A kind of Microphone calibration method, device and mobile terminalTechnical field
The present invention relates to technical field of mobile terminals, more particularly, relate to a kind of Microphone calibration method, device and mobile terminal.
Background technology
The sensitivity of microphone (MIC) refers to the voltage signal amplitude exported corresponding to microphone time given sound pressure (1 Pascal), and typical sensitivity requirement is 8 millivolts/Pascal.The sensitivity of microphone is relevant with factors such as the pole plate gap in sound-electric transition components, bias voltage, sensitive membrane tension force.Along with mobile terminal (such as, mobile phone) the continuous reduction of production cost, the continuous increase of delivering amount, between mobile phone, in the consistency of the production of microphone or assembling etc., also there is some difference, in order to obtain the more consistent microphone products of sensitivity, manufacturer needs strictly to control production technology.But due to the restriction of machining accuracy, the sensitivity of microphone that current typical microphone producer produces is in ± 3dB scope, this also causes microphone to flutter the signal strength signal intensity grasped certain otherness, thus the sound intensity performance consistency directly having influence on separate unit mobile phone transmitting terminal is very poor, it is bigger than normal or less than normal that the scene showed comprises the sound that user feedback the other side hears, thus cause Consumer's Experience to reduce.
Summary of the invention
The technical problem to be solved in the present invention is, for the above-mentioned defect of prior art, provide a kind of Microphone calibration method, device and mobile terminal, described method comprises step:
Receive the voice signal that the first microphone gathers, obtain the characteristic value of described first microphone as the First Eigenvalue, and using the standard value of described the First Eigenvalue as calibration;
Receive the voice signal that second microphone gathers, obtain the characteristic value of described second microphone as Second Eigenvalue;
Difference according to described the First Eigenvalue and Second Eigenvalue compensates second microphone gain, draws calibration-gain.
Alternatively, the characteristic value of described microphone is the loudness rating of microphone, is preserved by described the First Eigenvalue as loudness rating standard value.
Alternatively, the loudness rating of described acquisition microphone, comprises step:
Obtain the voice signal that microphone audio loop gathers, and generate corresponding voice data;
The loudness rating of microphone is calculated, using the loudness rating of the first microphone as loudness rating standard value according to described voice data and Scale Model of Loudness.
Alternatively, the described difference according to described the First Eigenvalue and Second Eigenvalue compensates second microphone gain, draws calibration-gain, is specially:
Calculate the difference of described the First Eigenvalue and Second Eigenvalue, then the calibration-gain of second microphone after overcompensation equals initial gain and described difference sum.
The present invention also proposes a kind of Microphone calibration device, is applied to mobile terminal, it is characterized in that, comprising:
Receiver module, for receiving the voice signal that microphone gathers;
Processing module, for calculating the characteristic value of microphone according to the voice signal gathered;
Gain compensation block, compensates second microphone gain for the difference according to described the First Eigenvalue and Second Eigenvalue, draws calibration-gain.
Alternatively, the characteristic value of described microphone is the loudness rating of microphone, using described the First Eigenvalue as loudness rating standard value.
Alternatively, it is characterized in that, described processing module also for:
Generate corresponding voice data according to the voice signal that microphone audio loop gathers, and calculate the loudness rating standard value of microphone according to Scale Model of Loudness.
Alternatively, it is characterized in that, described gain compensation block specifically for:
Calculate the difference of described the First Eigenvalue and Second Eigenvalue, then the calibration-gain of second microphone after overcompensation equals initial gain and described difference sum.
Alternatively, it is characterized in that, described device also comprises:
Memory module, for storing the First Eigenvalue of microphone as standard value, and the calibration-gain after compensating.
The present invention also proposes a kind of mobile terminal, it is characterized in that, described mobile terminal comprises one or more microphone, the sensitivity of microphone calibration steps of calibration steps employing according to any one of Claims 1-4 of described microphone or the sensitivity of microphone calibrating installation included according to any one of claim 5 to 9.
Implement a kind of Microphone calibration method of the present invention, device and mobile terminal, there is following beneficial effect: solve because of microphone when producing the consistency brought or assembling etc. reason, the problem of the signal strength signal intensity property of there are differences causing microphone to be flutterred grasping, even if produce sensitivity of microphone regulation within the specific limits, but through calibration after microphone mobile terminal integrated after its loudness there is consistency.
