about 1 second and judges once:
Avg_delay=cal_delay/timeCount
TimeCount is frequency of reckoning by time, and every 1 second is 1 time.
Owing to network condition can exist the situation of sudden change, time delay obtained above is likely to a simply instantaneous value.When network condition has been not as, the value of packet loss and time delay is likely to change rapidly, therefore carries out code rate selection according to average packet loss ratio and instantaneous time delay value and is likely to inaccurate.Such as certain flashy packet loss and time delay are relatively low, and at this time just at the time judged, thus have selected the code rate of 12.2k, and actually network environment is not good, it is possible to cause that packet loss and time delay increase, worsen communication effect.
Accordingly, it would be desirable to judge the variation tendency of time delay within a period of time, the change of instantaneous time delay thus can be avoided to affect the selection of code rate.According to the computing formula about Delay Variation trend proposed in document, and through reality test, have chosen rational weighted value, rate adjustment algorithm adds the consideration of Delay Variation trend, utilize computing formula calculation delay.Thus can effectively solve saltus step problem frequently, and make the adjustment algorithm of speed more reasonable.
Consider the situation of the instantaneous saltus step of time delay, this paper presents the processing method for delay variation, add a delay variation statistical value N, when current time delay exceedes threshold delta T=80ms continuous three times with previous time delay difference DELTA d, i.e. N=3, prove that now network condition continues to be deteriorated, it is therefore desirable to consider currently acquired time delay value more, obtain time-delay calculation formula as follows:
Dtr(n+1)=α Dtr(n)+(1-α)Delay
Wherein DtrN () is the value of last Delay Variation, Dtr(n+1) being value next time, Delay is the instantaneous time delay value calculating gained.The meaning of this formula is that and eliminates instantaneous time delay value shake and the very big impact that code rate is judged of amplitude of variation, DtrAll relevant with former time delay value, there is Memorability, be a continually varying value.
According to actual measurement, adding the consideration to delay variation statistical value, as N=3, when namely network condition continues to be deteriorated, α=0.7, the weight of instantaneous time delay is higher.When N is < when 3, it was demonstrated that time delay sudden change now is simply temporary, instantaneous less α=0.9 of time delay weight.Above-mentioned value is according to repeatedly testing the empirical value chosen.
Step 4, receiving terminal, according to RTP bag, utilize packet loss and delay parameter, it is judged that the current initial code rate of sender, if appropriate for the current network state of receiving terminal, if be suitable for, keeps current code rate；Otherwise, change CMR value and adjust AMR code rate in real time.
The every 1s of receiving terminal once calculates judgement；
As shown in Figure 8, specifically comprise the following steps that
Step 401, according to time delay level, two delay threshold: Dth1 and Dth2 are set；
Dth1 is 200ms, Dth2 is 600ms；Owing to time delay is when lower than 200ms, the quality of voice call is barely affected, the obvious effect when time delay more than 600ms is, to voice call.And select Dth1 to be set to 200ms, Dth2 according to actual measurement experience and be set to 600ms.
Step 402, judge whether time delay meets: Dtr< Dth1, if it is, enter step 403；Otherwise enter step 407；
When time delay is in reduced levels (lower than threshold value Dth1), AMR code rate is only determined by packet loss.
Step 403, judge whether packet loss meets lost≤1%, if it is, enter step 404；Otherwise, step 405 is entered；
The threshold value of two packet loss it has been horizontally disposed with, owing to when packet loss is lower than 1%, AMR voice call quality is better, and both-end is conversed according to initial code rate, illustrates that now network condition is good, it is possible to improve step by step from initial CMR value according to packet loss.
Step 404, judge whether CMR value is 7；If it is, keep CMR value constant；Otherwise, current CMR value is from increasing 1；
Step 405, determining whether whether packet loss meets: 1% < lost≤5%, if it is, enter step 406, otherwise current CMR value is set to 0；
When packet loss occurs between 1%～5%, voice call quality will be affected.Now, the situation according to current CMR value, drops one-level to CMR accordingly.Turn out network condition when packet loss is more than 5% poor, directly select the minimum code rate 4.75kbps of CMR=0.
