CN106385489A - Method for determining uplink voice data and mobile terminal - Google Patents
Method for determining uplink voice data and mobile terminal Download PDFInfo
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- CN106385489A CN106385489A CN201610798983.7A CN201610798983A CN106385489A CN 106385489 A CN106385489 A CN 106385489A CN 201610798983 A CN201610798983 A CN 201610798983A CN 106385489 A CN106385489 A CN 106385489A
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- speech data
- mobile terminal
- preset value
- secondary microphone
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
- H04M1/7243—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages
- H04M1/72433—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality with interactive means for internal management of messages for voice messaging, e.g. dictaphones
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/02—Constructional features of telephone sets
- H04M1/19—Arrangements of transmitters, receivers, or complete sets to prevent eavesdropping, to attenuate local noise or to prevent undesired transmission; Mouthpieces or receivers specially adapted therefor
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- Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Business, Economics & Management (AREA)
- General Business, Economics & Management (AREA)
- Human Computer Interaction (AREA)
- Computer Networks & Wireless Communication (AREA)
- Telephone Function (AREA)
Abstract
The present invention provides a method for determining uplink voice data and a mobile terminal. The mobile terminal is configured for a main microphone and at least one auxiliary microphone for collecting voice data generated by a sound source, there is a certain distance between the main microphone and the at least one auxiliary microphone, and the main microphone is configured to monitor whether the amplitude of the pulse coding modulation signals of the first voice data collected by the main microphone is larger than a first preset value or not; if the amplitude of the pulse coding modulation signals of the first voice data collected by the main microphone is larger than the first preset value at one moment, the second voice data collected by the at least one auxiliary microphone at the moment is obtained; and the second voice data replaces the first voice data at the same moment and is taken as the uplink voice data for transmission so as to effectively eliminate the distortion condition of the uplink voice data, improve the quality of the uplink voice data and enhance the usage feeling of the users for the mobile phone.
Description
Technical field
The present embodiments relate to communication technical field, more particularly, to a kind of method determining upstream voice data and movement
Terminal.
Background technology
The upstream voice data of mobile terminal mainly includes recording at present, QQ wechat language is chatted, video, phone etc..At these
In scene, when sound source distance microphone is nearer, or when speaking up, the air-flow that voice produces is tied by mike leading note
Structure is applied to mike, the acoustic pressure of generation can reach the acoustics overload point of mike (Acoustics Overload Point,
AOP), and acoustic pressure exceed the upstream voice data of microphone acoustic overload point amplify through the analog gain of mobile terminal, modulus
After conversion process, the pulse code entering data signal processing module (Digital Signal Processing, DSP) is adjusted
Signal amplitude processed can reach 0 full scale decibel (dB Full Scale, dBFS), thus leading to speech data the feelings of distortion
Condition, occurs noise during broadcasting.
In prior art, improve the situation of speech data distortion by the technological means reducing analog gain, but, fall
The drawbacks of low analog gain produces is to send signal to noise ratio to reduce, and this processing mode equally can affect the matter of upstream voice data
Amount.
Content of the invention
The present invention provide a kind of determine the method for upstream voice data and mobile terminal, with solve existing for prior art by
The problem of upstream voice data quality cannot be guaranteed in speech data distortion.
On the one hand, embodiments provide a kind of method determining upstream voice data, be applied to mobile terminal, institute
State a main microphon and at least one secondary Mike that mobile terminal specifically can include the speech data for gathering sound source generation
Wind, has a distance between described main microphon and at least one secondary microphone described;Methods described includes:
Whether the amplitude of the pulse code modulation signal of the first speech data of monitoring main microphon collection is pre- more than first
If value;
If the amplitude of the pulse code modulation signal of described first speech data in a moment is more than the first preset value, obtain
The second speech data that at least one secondary microphone described gathered in the described moment;
The first speech data that described second speech data is replaced synchronization is transmitted as upstream voice data.
On the other hand, the embodiment of the present invention additionally provides a kind of mobile terminal, and described mobile terminal is included for gathering sound
A main microphon of speech data and at least one secondary microphone that source produces, described main microphon and at least one secondary wheat described
Between gram wind, there is a distance;Described mobile terminal also includes:
Monitoring modular, the amplitude of pulse code modulation signal of the first speech data for monitoring main microphon collection is
No it is more than the first preset value;
Acquisition module, if the amplitude for the pulse code modulation signal of described first speech data in a moment is more than the
One preset value, obtains the second speech data that at least one secondary microphone described gathers in the described moment;
Transport module, for replacing the first speech data of synchronization as ascending voice using described second speech data
Data is transmitted.
From the above as can be seen that provided in an embodiment of the present invention determine the method for upstream voice data and move eventually
End, by being provided for gathering a main microphon and at least one secondary Mike of the speech data that sound source produces in mobile terminal
Wind, has a distance, and monitors the first of main microphon collection between described main microphon and at least one secondary microphone described
Whether the amplitude of the pulse code modulation signal of speech data is more than the first preset value;If the first speech data described in the moment
The amplitude of pulse code modulation signal is more than the first preset value, and acquisition at least one secondary microphone described gathered in the described moment
Second speech data;The first speech data that described second speech data is replaced synchronization is carried out as upstream voice data
Transmission.Thus the distortion situation of upstream voice data can effectively be eliminated, lifting upstream voice data quality, strengthen user for
The use feeling of mobile terminal.
Brief description
In order to be illustrated more clearly that the embodiment of the present invention or technical scheme of the prior art, below will be to embodiment or existing
Have technology description in required use accompanying drawing be briefly described it should be apparent that, drawings in the following description be only this
Some embodiments of invention, for those of ordinary skill in the art, without having to pay creative labor, also may be used
So that other accompanying drawings are obtained according to these accompanying drawings.