Accompanying drawing explanation
Below in conjunction with drawings and Examples, the invention will be further described, in accompanying drawing:
Fig. 1 is a kind of equal loudness contour figure that the embodiment of the present invention provides;
Fig. 2 is a kind of Microphone calibration method flow diagram that the embodiment of the present invention provides;
Fig. 3 is a kind of Microphone calibration apparatus structure schematic diagram that the embodiment of the present invention provides;
Fig. 4 is a kind of audio system physical structure schematic diagram that the embodiment of the present invention provides;
Fig. 5 is a kind of audio codec structural representation that the embodiment of the present invention provides;
Fig. 6 is a kind of Microphone calibration method flow diagram that the embodiment of the present invention provides;
Fig. 7 is a kind of Microphone calibration method flow diagram that the embodiment of the present invention provides;
Fig. 8 is a kind of audio system application schematic diagram that the embodiment of the present invention provides;
Fig. 9 is the hardware configuration schematic diagram of the optional mobile terminal realizing each embodiment of the present invention;
Figure 10 is the wireless communication system schematic diagram of mobile terminal as shown in Figure 10.
Embodiment
Should be appreciated that specific embodiment described herein only in order to explain the present invention, be not intended to limit the present invention.
The mobile terminal realizing each embodiment of the present invention is described referring now to accompanying drawing.In follow-up description, use the suffix of such as " module ", " parts " or " unit " for representing element only in order to be conducive to explanation of the present invention, itself is specific meaning not.Therefore, " module " and " parts " can mixedly use.
Acoustics ABC is introduced:
(1) threshold of audibility, the threshold of audibility divides intensity threshold and difference limen.The inadequate some strength of sound can not cause the sense of hearing.The minimal sound pressure levels of the sense of hearing is caused to be called intensity threshold (also claiming the threshold of audibility) the number of times of 50% repeatedly can be had on.The threshold of audibility has individual difference, and thus so-called normal threshold of audibility can only be the assembly average of the normal youthful threshold of audibility of some hearing.The threshold of audibility changes with frequency.Between 500 ~ 4000Hz, threshold value is minimum, on them and under high frequency sound and the threshold value of all-bottom sound all higher, if the threshold value of 20Hz pure tone is than the threshold value also 10dB about higher than the threshold value of 1000Hz pure tone of the threshold value of 1000Hz pure tone about high 70dB, 10000Hz pure tone.The most responsive frequency is about 3000Hz, and the amplitude of air molecule vibration reaches 10-11m and just can hear, this only has 1/10th of the diameter of hydrogen molecule.The threshold of audibility increased with the age, particularly HFS, and show as old deaf, as the old man of 70 years old, the threshold of audibility of 5000Hz pure tone about increased 45dB.
The concept of the threshold of audibility also comprises difference limen, and namely two sound cause other just noticeable difference of audible difference.Say with regard to frequency, can distinguish the difference of two pure tones of difference 0.5Hz at about 63Hz veteran ear, but this threshold value will be increased to 1.4Hz at 1000Hz, the higher difference limen of frequency is larger.The minimum 0.25dB of strength difference (1000 ~ 4000Hz, more than 70dB) that people's ear can be distinguished, when intensity is low or frequency is higher or lower, difference limen of intensity is larger.In whole audibility range, recognizable sound about 340,000.
(2) subjective attribute of sound, what namely loudness represented is the degree that a sound sounds polyphony.Loudness mainly changes with the intensity of sound, but also by the impact of frequency.The relation of both amounts, by classic psychophysics rule, loudness is directly proportional to the logarithm of intensity.In order to check the correctness of this hypothesis, modern psychology physics has carried out the rational judgment experiment of loudness, and establishes scaling of loudness, and its unit is Song (son).1 Song is defined as the loudness caused by 40dB1000Hz pure tone, is roughly equivalent to the sound level of whisper in sb.'s ear.Song's scale proves, loudness is proportional to 0.6 power of the sound pressures such as 1000Hz, and in other words, the sound pressure level of the sounds such as 1000Hz improves 10dB, and loudness doubles.The former is called loudness level, and this illustrates that the change of loudness is not merely be decided by intensity of sound, also relevant with frequency.Two pure tones of different frequency, though intensity is identical, the loudness caused is different.In a word, intermediate frequency pure tone sounds ringing than low frequency and high frequency pure tone.With the 1000Hz pure tone of different sound pressure level for reference sound, tested by loudness balance, cluster equal-loudness contour can be obtained, as shown above.In an equal-loudness contour, although the pure tone sound pressure level of each frequency is different, all ring with the 1000Hz pure tone on this curve etc.Namely this sound pressure level of 1000Hz pure tone is decided to be the loudness level of each pure tone on this curve, its side of being called of unit (PHON).