Step 406, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 407, determine whether whether time delay meets Dth1≤Dtr≤ Dth2, if it is, enter step 408；Otherwise, current CMR value is set to 0；
When time delay level is between threshold value Dth1 and Dth2, prove that now network delay is bigger, speech quality is had a certain impact, when time delay is in more than delay threshold Dth2, illustrate that network condition is in poor state, therefore, it is determined that CMR=0, the minimum code rate 4.75kbps of AMR is utilized to converse.
Step 408, judge whether packet loss meets lost≤1%, if it is, enter step 409；Otherwise, step 410 is entered；
When packet loss is lower than 1%, due to the impact of time delay, it is possible to suitably turn down the rank of CMR, reduce step by step.
Step 409, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 410, judge that whether packet loss meets 1% < lost≤5%, if it is, enter step 411；Otherwise current CMR value is set to 0；
When packet loss occurs between 1%～5%, voice call is relatively big by the impact of packet loss, and speech quality is also impacted by time delay simultaneously, and at this moment two-stage drops in CMR.
Step 411, judge that whether CMR value is be more than or equal to 2；If it is, current CMR value is from subtracting 2；Otherwise, current CMR value is set to 0；
Through the process to time delay and packet loss so that Adaptive Rate Shape strategy efficiently avoid the decision error that network sudden change brings, and advantageously ensures that the AMR effect conversed.Add the mode of the increasing or decreasing step by step of code rate, decrease the shake of code check, it also avoid occur that code rate improves suddenly too much cause the generation of the situation of network congestion, improve the quality of voice call.The code rate pattern of present terminal exports on screen, and the interface display in communication process is as shown in Figure 9.
The speed write RTP bag of the CMR value correspondence after adjustment is exported to sender by step 5, receiving terminal, and sender in like manner calculates packet loss and delay parameter, and judges by the adjustable strategies of step 4, changes CMR value and then real-time adjustment AMR code rate.
When carrying out the call of AMR codec speech, first software can determine initial code rate according to the current network conditions obtained, then in communication process, code rate can be adaptively adjusted according to the real-time change of network condition, to better profit from existing network condition, provide the user experience of better conversing.
Threshold method rate adaptation adjustable strategies is carried out voice call test, to prove the superiority of its performance.This test utilizes average suggestion value (MeanOpinionScore, MOS) to do criterion, and MOS value is primarily referred to as voice quality average index, it is simply that the voice masterplate contrast in the voice of test calls collection and MOS instrument.Score value is 1～5 point, and main reflection is the perceptibility of user, the intensity of general and signal, disturbed condition, and switch instances is relevant, 5 be divided into the highest.Test MOS value adopts subjective and objective two kinds of method of testings, wherein subjective MOS employing ITU-TP.800 and P.830 recommendation, there is the language material of decline to carry out subjective sensation contrast to original language material with after system processes respectively by different people, show that MOS divides, finally average.
Objective MOS evaluates subjective speech quality assessment (PerceptualEvaluationofSpeechQuality, the PESQ) method then adopting ITU-TP.862 recommendation to provide, special software PESQ test.The corresponding relation of PESQ and MOS value is as shown in table 1.
Table 1
| Rank | MOS value | User satisfaction | PESQ |
| Excellent | 5 | Very good, listen it is clear that undistorted sense, feel without delay | 3.5-4.5 |
| Good | 4 | Slightly worse, listen clear, postpone little, have a noise | 3.0-3.5 |
| In | 3 | It is also possible that listen not clear, there is certain delay, have noise, have distortion | 2.4-3.0 |
| Difference | 2 | Reluctantly, listening less clear, have relatively loud noise or interrupted, distortion is serious | 1.7-2.4 |
| Bad | 1 | Extreme difference, quiet or can not hear clearly completely, noise is very big | 1.0-1.7 |
Test by two ways, subjectivity evaluation and test and objective examination.Subjective evaluation and test is to utilize call software Sipdroid to carry out dual end communication, and speech quality is marked by caller according to MOS value evaluating standard.Objective examination is that the voice recording of both call sides is got off to be saved in mobile phone by design mechanism in software, is then marked by test and evaluation software PESQ by both sides' voice.Obtain the speech performance test result such as table 2.