Fig. 1 is mobile terminal structure schematic diagram one provided in an embodiment of the present invention;
Fig. 2 is that the method determining upstream voice data provided in an embodiment of the present invention realizes process schematic one;
Fig. 3 is mobile terminal structure schematic diagram two provided in an embodiment of the present invention;
Fig. 4 is that the method determining upstream voice data provided in an embodiment of the present invention realizes process schematic two;
Fig. 5 is mobile terminal structure schematic diagram three provided in an embodiment of the present invention;
Fig. 6 is mobile terminal structure schematic diagram four provided in an embodiment of the present invention;
Fig. 7 is acquisition module structural representation in mobile terminal provided in an embodiment of the present invention;
Fig. 8 is mobile terminal structure schematic diagram five provided in an embodiment of the present invention;
Fig. 9 is mobile terminal structure schematic diagram six provided in an embodiment of the present invention.
Specific embodiment
Purpose, technical scheme and advantage for making the embodiment of the present invention are clearer, below in conjunction with the embodiment of the present invention
Accompanying drawing, the technical scheme of the embodiment of the present invention is clearly and completely described.Obviously, described embodiment is this
Bright a part of embodiment, rather than whole embodiments.Based on described embodiments of the invention, ordinary skill
The every other embodiment that personnel are obtained, broadly falls into the scope of protection of the invention.
Unless otherwise defined, technical term used herein or scientific terminology should be in art of the present invention and have
The ordinary meaning that the personage of general technical ability is understood.Used in present patent application description and claims " the
One ", " second " and similar word are not offered as any order, quantity or importance, and are used only to distinguish different
Ingredient.Equally, the similar word such as " one " or " " does not indicate that quantity limits yet, but represents and there is at least one.
The word that " connection " or " being connected " etc. are similar to is not limited to physics or machinery connection, but can include electrical
Connect, no matter being direct or indirectly." on ", D score, "left", "right" etc. be only used for representing relative position relation, work as quilt
After the absolute position of description object changes, this relative position relation also correspondingly changes.
Embodiments provide a kind of method determining upstream voice data, be applied to mobile terminal 10, as Fig. 1 institute
Show, in this mobile terminal 10, specifically can be provided with the main microphon 11 and at least of the speech data for gathering sound source generation
One secondary microphone 12, and, this main microphon 11 is less than the distance apart from sound source for the secondary microphone 12 apart from the distance of sound source, that is,
Have one apart from L between main microphon 11 and at least one secondary microphone 12.
In one embodiment, as shown in figure 1, the main microphon 11 involved by the embodiment of the present invention specifically may be disposed at
Bottom one predetermined position of mobile terminal 10 is (near user's mouth when concretely mobile terminal 10 is normally used for this bottom
The i.e. part of sound source), at least one secondary microphone 12 is arranged at mobile terminal 10 top relative with described bottom one and presets position
Place's of putting (near the part of user's auricle when concretely mobile terminal 10 is normally used on this top).When secondary microphone 12 exists
When multiple, multiple secondary microphones 12 may be disposed at the predetermined position on mobile terminal 10 top.
As shown in Fig. 2 this determines that the method for upstream voice data specifically can include:
Step 201, the amplitude of the pulse code modulation signal of the first speech data of real-time monitoring main microphon 11 collection
Whether it is more than the first preset value.
When the acoustic pressure that sound source produces reaches the acoustics overload point (Acoustics Overload Point, AOP) of mike
When the situation of speech data distortion occur, so in the case of the amplitude of the pulse code modulation signal of speech data can be close
0dBFS.So in order to improve the quality of upstream voice data, it is to avoid the situation of distortion in upstream voice data, the present invention is real
The method applying determination upstream voice data that example is provided needs the first speech data of real-time monitoring main microphon 11 collection
Whether the amplitude of pulse code modulation signal is more than the first preset value (limit1), and in the first language gathering when main microphon 11
When the amplitude of the pulse code modulation signal of sound data is more than the first preset value, abandon this first speech data, by the application institute
The secondary microphone 12 being related to replace that this distortion in the second speech data with the first speech data synchronization collection the
One speech data is transmitted as final upstream voice data;If the pulse code modulation letter of above-mentioned first speech data
Number amplitude be less than or equal to the first preset value, this first speech data is transmitted as upstream voice data.
As described above, when speech data pulse code modulation signal amplitude can close to 0dBFS when speech data can go out
, therefore, the situation of distortion, involved by the embodiment of the present invention in order to avoid upstream voice data in the situation of existing distortion
First preset value (limit1) is smaller than 0dBFS, such as limit1=-1dBFS etc..This first preset value is an adjustable numerical value,
In concrete application, can be set based on actual demand and be chosen.
In order to obtain the amplitude of the pulse code modulation signal of speech data, the method involved by the embodiment of the present invention is being held
Following process step specifically can also be included before row step 301:
Gain process is simulated to the speech data of main microphon 11 and secondary microphone 12 collection;
Analog-to-digital conversion process is carried out to the speech data being simulated gain process, obtains main microphon 11 and secondary microphone
The pulse code modulation signal of the speech data of 12 collections;
The pulse code modulation signal of the speech data that main microphon 11 and secondary microphone 12 are gathered stores respectively to right
In the buffer 15 answered.
As shown in figure 3, analog gain module 13, mould specifically can also be included within mobile terminal 10 involved by the application
The devices such as number modular converter (ADC) 14, buffer (Buffer) 15, to realize above-mentioned process step.