(3) equal loudness contour, the relation curve ringing sound pressure level and frequency of sound wave under condition such as to describe and is called equal loudness contour, is one of important aural signature.Namely pure tone at different frequencies needs to reach which kind of sound pressure level, could obtain hearing loudness consistent hearer.In FIG, abscissa is frequency, and ordinate is sound pressure level.Acoustic pressure is exactly the change produced after atmospheric pressure is disturbed, and be the overbottom pressure of atmospheric pressure, it is equivalent to the pressure change that the superposition disturbance on atmospheric pressure causes.Measurement ratio due to acoustic pressure is easier to realize, and also indirectly can try to achieve other physical quantitys such as particle velocity by the measurement of acoustic pressure, so this physical quantity conventional describes sound wave in acoustics.Sound pressure level represents with symbol SPL, and it is defined as gets common logarithm by acoustic pressure effective value p (e) to be measured and the ratio of reference sound pressure p (ref), then is multiplied by 20.The sound pressure level in figure, every bar curve corresponding to different frequency is not identical, but the response that people's ear is felt is the same, every bar curve is marked with a numeral, for volume unit, can be learnt by equal loudness contour race, when volume is less, people's ear is when to high bass perception is not enough, volume is larger, high bass perception is abundant, and people is the most responsive to sound between 1000Hz-4000Hz.Equal loudness contour is the foundation of loudness computing formula.
Be described in detail below by way of specific embodiment.
Embodiment one
See Fig. 2, give a kind of Microphone calibration method flow diagram, microphone applications, in mobile terminal, comprises step:
S11, receives the voice signal that the first microphone gathers, obtains the characteristic value of described first microphone as the First Eigenvalue, and using the standard value of described the First Eigenvalue as calibration.
Particularly, the mobile terminal in the present embodiment include but not limited to mobile phone, smart phone, notebook computer, digit broadcasting receiver, PDA (personal digital assistant), PAD (panel computer), PMP (portable media player), etc. mobile terminal.Due to the restriction of microphone machining accuracy, the sensitivity of microphone that microphone producer produces has the output of 80%-90% in the qualified index of ± 3dB scope.In the present embodiment, the first microphone is the microphone meeting qualified index in the sample of sampling, using the loudness rating corresponding to this microphone as loudness rating standard value.Wherein, the characteristic value of the first microphone is obtained by following process implementation:
Step 1: the test voice signal that the first microphones is sent by audio-frequency test equipment.
In present embodiment, the first microphone is arranged in mobile terminal, and this mobile terminal built-in automation produces the application program of line.When the calibration of audio-frequency test device start and when sending test voice signal, this application program directly calls microphone testing audio loop in mobile terminal to gather this test voice signal.
Step 2: the test voice signal received is sent to audio-frequency test equipment.
The voice signal collected by the first microphone sends to audio-frequency test equipment to analyze by mobile terminal.Namely this audio-frequency test equipment by testing voice signal digitized processing and carrying out preserving according to the mode of data vector and be transferred to the storage device of mobile terminal to be calibrated, see table 1, can comprise frequency and sound pressure level.
Table 1
| F (frequency) | F1 | F2 | … | Fn |
| SPL (sound pressure level) | SPL1 | SPL2 | … | SPLn |
Calculate the characteristic value of described first microphone:
Particularly, the characteristic value of microphone can be loudness rating, is obtained and the calculating of psychologic acoustics computing formula, finally obtain the first microphone loudness rating standard value by the data of digital signal processing module to storing apparatus of mobile terminal.The gain compensation block be stored in digital processing element is carried out standard value and presets by the loudness rating standard value of this microphone, and this preset value can compensate the microphone gain of calibration.Its psychologic acoustics computing formula is as follows:
Wherein, receive and send the constant coefficient of loudness, and the value relation of weight corresponding to corresponding frequencies is as shown in table 2, the m=0.175 in this formula.In the present embodiment, mainly for the calibration of microphone, refer to the sound of sending direction.
Table 2
Table1-WeightingfactorsWiforSLRandRLR
S12, receives the voice signal that second microphone gathers, obtains the characteristic value of described second microphone as Second Eigenvalue.
In the present embodiment, second microphone represents a microphone to be calibrated, be arranged in the mobile terminal to be calibrated of product line, the eigenvalue method obtaining second microphone is consistent with the method for above-mentioned acquisition first microphone properties value, and described Second Eigenvalue is the loudness rating actual value of this microphone.The eigenvalue method obtaining described second microphone is:
Step 1: second microphone receives the test voice signal sent by audio-frequency test equipment.
In present embodiment, second microphone is arranged in mobile terminal to be calibrated, and this mobile terminal built-in automation produces the application program of line.When the calibration of audio-frequency test device start and when sending test voice signal, this application program directly calls microphone testing audio loop in mobile terminal to gather this test voice signal.
Step 2: the test voice signal received is sent to audio-frequency test equipment.
The voice signal collected by second microphone sends to audio-frequency test equipment to analyze by mobile terminal.Namely this audio-frequency test equipment by testing voice signal digitized processing and carrying out preserving according to the mode of data vector and be transferred to the storage device of mobile terminal to be calibrated, see table 1, can comprise frequency and sound pressure level.