Table 2
By data in table 2 it can be seen that the threshold method rate adaptation adjustable strategies effect when realizing voice call is better, MOS value score value is higher, does not have significant difference with the MOS value of other speed.This proves the better performances of threshold method rate adaptation adjustable strategies, it is possible to realizes in time speech encoding rate being adjusted along with the real-time change of network, and ensures speech quality.
The present invention proposes the AMR code rate adaptive re-configuration police based on threshold method, achieve the audio communication system based on mobile Internet, real-time change situation according to network condition, time delay and packet loss is utilized to make parameter, and introduce the consideration of time delay trend prediction, dynamically adjust ARM code rate, adapt to the real-time change of network；The call of AMR voice coding is made to carry out Adaptive Rate Shape in real time according to network condition, to provide the user experience of better conversing.
Claims (2)
1. the AMR code rate self-adapting regulation method of a state Network Based, it is characterised in that specifically comprise the following steps that
Step one, for call sender, determine initial code pattern according to oneself residing current network type；
If what current sender connected is that 2G type network is conversed, then initial code pattern is CMR=0；
If the network type that current sender connects is 3G, more than 3G or WiFi, then initial code pattern is CMR=7；
Step 2, by corresponding for the initial code pattern of sender speed write RTP bag；
During CMR=0, AMR code rate is arranged on 4.75kbps；During CMR=7, AMR code rate is arranged on 12.2kbps；
Step 3, RTP bag are transferred to the receiving terminal of call, and receiving terminal calculates packet loss and the delay parameter of sender；
The packet loss cal_lost of RTP bag is calculated as follows:
Cal_lost=(packetlost/good*100)
Packetlost is packet loss quantity；Good is the RTP bag quantity of transmission；
Accumulative instantaneous time delay value cal_delay:
Cal_delay+=(jitter-250*mu)/8/mu；
Jitter is delay variation；Mu is sample rate, for the multiple of 8K；
Average delay avg_delay, arranges every about 1 second and judges once:
Avg_delay=cal_delay/timeCount
TimeCount is frequency of reckoning by time, and every 1 second is 1 time；
According to delay variation statistical value, calculate final time delay Dtr:
Dtr(n+1)=α Dtr(n)+(1-α)Delay
Wherein DtrN () is the value of last Delay Variation, Delay is the instantaneous time delay value calculating gained；
Step 4, receiving terminal utilize packet loss and delay parameter, it is judged that the current initial code rate of sender, if appropriate for the current network state of receiving terminal, if be suitable for, keeps current code rate；Otherwise, CMR value is changed according to speed adjustable strategies；
Speed write RTP bag corresponding for CMR value after adjustment is exported to sender by step 5, receiving terminal, and sender in like manner calculates packet loss and delay parameter, and carries out judging change CMR value by the adjustable strategies of step 4, and then adjusts AMR code rate in real time.
2. the AMR code rate self-adapting regulation method of a kind of state Network Based as claimed in claim 1, it is characterised in that described step 4 particularly as follows:
Step 401, according to time delay level, two delay threshold: Dth1 and Dth2 are set；
Step 402, judge whether time delay meets: Dtr< Dth1, if it is, enter step 403；Otherwise enter step 407；
Step 403, judge whether packet loss meets lost≤1%, if it is, enter step 404；Otherwise, step 405 is entered；
Step 404, judge whether CMR value is 7；If it is, keep CMR value constant；Otherwise, current CMR value is from increasing 1；
Step 405, determining whether whether packet loss meets: 1% < lost≤5%, if it is, enter step 406, otherwise current CMR value is set to 0；
Step 406, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 407, determine whether whether time delay meets Dth1≤Dtr≤ Dth2, if it is, enter step 408；Otherwise, current CMR value is set to 0；
Step 408, judge whether packet loss meets lost≤1%, if it is, enter step 409；Otherwise, step 410 is entered；
Step 409, judge that whether CMR value is be more than or equal to 1；If it is, current CMR value is from subtracting 1；Otherwise, current CMR value is set to 0；
Step 410, judge that whether packet loss meets 1% < lost≤5%, if it is, enter step 411；Otherwise current CMR value is set to 0；
Step 411, judge that whether CMR value is be more than or equal to 2；If it is, current CMR value is from subtracting 2；Otherwise, current CMR value is set to 0.
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