And there is multiple mikes (main microphon 11 He corresponding to the mobile terminal 10 involved by the embodiment of the present invention
At least one secondary microphone 12) situation, there may be and multiple mikes in the mobile terminal 10 involved by the embodiment of the present invention
Corresponding multirouting analog gain module 13, analog-to-digital conversion module 14, the language data process passage of buffer 15 composition, with reality
Now respectively gain process, analog-to-digital conversion process and buffered are simulated to the speech data of each mike collection.
Above-mentioned involved buffer 15 specifically may be disposed in the CPU16 of mobile terminal 10, within this CPU16 specifically also
Digital signal processing module (DSP) 17 can be set, for realizing the amplitude inspection of the pulse code modulation signal of speech data
Survey, compare, determine the concrete operations of upstream voice data.Analog gain module 13 involved by the embodiment of the present invention, modulus turn
Die change block 14, buffer 15 etc. specifically can be realized using ripe, positive means.
Step 202, if the amplitude of the pulse code modulation signal of moment first speech data is more than the first preset value, obtains
Take the second speech data that at least one secondary microphone 12 gathers in this moment.
If the amplitude of the pulse code modulation signal of moment first speech data is more than the first preset value, this voice is described
There is the possibility of distortion in data, therefore, can abandon this first speech data, and obtains at least one secondary microphone 12 in this moment
Collection second speech data come to replace main microphon 11 collection the first speech data as upstream voice data.
In order to ensure the second speech data for replacing is identical with the content in the first speech data being replaced, this
Bright embodiment, while opening main microphon 11 and at least one secondary microphone 12 collection speech data, records main microphon 11
With secondary microphone 12 gather speech data time, so, when needing to be replaced, can be possible based on there is distortion first
The time tag of speech data, the second speech data searching the same time replacing this first speech data, thus can ensure that
Speech data content after replacement does not change.I.e. in the present invention one specific embodiment, realizing obtaining at least one pair
During the step of the second speech data that mike 12 gathered in a moment, specifically can based on record second speech data collection
Time, obtain the second speech data that at least one secondary microphone 12 gathers in this moment.
Corresponding to the quantity of the secondary microphone 12 involved by the embodiment of the present invention, this step 202 specifically can include following
Technology contents:
1) if being provided with a secondary microphone 12 in mobile terminal 10, in step 202. can be directly by this secondary microphone 12
The speech data of collection is as second speech data.
Due to having one apart from L between main microphon 11 and secondary microphone 12 in the embodiment of the present invention, and usual sound source (example
Mouth as user) it is located near main microphon 11, also can there is a distance apart from secondary microphone 12 in such sound source, then sound exists
The transmitting procedure of this distance can reduce acoustic pressure, so that the acoustic pressure of the speech data of secondary microphone 12 collection can't exceed pair
The acoustics overload point of mike 12, can't there is the situation of distortion in the speech data that is, secondary microphone 12 gathers, thus can be true
Protect the quality of upstream voice data.
2) if being provided with multiple (two and two or more) secondary microphone 12 in mobile terminal 10, concrete in step 202
May comprise steps of:
Obtain multiple speech datas that multiple secondary microphones 12 gather in the described moment;
In multiple speech datas, the amplitude of strobe pulse modulation signals is less than the second preset value (limit2) and arteries and veins
Rush the maximum speech data of the amplitude of modulation signals;
The amplitude of the pulse code modulation signal of the speech data of selection is amplified to the 3rd preset value, obtains two voice numbers
According to.
The realization of above-mentioned steps, can multiple secondary microphones 12 collection multiple speech datas in selection signal optimal quality
Or the speech data close to preset value is final transmitted upstream voice data as second speech data, thus can ensure that
The quality of upstream voice data, it is to avoid the situation that distortion in upstream voice data occurs.
Above-mentioned involved the second preset value (limit2) also optional, adjustable numerical value, its object is to can
Select top-quality one voice that there is not distortion situation from multiple speech datas of multiple secondary microphones 12 collection
Data.In one embodiment, this second preset value is smaller than the first preset value, such as -2dBFS.
Above-mentioned involved the 3rd preset value (limit3) also optional, adjustable numerical value, and the language that will select
The process that the amplitude of the pulse code modulation signal of sound data is amplified to the 3rd preset value is the process that digital gain amplifies, mesh
The transmission quality being to ensure that this speech data, that is, strengthen this speech data send when signal to noise ratio, it is to avoid this speech data
During transmission, quality problems occur.In one embodiment, the 3rd preset value can be more than the first preset value, example
As -0.5dBFS.
Step 203, the first speech data that second speech data is replaced synchronization is carried out as upstream voice data
Transmission.
Through above-mentioned speech data monitoring, contrast, select, determine operation process after, can be from secondary microphone 12 institute
Select at least one speech data obtaining to determine the second speech data that can replace synchronization first speech data, will
The first speech data that this second speech data replaces synchronization is transmitted as final upstream voice data, thus can
Guarantee the quality of upstream voice data, it is to avoid upstream voice data has distortion.Corresponding to the speech data of conversational class,
Communication effect can be lifted, enable recipient not hear the voice call content sending words side;Corresponding to recording class speech data, can carry
Rise the effect of recording broadcasting, reduce noise.