Calculate the characteristic value of described second microphone:
Particularly, the characteristic value of microphone can be loudness rating, is obtained and the calculating of psychologic acoustics computing formula, finally obtain second microphone loudness rating by the data of digital signal processing module to storing apparatus of mobile terminal.Its psychologic acoustics computing formula is as follows:
Wherein, receive and send the constant coefficient of loudness, and the value relation of weight corresponding to corresponding frequencies is as shown in table 2, the m=0.175 in this formula.In the present embodiment, mainly for the calibration of microphone, refer to the sound of sending direction.
S13, the difference according to described the First Eigenvalue and Second Eigenvalue compensates second microphone gain, draws calibration-gain.
Particularly, in above-mentioned steps, the First Eigenvalue result of calculation is expressed as LR0, preserve in a storage module as standard value, Second Eigenvalue result of calculation is expressed as LR.The initial gain of MIC is G0, then gain G=the G after compensating0+ (LR-LR0), and the gain after compensating is kept at memory module.This calibration-gain solves the problem because microphone causes sensitivity inconsistent due to reasons such as production technologies, and the data after gain compensation block directly give next digital signal processing module, thus ensures the loudness consistency of microphone.
Embodiment two
See Fig. 3, be a kind of sensitivity of microphone calibrating installation that the embodiment of the present invention provides, comprise: receiver module 11, processing module 12, gain compensation block 13, memory module 14, wherein,
Receiver module 11, for receiving the voice signal that microphone gathers.
Particularly, the mobile terminal in the present embodiment include but not limited to mobile phone, smart phone, notebook computer, digit broadcasting receiver, PDA (personal digital assistant), PAD (panel computer), PMP (portable media player), etc. mobile terminal.The sound that microphones is sent by audio-frequency test equipment artificial mouth, to mobile terminal (such as, mobile phone) frequency response curve of microphone tests, this method of testing presets the APP of automatic production line by mobile phone, when audio-frequency test device start is calibrated time, this APP directly can call the sound of the audio loop collection audio-frequency test equipment artificial mouth of microphone test.
Processing module 12, for calculating the characteristic value of microphone according to the voice signal gathered.
Particularly, after receiver module 11 collects voice signal, terminal is collected voice data corresponding to voice signal and sends to audio-frequency test equipment to analyze by processing module 12.The data of testing can be carried out preserving according to the mode of data vector and be transferred to mobile terminal storage device to be calibrated by this audio-frequency test equipment, see table 1, namely comprise frequency and sound pressure level.
Table 1
| F (frequency) | F1 | F2 | … | Fn |
| SPL (sound pressure level) | SPL1 | SPL2 | … | SPLn |
Calculate the characteristic value of described microphone:
Particularly, the characteristic value of microphone can be loudness rating, is obtained and the calculating of psychologic acoustics computing formula, finally obtain microphone loudness rating standard value by the data of digital signal processing module to storing apparatus of mobile phone.This microphone loudness rating standard value will be stored in gain compensation block in digital processing element and carry out standard value and preset, and this preset value can compensate the microphone gain of calibration.Its psychologic acoustics computing formula is as follows:
Wherein, receive and send the constant coefficient of loudness, and the weight that corresponding frequencies is corresponding, see table 2.In the present embodiment mainly for the calibration of microphone, refer to the sound of sending direction.
Gain compensation block 13, compensates second microphone gain for the difference according to described the First Eigenvalue and Second Eigenvalue, draws calibration-gain.
Particularly, the result of calculation providing the First Eigenvalue in processing module 12 is expressed as LR0, certainly deposit in a storage module as standard, Second Eigenvalue result of calculation is expressed as LR.The initial gain of MIC is G0, then gain G=the G after compensating0+ (LR-LR0), and the gain after compensating is kept at memory module.This calibration-gain solves the problem because microphone causes sensitivity inconsistent due to reasons such as production technologies, and the data after gain compensation block directly give next digital signal processing module, thus ensures the loudness consistency of microphone.
Memory module 14, for storing the First Eigenvalue of microphone as standard value, and the calibration-gain after compensating.
Particularly, processing module 12 sends the loudness rating of the first microphone to memory module, when calculating the difference with the loudness rating of second microphone, can call the loudness rating standard value of memory module.Gain after adjustment is carried out preserving the loudness consistency that its object is to ensure microphone.
Embodiment three
See Fig. 4, it is a kind of audio frequency processing system physical structure schematic diagram that the embodiment of the present invention provides.Comprise: audio codec 1000, processor 1100, audio codec (Codec) is for the treatment of audio-frequency information, all functions relevant to audio frequency such as input, output and volume control, communicated with digital audio interface DAI (IIS) by I2C bus between audio codec with processor (CPU), I2C bus is used for CPU by its read-write of realization to Codec register data, and DAI (IIS) is for realizing voice data communicating between CPU with Codec.