The realization of the method for determination upstream voice data that the embodiments of the present invention are provided, by mobile terminal 10
It is provided for gathering a main microphon 11 of the speech data that sound source produces and at least one secondary microphone 12, this main microphon 11
There is and at least one secondary microphone 12 between a distance, and monitor the pulse of the first speech data of main microphon 11 collection and compile
Whether the amplitude of code modulated signal is more than the first preset value;If the width of the pulse code modulation signal of moment first speech data
Degree is more than the first preset value, obtains the second speech data that at least one secondary microphone 12 described gathers in the described moment;By institute
The first speech data stating second speech data replacement synchronization is transmitted as upstream voice data.Thus can effectively disappear
Except the distortion situation of upstream voice data, lift the quality of upstream voice data, strengthen the use feeling for mobile terminal for the user
It is subject to.
One specific embodiment of the method for the determination upstream voice data being provided below for the embodiment of the present invention
The process of realization is described in detail.
In this embodiment, in mobile terminal 10, a main microphon 11 and two secondary microphones 12, wherein main wheat can be provided with
Gram wind 11 is arranged at mobile terminal 10 bottom one predetermined position, and two secondary microphones 12 are respectively arranged at mobile terminal 10 top
Predetermined position.
As shown in figure 4, this embodiment specifically can include:
Step 401, collection speech data timing.
Specifically, one of mobile terminal 10 main microphon 11 and two secondary microphones 12 can synchronously be opened, respectively
Collection speech data, meanwhile, starts timing to record the time that three mikes gather speech data respectively.
Step 402, is simulated gain, analog-to-digital conversion process to the speech data of collection, and stores to corresponding buffering
In device 15.
Step 403, the first speech data of real-time monitoring main microphon 11 collection, judge the pulse of this first speech data
Whether the amplitude of modulation signals is more than the first preset value, and when judged result be more than when, trigger execution step 404,
Otherwise trigger step 407, this first speech data is defined as upstream voice data and is transmitted.
Step 404, obtains secondary microphone 12 in the speech data with the first speech data synchronization collection.
By inquiring about the record information of the first speech data, determine this main microphon 11 gather this first speech data when
Between information, be then based on this temporal information, stored in query buffer 15, two secondary microphones 12 gathered in this time
Speech data.
Step 405, selects a speech data as the second voice number in the speech data of two secondary microphone 12 collections
According to.
Specifically, i.e. strobe pulse modulation signals in two speech datas of two secondary microphones 12 collection
Amplitude is less than the speech data of the amplitude maximum of the second preset value and pulse code modulation signal as second speech data.
Step 406, the second speech data selecting is replaced the first speech data of synchronization as final up
Speech data is transmitted.
The realization of above-described embodiment, can effectively eliminate the distortion situation of upstream voice data, lifting upstream voice data
Quality, strengthens the use feeling for mobile terminal for the user.
The embodiment of the present invention additionally provides a kind of mobile terminal 500, includes for gathering voice number in this mobile terminal 500
According to a main microphon and at least one secondary microphone (referring to shown in accompanying drawing 1, accompanying drawing 5 does not show place), this main microphon with least
Have one apart from L between one secondary microphone.
As shown in figure 5, this mobile terminal 500 specifically can also include:
Monitoring modular 510, for monitoring the width of the pulse code modulation signal of the first speech data of main microphon collection
Whether degree is more than the first preset value;
Acquisition module 520, if the amplitude for the pulse code modulation signal of moment first speech data is more than first
Preset value, obtains the second speech data that at least one secondary microphone gathers in this moment;
Transport module 530, for replacing the first language of synchronization by the second speech data acquired in acquisition module 520
Sound data is transmitted as upstream voice data.
In one embodiment, as shown in fig. 6, this mobile terminal 500 specifically can also include:
Timing module 540, gathers the time of speech data for recording main microphon and at least one secondary microphone.
The acquisition time of the second speech data based on record for the acquisition module 520, obtains at least one secondary microphone one
The second speech data of moment collection.
In one embodiment, as shown in fig. 7, acquisition module 520 specifically can include:
Acquiring unit 521, multiple languages that the multiple secondary microphones for obtaining mobile terminal 500 setting gathered in a moment
Sound data.
Specifically, acquiring unit 521 obtains the voice number of its storage by read buffers 15 (as shown in Figure 3)
According to.
Select unit 522, the amplitude for strobe pulse modulation signals in multiple speech datas is pre- less than second
If the maximum speech data of the amplitude of value and pulse code modulation signal, this second preset value is less than the first preset value.
Amplifying unit 523, the amplitude for the pulse code modulation signal of the speech data selecting select unit 522 is put
Greatly to the 3rd preset value, obtain second speech data, the 3rd preset value is more than the first preset value.
In one embodiment, analog gain module 13 as shown in Figure 3 can specifically be included in mobile terminal 500
(the speech data simulation gain process for main microphon and secondary microphone collection), analog-to-digital conversion module 14 (are used for
Speech data after analog gain module 13 is processed carries out analog digital conversion, obtains main microphon and the voice of secondary microphone collection
The pulse code modulation signal of data), and buffer 15 is (for storing the voice of main microphon and secondary microphone collection respectively
The pulse code modulation signal of data).
The mobile terminal 500 that the embodiment of the present invention is provided is capable of involved in the embodiment of the method shown in Fig. 1 to Fig. 4
And each process, for avoid repeat, repeat no more here.
The mobile terminal 500 that the embodiment of the present invention is provided can effectively eliminate the distortion situation of upstream voice data, lifting
The quality of upstream voice data, strengthens the use feeling for mobile terminal for the user.