Wherein, audio codec 1000 also comprises: Mixer1001, ADC1002, DSP1003, DAC1004.See Fig. 5,
The input of audio codec comprises microphone (MIC), telephone signal (PhoneIn), exports and comprises earphone (HP, HeadPhone), loud speaker (Spk, Speaker), telephone signal (PhoneOut), wherein
Mixer1001, mixer, for being mixed into a kind of analog signal by several audio analog signals from different passage.When the sound that microphones is sent by audio-frequency test equipment artificial mouth, the voice signal of different sound channel is mixed into a kind of analog signal by mixer.
ADC1002, A/D converter, for being converted to discrete digital signal by the analog signal of continuous variable.For microphones to voice signal be converted to digital signal as voice data, voice data terminal collected sends to audio-frequency test equipment to analyze, and these data are carried out preserving according to the mode of data vector and are transferred to the mobile terminal of collected sound signal the most at last.Described data vector mode comprises the frequency of voice signal and the sound pressure level of correspondence.
DSP1003, digital signal processor, for the treatment of the voice data of ADC module.The loudness rating of sound is calculated according to this voice data and psychologic acoustics computing formula, the loudness rating standard value produced for microphone and the standard microphone of a qualified index of satisfied production is kept at memory module, after drawing the loudness rating actual value of microphone to be calibrated, calculate the difference with loudness rating standard value, utilize this difference to adjust the gain of microphone, and the loudness consistency of carrying out preserving and its object is to ensure microphone is preserved in the gain after compensating.
Embodiment four
When mobile terminal has two or more microphone, should calibrate each microphone respectively, see Fig. 6, give a kind of Microphone calibration method flow diagram, comprise step:
S21, receives the voice signal that the first microphone gathers, obtains the characteristic value of described first microphone as the First Eigenvalue, and using the standard value of described the First Eigenvalue as calibration.
Particularly, the mobile terminal in the present embodiment include but not limited to mobile phone, smart phone, notebook computer, digit broadcasting receiver, PDA (personal digital assistant), PAD (panel computer), PMP (portable media player), etc. mobile terminal.Due to the restriction of microphone machining accuracy, the sensitivity of microphone that microphone producer produces has the output of 80%-90% in the qualified index of ± 3dB scope.In the present embodiment, the first microphone is the microphone meeting qualified index in the sample of sampling, and the loudness rating produced by this microphone is as standard value.Its characteristic value process obtaining this microphone is as follows:
Step 1: the test voice signal that the first microphones is sent by audio-frequency test equipment.
In present embodiment, the first microphone is arranged in mobile terminal, and this mobile terminal built-in automation produces the application program of line.When the calibration of audio-frequency test device start and when sending test voice signal, this application program directly calls microphone testing audio loop in mobile terminal to gather this test voice signal.
Step 2: the test voice signal received is sent to audio-frequency test equipment.
The voice signal collected by the first microphone sends to audio-frequency test equipment to analyze by mobile terminal.Namely this audio-frequency test equipment by testing voice signal digitized processing and carrying out preserving according to the mode of data vector and be transferred to the storage device of mobile terminal to be calibrated, see table 1, can comprise frequency and sound pressure level.
Table 1
| F (frequency) | F1 | F2 | … | Fn |
| SPL (sound pressure level) | SPL1 | SPL2 | … | SPLn |
Calculate the characteristic value of described first microphone:
Particularly, the characteristic value of microphone can be loudness rating, is obtained and the calculating of psychologic acoustics computing formula, finally obtain the first microphone loudness rating standard value by the data of digital signal processing module to storing apparatus of mobile terminal.The gain compensation block be stored in digital processing element is carried out standard value and presets by the loudness rating standard value of this microphone, and this preset value can compensate the microphone gain of calibration.Its psychologic acoustics computing formula is as follows:
Wherein, receive and send the constant coefficient of loudness, and the value relation of weight corresponding to corresponding frequencies is as shown in table 2, the m=0.175 in this formula.In the present embodiment, mainly for the calibration of microphone, refer to the sound of sending direction.
S22, receives the voice signal that microphone to be calibrated gathers respectively, obtains the characteristic value of described microphone as Second Eigenvalue.
In the present embodiment, microphone to be calibrated has two or more microphones, the eigenvalue method obtaining described microphone to be calibrated is respectively consistent with the method for above-mentioned acquisition first microphone properties value, and described Second Eigenvalue is the loudness rating actual value of corresponding microphone to be calibrated.Acquisition methods is:
The sound that microphones to be calibrated is sent by audio-frequency test equipment artificial mouth, to mobile terminal (such as, mobile phone) frequency response curve of microphone tests, this method of testing presets the APP of automatic production line by mobile phone, when audio-frequency test device start is calibrated time, this APP directly can call the sound of the audio loop collection audio-frequency test equipment artificial mouth of this microphone test, and the voice data that terminal collects the most at last sends to audio-frequency test equipment to analyze.The data of testing can be carried out preserving according to the mode of data vector and be transferred to mobile phone storage device by this audio-frequency test equipment.Then, obtained by the data of digital signal processing module to storing apparatus of mobile phone and the calculating of psychologic acoustics computing formula, finally obtain each microphone loudness rating actual value to be calibrated, and preserve corresponding loudness rating actual value respectively.The quantity of microphone is mated according to the number of microphone of Cell Phone Design in the present embodiment.