Fig. 8 is the block diagram of the mobile terminal 800 of another embodiment of the present invention.Mobile terminal 800 shown in Fig. 8 includes:
At least one processor 801, memorizer 802, at least one network interface 804 and user interface 803.In mobile terminal 800
Each assembly is coupled by bus system 805.It is understood that bus system 805 is used for realizing the company between these assemblies
Connect letter.Bus system 805, in addition to including data/address bus, also includes power bus, controlling bus and status signal bus in addition.But
It is for the sake of clear explanation, in fig. 8 various buses are all designated as bus system 805.
Wherein, user interface 803 can include display, keyboard or pointing device (for example, mouse, trace ball
(trackball), touch-sensitive plate or touch screen etc..
It is appreciated that the memorizer 802 in the embodiment of the present invention can be volatile memory or nonvolatile memory,
Or may include volatibility and nonvolatile memory.Wherein, nonvolatile memory can be read only memory (Read-
Only Memory, ROM), programmable read only memory (Programmable ROM, PROM), the read-only storage of erasable programmable
Device (Erasable PROM, EPROM), Electrically Erasable Read Only Memory (Electrically EPROM, EEPROM) or
Flash memory.Volatile memory can be random access memory (Random Access Memory, RAM), and it is used as outside high
Speed caching.By exemplary but be not restricted explanation, the RAM of many forms can use, such as static RAM
(Static RAM, SRAM), dynamic random access memory (Dynamic RAM, DRAM), Synchronous Dynamic Random Access Memory
(Synchronous DRAM, SDRAM), double data speed synchronous dynamic RAM (Double Data Rate
SDRAM, DDRSDRAM), enhancement mode Synchronous Dynamic Random Access Memory (Enhanced SDRAM, ESDRAM), synchronized links
Dynamic random access memory (Synch link DRAM, SLDRAM) and direct rambus random access memory (Direct
Rambus RAM, DRRAM).The memorizer 802 of system and method described herein be intended to including but not limited to these and arbitrarily its
It is suitable for the memorizer of type.
In some embodiments, memorizer 802 stores following element, executable module or data structure, or
Their subset of person, or their superset:Operating system 8011 and application program 8012.
Wherein, operating system 8011, comprise various system programs, such as ccf layer, core library layer, driving layer etc., are used for
Realize various basic businesses and process hardware based task.Application program 8012, comprises various application programs, such as media
Player (Media Player), browser (Browser) etc., are used for realizing various applied business.Realize the embodiment of the present invention
The program of method may be embodied in application program 8012.
In embodiments of the present invention, can be by calling program or the instruction of memorizer 802 storage, specifically, application
The program of storage or instruction in program 8012, processor 801 is used for:The pulse of the first speech data of monitoring main microphon collection
Whether the amplitude of modulation signals is more than the first preset value;If the pulse code modulation of described first speech data in a moment
The amplitude of signal is more than the first preset value, obtains the second voice number that at least one secondary microphone described gathers in the described moment
According to;The first speech data that described second speech data is replaced synchronization is transmitted as upstream voice data.
The method that the embodiments of the present invention disclose can apply in processor 801, or is realized by processor 801.
Processor 801 is probably a kind of IC chip, has the disposal ability of signal.Realize during, said method each
Step can be completed by the instruction of the integrated logic circuit of the hardware in processor 801 or software form.Above-mentioned process
Device 801 can be general processor, digital signal processor (Digital Signal Processor, DSP), special integrated electricity
Road (Application Specific Integrated Circuit, ASIC), ready-made programmable gate array (Field
Programmable Gate Array, FPGA) or other PLDs, discrete gate or transistor logic,
Discrete hardware components.Can realize or execute disclosed each method in the embodiment of the present invention, step and logic diagram.General
Processor can be microprocessor or this processor can also be any conventional processor etc..In conjunction with embodiment of the present invention institute
The step of disclosed method can be embodied directly in hardware decoding processor execution and complete, or with the hardware in decoding processor
And software module combination execution completes.Software module may be located at random access memory, and flash memory, read only memory are programmable read-only
In the ripe storage medium in this area such as memorizer or electrically erasable programmable memory, depositor.This storage medium is located at
Memorizer 802, processor 801 reads the information in memorizer 802, the step completing said method in conjunction with its hardware.
It is understood that embodiments described herein can with hardware, software, firmware, middleware, microcode or its
Combine and to realize.Hardware is realized, processing unit can be implemented in one or more special IC (Application
Specific Integrated Circuits, ASIC), digital signal processor (Digital Signal Processing,
DSP), digital signal processing appts (DSP Device, DSPD), programmable logic device (Programmable Logic
Device, PLD), field programmable gate array (Field-Programmable Gate Array, FPGA), general processor,
Controller, microcontroller, microprocessor, for executing in other electronic units or a combination thereof of the application function.
Software is realized, can be realized herein by executing the module (such as process, function etc.) of function described herein
Described technology.Software code is storable in memorizer and passes through computing device.Memorizer can within a processor or
Realize outside processor.
Alternatively, processor 801 is additionally operable to:Record main microphon and at least one secondary microphone gather speech data when
Between, and the acquisition time of the second speech data based on record, obtain at least one secondary microphone in moment collection described in
Second speech data.
Alternatively, processor 801 is additionally operable to:Obtain multiple speech datas that multiple secondary microphones gather in the described moment;
In the plurality of speech data, the amplitude of strobe pulse modulation signals is less than the second preset value and pulse code modulation letter
Number the maximum speech data of amplitude, described second preset value is less than described first preset value;Arteries and veins by the speech data selecting
The amplitude rushing modulation signals is amplified to the 3rd preset value, obtains described second speech data, and described 3rd preset value is more than
Described first preset value.