S23, the difference according to described the First Eigenvalue and Second Eigenvalue to be calibrated compensates microphone gain to be calibrated, draws calibration-gain.
Particularly, in above-mentioned steps, the First Eigenvalue result of calculation is expressed as LR0, preserve in a storage module as standard value, Second Eigenvalue result of calculation is expressed as LR1, LR2.The initial gain of MIC is G0, then gain G=the G after first microphone compensation to be calibrated0+ (LR1-LR0), the gain G=G after second microphone to be calibrated compensates0+ (LR2-LR0), and respectively the gain after compensation is kept at corresponding microphone gain memory module.Data after gain compensation block directly give next digital signal processing module, thus ensure the loudness consistency of microphone.
Should be understood that, because terminal often possesses two or more microphone, the First Eigenvalue of described microphone is not limited to one, each microphone is all corresponding to a First Eigenvalue.
Based on above-described embodiment three and embodiment four, provide a kind of application process of audio frequency processing system, see Fig. 7, a kind ofly be applied to the audio frequency processing system schematic diagram called, MIC, Mixer1001 of audio frequency processing system is related to, ADC1002, DSP1003 in the present embodiment, DAC1004 and earphone (HP, HeadPhone).In this application scenarios, MIC receives the voice signal of sound as input of user, again by Mixer1001 in audio codec, ADC1002, DSP1003, DAC1004 carry out processing transmission of sound signals to earphone (HP, HeadPhone), microphone after calibration is due to the gain after DSP stores compensation, and therefore, the telephone signal loudness that different microphone adopts PhoneOut to export has consistency.See Fig. 8, be a kind of audio frequency processing system schematic diagram being applied to recording, in the present embodiment, relate to MIC, Mixer1001 of audio frequency processing system, ADC1002, DSP1003, CPU1100 and earphone (HP, HeadPhone).In this application scenarios, MIC receives the voice signal of sound as input of user, again by Mixer1001 in audio codec, ADC1002, DSP1003 carries out processing transmission of sound signals to processor CPU, microphone after calibration is due to the gain after DSP stores compensation, and therefore, the loudness that different microphone gathers recorded audio signals has consistency.
Embodiment five
Mobile terminal can be implemented in a variety of manners.Such as, the terminal described in the present invention can comprise the such as mobile terminal of mobile phone, smart phone, notebook computer, digit broadcasting receiver, PDA (personal digital assistant), PAD (panel computer), PMP (portable media player), guider etc. and the fixed terminal of such as digital TV, desktop computer etc.Below, suppose that terminal is mobile terminal.But it will be appreciated by those skilled in the art that except the element except being used in particular for mobile object, structure according to the embodiment of the present invention also can be applied to the terminal of fixed type.
Fig. 9 is the hardware configuration schematic diagram of the optional mobile terminal realizing each embodiment of the present invention.
See Fig. 9, mobile terminal 100 comprises controller 200, storage device 310, GPS chip 320, communicator 330, video processor 340, audio process 350, interface unit 360, microphone 370, loud speaker 380 and power subsystem 390.
Storage device 310 software program that can store process and the control operation performed by controller 180 etc., or temporarily can store oneself through exporting the data (such as, telephone directory, message, still image, video etc.) that maybe will export.And storage device 310 can store corresponding program for adjusting amplifier of microphone gain and data.
Storage device 310 can comprise the storage medium of at least one type, described storage medium comprises flash memory, hard disk, multimedia card, card-type memory (such as, SD or DX memory etc.), random access storage device (RAM), static random-access memory (SRAM), read-only memory (ROM), Electrically Erasable Read Only Memory (EEPROM), programmable read only memory (PROM), magnetic storage, disk, CD etc.And mobile terminal 100 can be connected the memory function performing storage device 310 network storage device with by network cooperates.
Controller 200 is undertaken transmit by using the program that is stored in storage device 310 and data and processes.
Controller 200 comprises RAM210, ROM220, CPU230, GPU (Graphics Processing Unit) 240 and bus 250.RAM210, ROM220, CPU230 and GPU240 can be connected to each other by bus 250.
CPU (processor) 230 access to storage device 310 and use the operating system (OS) be stored in storage device 310 perform startup.And CPU230 performs various operation by using various programs, content and the data be stored in storage device 310.