Alternatively, processor 801 is additionally operable to:Gain is simulated to the speech data of main microphon and secondary microphone collection
Process;Analog-to-digital conversion process is carried out to the speech data being simulated gain process, obtains main microphon and secondary microphone collection
Speech data pulse code modulation signal;Pulse code by described main microphon and the speech data of secondary microphone collection
Modulated signal stores respectively to corresponding buffer.
Mobile terminal 800 is capable of each process that in previous embodiment, mobile terminal 500 is realized, for avoiding repeating,
Here repeat no more.
The mobile terminal 800 that the embodiment of the present invention is provided can effectively eliminate the distortion situation of upstream voice data, lifting
The quality of upstream voice data, strengthens the use feeling for mobile terminal for the user.
Fig. 9 is the structural representation of the mobile terminal 900 of another embodiment of the present invention.Specifically, the mobile end in Fig. 9
End 900 can be mobile phone, panel computer, personal digital assistant (Personal Digital Assistant, PDA) or vehicle-mounted
Computer etc..
Mobile terminal 900 in Fig. 9 includes radio frequency (Radio Frequency, RF) circuit 910, memorizer 920, input
Unit 930, display unit 940, processor 960, voicefrequency circuit 970, WiFi (Wireless Fidelity) module 980 and electricity
Source 990.
Wherein, input block 930 can be used for numeral or the character information of receiving user's input, and produces and mobile terminal
900 user setup and the relevant signal input of function control.Specifically, in the embodiment of the present invention, this input block 930 can
To include contact panel 931.Contact panel 931, also referred to as touch screen, can collect user thereon or neighbouring touch operation
(such as user uses the operation on contact panel 931 of any suitable object such as finger, stylus or adnexa), and according in advance
The formula setting drives corresponding attachment means.Optionally, contact panel 931 may include touch detecting apparatus and touch controller
Two parts.Wherein, touch detecting apparatus detect the touch orientation of user, and detect the signal that touch operation brings, by signal
Send touch controller to;Touch controller receives touch information from touch detecting apparatus, and is converted into contact coordinate,
Give this processor 960 again, and can the order sent of receiving processor 960 being executed.Furthermore, it is possible to using resistance-type,
The polytypes such as condenser type, infrared ray and surface acoustic wave realize contact panel 931.Except contact panel 931, input block
930 can also include other input equipments 932, and other input equipments 932 can include but is not limited to physical keyboard, function key
One or more of (such as volume control button, switch key etc.), trace ball, mouse, action bars etc..
Wherein, display unit 940 can be used for showing by the information of user input or the information and the movement that are supplied to user
The various menu interfaces of terminal 900.Display unit 940 may include display floater 941, optionally, can adopt LCD or organic
The forms such as optical diode (Organic Light-Emitting Diode, OLED) are configuring display floater 941.
It should be noted that contact panel 931 can cover display floater 941, form touch display screen, when the inspection of this touch display screen
Measure thereon or after neighbouring touch operation, send processor 960 to determine the type of touch event, with preprocessor
960 provide corresponding visual output according to the type of touch event in touch display screen.
Touch display screen includes Application Program Interface viewing area and conventional control viewing area.This Application Program Interface viewing area
And the arrangement mode of this conventional control viewing area does not limit, can for being arranged above and below, left-right situs etc. can distinguish two and show
Show the arrangement mode in area.This Application Program Interface viewing area is displayed for the interface of application program.Each interface is permissible
Comprise the interface elements such as icon and/or the widget desktop control of at least one application program.This Application Program Interface viewing area
It can also be the empty interface not comprising any content.This conventional control viewing area is used for showing the higher control of utilization rate, for example,
Application icons such as settings button, interface numbering, scroll bar, phone directory icon etc..
Wherein processor 960 is the control centre of mobile terminal 900, using various interfaces and connection whole mobile phone
Various pieces, by running or executing software program and/or the module being stored in first memory 921, and call storage
Data in second memory 922, the various functions of execution mobile terminal 900 and processing data, thus to mobile terminal 900
Carry out integral monitoring.Optionally, processor 960 may include one or more processing units.
In embodiments of the present invention, by call store this first memory 921 in software program and/or module and/
Or the data in this second memory 922, processor 960 is used for:The pulse of the first speech data of monitoring main microphon collection
Whether the amplitude of modulation signals is more than the first preset value;If the pulse code modulation of described first speech data in a moment
The amplitude of signal is more than the first preset value, obtains the second voice number that at least one secondary microphone described gathers in the described moment
According to;The first speech data that described second speech data is replaced synchronization is transmitted as upstream voice data.
Alternatively, processor 960 is additionally operable to:Record main microphon and at least one secondary microphone gather speech data when
Between, and the acquisition time of the second speech data based on record, obtain at least one secondary microphone in moment collection described in
Second speech data.
Alternatively, processor 960 is additionally operable to:Obtain multiple speech datas that multiple secondary microphones gather in the described moment;
In the plurality of speech data, the amplitude of strobe pulse modulation signals is less than the second preset value and pulse code modulation letter
Number the maximum speech data of amplitude, described second preset value is less than described first preset value;Arteries and veins by the speech data selecting
The amplitude rushing modulation signals is amplified to the 3rd preset value, obtains described second speech data, and described 3rd preset value is more than
Described first preset value.
Alternatively, processor 960 is additionally operable to:Gain is simulated to the speech data of main microphon and secondary microphone collection
Process;Analog-to-digital conversion process is carried out to the speech data being simulated gain process, obtains main microphon and secondary microphone collection
Speech data pulse code modulation signal;Pulse code by described main microphon and the speech data of secondary microphone collection
Modulated signal stores respectively to corresponding buffer.
Mobile terminal 900 is capable of each process that in previous embodiment, mobile terminal 500 is realized, for avoiding repeating,
Here repeat no more.
It can be seen that, the mobile terminal 900 that the embodiment of the present invention is provided can effectively eliminate the distortion situation of upstream voice data,
The quality of lifting upstream voice data, strengthens the use feeling for mobile terminal for the user.
Those of ordinary skill in the art are it is to be appreciated that combine the list of each example of the embodiments described herein description
Unit and algorithm steps, being capable of being implemented in combination in electronic hardware or computer software and electronic hardware.These functions are actually
To be executed with hardware or software mode, the application-specific depending on technical scheme and design constraint.Professional and technical personnel
Each specific application can be used different methods to realize described function, but this realization is it is not considered that exceed
The scope of the present invention.
Those skilled in the art can be understood that, for convenience and simplicity of description, the system of foregoing description,
Device and the specific work process of unit, may be referred to the corresponding process in preceding method embodiment, will not be described here.
It should be understood that disclosed apparatus and method in embodiment provided herein, can pass through other
Mode is realized.For example, device embodiment described above is only schematically, for example, the division of described unit, it is only
A kind of division of logic function, actual can have other dividing mode when realizing, for example multiple units or assembly can in conjunction with or
Person is desirably integrated into another system, or some features can be ignored, or does not execute.Another, shown or discussed is mutual
Between coupling or direct-coupling or communication connection can be by some interfaces, the INDIRECT COUPLING of device or unit or communication link
Connect, can be electrical, mechanical or other forms.
The unit illustrating as separating component can be or may not be physically separate, show as unit
Part can be or may not be physical location, you can with positioned at a place, or can also be distributed to multiple networks
On unit.The purpose to realize this embodiment scheme for some or all of unit therein can be selected according to the actual needs.
In addition, can be integrated in a processing unit in each functional unit in each embodiment of the present invention it is also possible to
It is that unit is individually physically present it is also possible to two or more units are integrated in a unit.
If described function realized using in the form of SFU software functional unit and as independent production marketing or use when, permissible
It is stored in a computer read/write memory medium.Based on such understanding, technical scheme is substantially in other words
Partly being embodied in the form of software product of part that prior art is contributed or this technical scheme, this meter
Calculation machine software product is stored in a storage medium, including some instructions with so that a computer equipment (can be individual
People's computer, server, or network equipment etc.) execution each embodiment methods described of the present invention all or part of step.
And aforesaid storage medium includes:USB flash disk, portable hard drive, ROM, RAM, magnetic disc or CD etc. are various can be with store program codes
Medium.
The above, the only specific embodiment of the present invention, but protection scope of the present invention is not limited thereto, and any
Those familiar with the art the invention discloses technical scope in, change or replacement can be readily occurred in, all should contain
Cover within protection scope of the present invention.Therefore, protection scope of the present invention should be defined by scope of the claims.
Claims (10)
1. a kind of method determining upstream voice data, is applied to mobile terminal it is characterised in that described mobile terminal includes using
The main microphon of speech data producing in collection sound source and at least one secondary microphone, described main microphon with described at least
Between one secondary microphone, there is a distance;Methods described includes:
Whether the amplitude of the pulse code modulation signal of the first speech data of monitoring main microphon collection is more than the first preset value;
If the amplitude of the pulse code modulation signal of the first speech data described in the moment is more than the first preset value, described in acquisition extremely
The second speech data that a few secondary microphone gathered in the described moment;
The first speech data that described second speech data is replaced synchronization is transmitted as upstream voice data.
2. the method for claim 1 is it is characterised in that first speech data in described monitoring main microphon collection
Whether the amplitude of pulse code modulation signal also includes more than before the step of the first preset value:
Record described main microphon and at least one secondary microphone described gathers the time of speech data;
The described step obtaining the second speech data that at least one secondary microphone described gathers in the described moment includes:
The acquisition time of the described second speech data based on record, obtains at least one secondary microphone described and adopts in the described moment
The second speech data of collection.
If 3. the method for claim 1 is it is characterised in that described mobile terminal includes multiple secondary microphones, described
The step obtaining the second speech data that at least one secondary microphone described gathers in the described moment includes:
Obtain multiple speech datas that multiple secondary microphones gather in the described moment;
In the plurality of speech data, the amplitude of strobe pulse modulation signals is less than the second preset value and pulse code is adjusted
The maximum speech data of the amplitude of signal processed, described second preset value is less than described first preset value;
The amplitude of the pulse code modulation signal of the speech data of selection is amplified to the 3rd preset value, obtains described second voice
Data, described 3rd preset value is more than described first preset value.
4. the method for claim 1 is it is characterised in that first speech data in described monitoring main microphon collection
Before whether the amplitude of pulse code modulation signal is more than the step of the first preset value, also include:
Gain process is simulated to the speech data of main microphon and secondary microphone collection;
To be simulated gain process speech data carry out analog-to-digital conversion process, obtain main microphon and secondary microphone collection
The pulse code modulation signal of speech data;
The pulse code modulation signal of described main microphon and the speech data of secondary microphone collection is stored respectively to corresponding
In buffer.
5. the method for claim 1 it is characterised in that described main microphon mobile terminal set location apart from institute
State sound source distance be less than at least one secondary microphone described mobile terminal set location apart from described sound source distance.