ROM220 stores the command set being used for system and starting.When open command is transfused to and electric power is provided, the OS be stored in storage device 310 is copied to RAM210 according to being stored in command set in ROM220 by CPU230, and by running OS start up system.When startup completes, the various program copies that are stored in storage device 310 to RAM210, and are performed various operation by the reproducer run in RAM210 by CPU230.Specifically, GPU240 can generate by using calculator (not shown) and renderer (not shown) the screen comprising the so various objects of such as icon, image and text.Calculator calculates the characteristic value that such as coordinate figure, form, size and color are such, wherein respectively according to the layout of screen color mark object.
GPS chip 320 is the unit from GPS (global positioning system) satellite reception gps signal, and calculates the current location of mobile terminal 100.When using Navigator or when asking the current location of user, controller 200 can by the position using GPS chip 320 to calculate user.
Communicator 330 is the unit according to various types of communication means and various types of external equipment executive communication.Communicator 330 comprises WiFi chip 331, Bluetooth chip 332, wireless communication chips 333 and NFC chip 334.Controller 200 performs the communication with various external equipment by using communicator 330.
WiFi chip 331 and Bluetooth chip 332 are respectively according to WiFi method and bluetooth approach executive communication.When using WiFi chip 331 or Bluetooth chip 332, such as service set identifier (servicesetidentifier, SSID) and the such various link informations of session key can first be received and dispatched, communication can be connected by using link information, and various information can be received and dispatched.Wireless communication chips 333 is the chips according to such as IEEE, Zigbee, 3G (third generation), the such various communication standard executive communications of 3GPP (third generation collaborative project) and LTE (Long Term Evolution).NFC chip 334 is that various RF-ID frequency bandwidth is 135 KHz, 13.56 megahertzes, 433 megahertzes, 860 ~ 960 megahertzes and 2.45 gigahertz (GHZ)s such as according to using NFC (near-field communication) method of 13.56 gigahertz bandwidth in the middle of various RF-ID frequency bandwidth to carry out the chip operated.
Video processor 340 is that process is included in the content that received by communicator 330 or is stored in the unit of the video data in the content in storage device 310.Video processor 340 can perform the various image procossing for video data, such as decoding, convergent-divergent, noise filtering, frame rate conversion and resolution conversion.
Audio process 350 is that process is included in the content that received by communicator 330 or is stored in the unit of the voice data in the content in storage device 310.Audio process 350 can perform the various process for voice data, such as decodes, amplifies and noise filtering.
Corresponding contents can be reproduced by driving video processor 340 and audio process 350 when running playback program Time Controller 200 for content of multimedia.
Loud speaker 380 exports the voice data generated in audio process 350.
Microphone 370 is the unit receiving user speech or other sound and they are transformed to voice data.The user speech inputted by microphone 370 during controller 200 can be used in calling procedure, or they are transformed to voice data and are stored in storage device 310.
Interface unit 170 is used as at least one external device (ED) and is connected the interface that can pass through with mobile terminal 100.Such as, external device (ED) can comprise wired or wireless head-band earphone port, external power source (or battery charger) port, wired or wireless FPDP, memory card port, for connecting the port, audio frequency I/O (I/O) port, video i/o port, ear port etc. of the device with identification module.Identification module can be that storage uses the various information of mobile terminal 100 for authentication of users and can comprise subscriber identification module (UIM), client identification module (SIM), Universal Subscriber identification module (USIM) etc.In addition, the device (hereinafter referred to " recognition device ") with identification module can take the form of smart card, and therefore, recognition device can be connected with mobile terminal 100 via port or other jockey.Interface unit 170 may be used for receive from external device (ED) input (such as, data message, electric power etc.) and the input received be transferred to the one or more element in mobile terminal 100 or may be used for transmitting data between mobile terminal and external device (ED).
In addition, when mobile terminal 100 is connected with external base, interface unit 170 can be used as to allow by it electric power to be provided to the path of mobile terminal 100 from base or can be used as the path that allows to be transferred to mobile terminal by it from the various command signals of base input.The various command signal inputted from base or electric power can be used as and identify whether mobile terminal is arranged on the signal base exactly.
Mobile terminal 100 as shown in Figure 9 can be constructed to utilize and send the such as wired and wireless communication system of data via frame or grouping and satellite-based communication system operates.
Describe wherein according to the communication system that mobile terminal of the present invention can operate referring now to Figure 10.
Such communication system can use different air interfaces and/or physical layer.Such as, the air interface used by communication system comprises such as frequency division multiple access (FDMA), time division multiple access (TDMA), code division multiple access (CDMA) and universal mobile telecommunications system (UMTS) (especially, Long Term Evolution (LTE)), global system for mobile communications (GSM) etc.As non-limiting example, description below relates to cdma communication system, but such instruction is equally applicable to the system of other type.