6. a kind of mobile terminal is it is characterised in that described mobile terminal includes of the speech data for gathering sound source generation
Main microphon and at least one secondary microphone, have a distance between described main microphon and at least one secondary microphone described;
Described mobile terminal also includes:
Monitoring modular, whether the amplitude for monitoring the pulse code modulation signal of the first speech data of main microphon collection is big
In the first preset value;
Acquisition module, if the amplitude for the pulse code modulation signal of the first speech data described in the moment is preset more than first
Value, obtains the second speech data that at least one secondary microphone described gathers in the described moment;
Transport module, for replacing the first speech data of synchronization as upstream voice data using described second speech data
It is transmitted.
7. mobile terminal as claimed in claim 6 is it is characterised in that also include:
Timing module, gathers the time of speech data for recording described main microphon and at least one secondary microphone described;
The acquisition time of the described second speech data based on record for the described acquisition module, obtains at least one secondary microphone described
Second speech data in the collection of described moment.
8. mobile terminal as claimed in claim 6 is it is characterised in that described mobile terminal includes multiple secondary microphones, institute
State acquisition module to include:
Acquiring unit, for obtaining multiple speech datas that multiple secondary microphones gather in the described moment;
Select unit, the amplitude for strobe pulse modulation signals in the plurality of speech data is less than the second preset value
And the speech data of the amplitude maximum of pulse code modulation signal, described second preset value is less than described first preset value;
Amplifying unit, the amplitude for the pulse code modulation signal of the speech data selecting described select unit is amplified to
Three preset values, obtain described second speech data, and described 3rd preset value is more than described first preset value.
9. mobile terminal as claimed in claim 6 is it is characterised in that also include:
Analog gain module, for the speech data simulation gain process to main microphon and secondary microphone collection;
Analog-to-digital conversion module, for carrying out analog digital conversion to the speech data after described analog gain resume module, obtains main wheat
The pulse code modulation signal of the speech data of gram wind and secondary microphone collection;
Buffer, for storing the pulse code modulation letter of the speech data of described main microphon and secondary microphone collection respectively
Number.
10. mobile terminal as claimed in claim 6 it is characterised in that described main microphon mobile terminal set location
It is less than the set location in mobile terminal at least one secondary microphone described apart from the distance of described sound source apart from described sound source
Distance.
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Cited By (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN111432160A (en) * | 2020-04-01 | 2020-07-17 | 广州市百果园信息技术有限公司 | Method, device, server and storage medium for realizing multi-person call |
CN112333608A (en) * | 2018-07-26 | 2021-02-05 | Oppo广东移动通信有限公司 | Voice data processing method and related product |
CN112584293A (en) * | 2018-07-26 | 2021-03-30 | Oppo广东移动通信有限公司 | Microphone hole blockage detection method and related product |
CN113992783A (en) * | 2021-11-05 | 2022-01-28 | 珠海格力电器股份有限公司 | Network communication method, device, storage medium and user equipment |
Citations (6)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN101447771A (en) * | 2008-12-24 | 2009-06-03 | 北京中星微电子有限公司 | Method and system for automatically controlling gains |
CN101771913A (en) * | 2009-09-28 | 2010-07-07 | 瑞声声学科技(深圳)有限公司 | Method for controlling bass sound reproduction of audio frequency signal and method |
JP2012074976A (en) * | 2010-09-29 | 2012-04-12 | Nec Casio Mobile Communications Ltd | Mobile terminal, mobile system, and warning method |
KR20130064243A (en) * | 2011-12-08 | 2013-06-18 | 삼성전자주식회사 | Method and apparatus for processing audio in mobile terminal |
CN105679356A (en) * | 2014-11-17 | 2016-06-15 | 中兴通讯股份有限公司 | Recording method, device and terminal |
CN105702264A (en) * | 2015-12-30 | 2016-06-22 | 努比亚技术有限公司 | Audio processing device and method |
-
2016
- 2016-08-31 CN CN201610798983.7A patent/CN106385489B/en active Active
Patent Citations (6)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN101447771A (en) * | 2008-12-24 | 2009-06-03 | 北京中星微电子有限公司 | Method and system for automatically controlling gains |
CN101771913A (en) * | 2009-09-28 | 2010-07-07 | 瑞声声学科技(深圳)有限公司 | Method for controlling bass sound reproduction of audio frequency signal and method |
JP2012074976A (en) * | 2010-09-29 | 2012-04-12 | Nec Casio Mobile Communications Ltd | Mobile terminal, mobile system, and warning method |
KR20130064243A (en) * | 2011-12-08 | 2013-06-18 | 삼성전자주식회사 | Method and apparatus for processing audio in mobile terminal |
CN105679356A (en) * | 2014-11-17 | 2016-06-15 | 中兴通讯股份有限公司 | Recording method, device and terminal |
CN105702264A (en) * | 2015-12-30 | 2016-06-22 | 努比亚技术有限公司 | Audio processing device and method |
Cited By (6)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN112333608A (en) * | 2018-07-26 | 2021-02-05 | Oppo广东移动通信有限公司 | Voice data processing method and related product |
CN112584293A (en) * | 2018-07-26 | 2021-03-30 | Oppo广东移动通信有限公司 | Microphone hole blockage detection method and related product |
CN112333608B (en) * | 2018-07-26 | 2022-03-22 | Oppo广东移动通信有限公司 | Voice data processing method and related product |
CN112584293B (en) * | 2018-07-26 | 2022-04-08 | Oppo广东移动通信有限公司 | Microphone hole blockage detection method and related product |
CN111432160A (en) * | 2020-04-01 | 2020-07-17 | 广州市百果园信息技术有限公司 | Method, device, server and storage medium for realizing multi-person call |
CN113992783A (en) * | 2021-11-05 | 2022-01-28 | 珠海格力电器股份有限公司 | Network communication method, device, storage medium and user equipment |
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