With reference to Figure 10, cdma wireless communication system can comprise multiple mobile terminal 100, multiple base station (BS) 270, base station controller (BSC) 275 and mobile switching centre (MSC) 280.MSC280 is constructed to form interface with Public Switched Telephony Network (PSTN) 290.MSC280 is also constructed to form interface with the BSC275 that can be couple to base station 270 via back haul link.Back haul link can construct according to any one in some interfaces that oneself knows, described interface comprises such as E1/T1, ATM, IP, PPP, frame relay, HDSL, ADSL or xDSL.Will be appreciated that system as shown in Figure 10 can comprise multiple BSC2750.
Each BS270 can serve one or more subregion (or region), by multidirectional antenna or point to specific direction each subregion of antenna cover radially away from BS270.Or each subregion can by two or more antenna covers for diversity reception.Each BS270 can be constructed to support multiple parallel compensate, and each parallel compensate has specific frequency spectrum (such as, 1.25MHz, 5MHz etc.).
Subregion can be called as CDMA Channel with intersecting of parallel compensate.BS270 also can be called as base station transceiver subsystem (BTS) or other equivalent terms.Under these circumstances, term " base station " may be used for broadly representing single BSC275 and at least one BS270.Base station also can be called as " cellular station ".Or each subregion of particular B S270 can be called as multiple cellular station.
As shown in Figure 10, broadcast singal is sent to the mobile terminal 100 at operate within systems by broadcsting transmitter (BT) 295.Broadcast reception module 111 as shown in Figure 9 is arranged on mobile terminal 100 and sentences the broadcast singal receiving and sent by BT295.In Fig. 10, several global positioning system (GPS) satellite 300 is shown.Satellite 300 helps at least one in the multiple mobile terminal 100 in location.
In Fig. 10, depict multiple satellite 300, but understand, the satellite of any number can be utilized to obtain useful locating information.GPS module 115 as shown in Figure 9 is constructed to coordinate to obtain the locating information wanted with satellite 300 usually.Substitute GPS tracking technique or outside GPS tracking technique, can use can other technology of position of tracking mobile terminal.In addition, at least one gps satellite 300 optionally or extraly can process satellite dmb transmission.
As a typical operation of wireless communication system, BS270 receives the reverse link signal from various mobile terminal 100.Mobile terminal 100 participates in call usually, information receiving and transmitting communicates with other type.Each reverse link signal that certain base station 270 receives is processed by particular B S270.The data obtained are forwarded to relevant BSC275.BSC provides call Resourse Distribute and comprises the mobile management function of coordination of the soft switching process between BS270.The data received also are routed to MSC280 by BSC275, and it is provided for the extra route service forming interface with PSTN290.Similarly, PSTN290 and MSC280 forms interface, and MSC and BSC275 forms interface, and BSC275 correspondingly control BS270 so that forward link signals is sent to mobile terminal 100.
In describing the invention, it is to be appreciated that term " first ", " second " etc. are only for describing object, and instruction or hint relative importance can not be interpreted as.In addition, in describing the invention, except as otherwise noted, the implication of " multiple " is two or more.
In flow chart or any process otherwise described in an embodiment of the present invention or method describe and can be understood to, represent and comprise one or more for realizing the module of the code of the executable instruction of the step of specific logical function or process, fragment or part, and the scope of embodiment of the present invention comprises other realization, wherein can not according to order that is shown or that discuss, comprise according to involved function by the mode while of basic or by contrary order, carry out n-back test, this should understand by those skilled in the art described in embodiments of the invention.
Professional can also recognize, in conjunction with unit and the algorithm steps of each example of embodiment disclosed herein description, can realize with electronic hardware, computer software or the combination of the two, in order to the interchangeability of hardware and software is clearly described, generally describe composition and the step of each example in the above description according to function.These functions perform with hardware or software mode actually, depend on application-specific and the design constraint of technical scheme.Professional and technical personnel can use distinct methods to realize described function to each specifically should being used for, but this realization should not thought and exceeds scope of the present invention.
The software module that the method described in conjunction with embodiment disclosed herein or the step of method can use hardware, processor to perform, or the combination of the two is implemented.Software module can be placed in the storage medium of random asccess memory (RAM), internal memory, read-only memory (ROM), electrically programmable ROM, electrically erasable ROM, register, hard disk, moveable magnetic disc, CD-ROM or other form arbitrarily.
By reference to the accompanying drawings embodiments of the invention are described above; but the present invention is not limited to above-mentioned embodiment; above-mentioned embodiment is only schematic; instead of it is restrictive; those of ordinary skill in the art is under enlightenment of the present invention; do not departing under the ambit that present inventive concept and claim protect, also can make a lot of form, these all belong within protection of the present invention.