CN106653039A - Audio signal processing system and audio signal processing method - Google Patents
Audio signal processing system and audio signal processing method Download PDFInfo
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- CN106653039A CN106653039A CN201611111467.9A CN201611111467A CN106653039A CN 106653039 A CN106653039 A CN 106653039A CN 201611111467 A CN201611111467 A CN 201611111467A CN 106653039 A CN106653039 A CN 106653039A
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- 230000005236 sound signal Effects 0.000 title claims abstract description 122
- 238000003672 processing method Methods 0.000 title claims abstract description 13
- 230000002452 interceptive effect Effects 0.000 claims abstract description 15
- 230000009467 reduction Effects 0.000 claims description 20
- 230000008878 coupling Effects 0.000 claims description 19
- 238000010168 coupling process Methods 0.000 claims description 19
- 238000005859 coupling reaction Methods 0.000 claims description 19
- 238000011946 reduction process Methods 0.000 claims description 15
- 230000000052 comparative effect Effects 0.000 claims description 12
- 238000000034 method Methods 0.000 claims description 11
- 230000003993 interaction Effects 0.000 claims description 10
- 230000008859 change Effects 0.000 claims description 8
- 230000008569 process Effects 0.000 claims description 6
- 230000003044 adaptive effect Effects 0.000 abstract description 2
- 230000001603 reducing effect Effects 0.000 abstract 3
- 238000003491 array Methods 0.000 description 5
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Classifications
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
-
- 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
- H04M1/03—Constructional features of telephone transmitters or receivers, e.g. telephone hand-sets
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/08—Mouthpieces; Microphones; Attachments therefor
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
Abstract
According to the embodiment, the invention provides an audio signal processing system and an audio signal processing method. The system comprises a first microphone array and a second microphone array which are oppositely arranged at intervals, a connecting component, a controller and a noise reducing component, wherein a main microphone port and an auxiliary microphone port are formed in the controller; an input end of the noise reducing component is connected to the controller; by virtue of the connecting component, a current input connecting relation is kept between the first microphone array and the main microphone port as well as between the second microphone array and the auxiliary microphone port; and by virtue of the controller, intensity of a first audio signal which is received from the main microphone port and intensity of a second audio signal which is received from the auxiliary microphone port are compared, and the input connecting relation is kept or changed by virtue of the connecting component in accordance with a comparison result, so that the main and auxiliary microphone ports are switched to input signals in an adaptive mode in accordance with the audio signal intensity so as to adapt to a current intelligent voice interactive scene and to guarantee a good noise reducing effect; therefore, the processing system and the processing method are conducive to the improvement of the accuracy of a speed recognition result.
Description
Technical field
The present invention relates to signal processing technology field, more particularly to a kind of audio signal processing and Audio Signal Processing
Method.
Background technology
Interactive voice, the such as every field lived throughout us, mobile phone, TV, vehicle-mounted, air-conditioning field.Energy
The premise for enough carrying out normal voice interaction is to accurately identify voice signal.By taking mobile phone terminal as an example, due to the presence of ambient noise,
The microphone of mobile phone terminal can also collect ambient noise signal, these rings while effective sound-source signal of user is gathered
Border noise signal can be interfered to accurately identifying for effective sound-source signal, accordingly, it would be desirable to the voice letter collected to microphone
Noise reduction process number is carried out, impact of the noise to sound-source signal is reduced.
A kind of existing relatively conventional noise reduction process mode is to carry out noise reduction process using dual microphone.Dual microphone drops
Main thought of making an uproar is two microphones of setting in terminal, and ideally the main microphon signal of main microphon collection is band
The voice signal of border noise, the auxiliary microphone signal of auxiliary microphone collection only includes ambient noise, and the collection of main and auxiliary microphone
Ambient noise characteristic is consistent, and both subtract each other, and obtain pure sound-source signal.
At present, in the scenes such as mobile phone terminal, when using mobile phone terminal, sound source orientation is relatively fixed user,
So as to dual microphone position is fixed, i.e., main microphon is set at the position that user speaks, is set away from the position that user speaks
Put auxiliary microphone.But, with the continuous appearance of various intelligent interaction products, the scene of interactive voice there occurs very big change,
The orientation of the relatively intelligent interactive product of sound source no longer immobilizes, now, be fixedly installed certain microphone as main microphon,
Another microphone has been unable to the demand of flexible adaptation intelligent sound interaction scenarios as auxiliary microphone, it is likely that cause sound source to be believed
Number identification accuracy substantially reduce.
The content of the invention
In view of this, the embodiment of the present invention provides a kind of audio signal processing and acoustic signal processing method, can
Adaptive voice interaction scenarios, are favorably improved the accuracy of voice identification result.
The embodiment of the present invention provides a kind of audio signal processing, including：
The first microphone array for dorsad arranging and second microphone array, coupling assembly, controller, and noise reduction group
Part；Wherein,
The setting space of first microphone array and the second microphone array is more than predeterminable range；
Main microphone interface and auxiliary microphone interface are provided with the controller；
The input of the noise reduction components is connected with the controller；
The coupling assembly, for making first microphone array and the second microphone array and the main Mike
Interface and the auxiliary microphone interface have current input annexation；
The controller, for the first audio signal received from the main microphone interface with connect from the auxiliary Mike
The second audio signal that mouth is received carries out signal strength signal intensity comparison, is kept or is changed by the coupling assembly according to comparative result
The input annexation.
The embodiment of the present invention provides a kind of acoustic signal processing method, including：
The first audio signal from the input of the first input interface is received, and from the second audio frequency of the second input interface input
Signal；
Signal strength signal intensity comparison is carried out to first audio signal and second audio signal；
According to the comparative result of the signal strength signal intensity, first audio signal and second audio frequency letter are kept or switched
Number input interface；
The audio signal being input into the audio signal being input into from first input interface and from second input interface
Carry out noise reduction process.
Audio signal processing provided in an embodiment of the present invention and acoustic signal processing method, the system is included dorsad
Spaced first microphone array and second microphone array, coupling assembly, controller, and noise reduction components, controller
On be provided with main microphone interface and auxiliary microphone interface.Assume that current coupling assembly causes the first microphone array and second microphone
Array has certain input annexation, under the input annexation, controller pair with main microphone interface and auxiliary microphone interface
It is strong signal to be carried out from main microphone interface the first audio signal for receiving and the second audio signal received from auxiliary microphone interface
Degree compares, and to control coupling assembly according to comparative result input annexation is kept or change.That is, if it find that at this
Under input annexation, second from auxiliary Mike input is less than from the signal strength signal intensity of the first audio signal of main microphone interface input
The signal strength signal intensity of audio signal, then illustrate that the microphone array being connected with main microphone interface should not be reconnected and connect in main Mike
Mouthful, it should switching is connected to auxiliary microphone interface, so as to switch the annexation of two microphone arrays and main and auxiliary microphone interface,
It is achieved thereby that main and auxiliary microphone interface input signal is adaptively switched according to audio signal strength, with the current intelligence of self adaptation
Energy interactive voice scene, also ensure that good noise reduction, be favorably improved the accuracy of voice identification result.
Description of the drawings
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
The accompanying drawing to be used needed for having technology description is briefly described, it should be apparent that, drawings in the following description are these
Some bright embodiments, for those of ordinary skill in the art, on the premise of not paying creative work, can be with root
Other accompanying drawings are obtained according to these accompanying drawings.
Fig. 1 is the structural representation of audio signal processing embodiment one provided in an embodiment of the present invention；
Fig. 2 is the structural representation of audio signal processing embodiment two provided in an embodiment of the present invention；
Fig. 3 is the flow chart of acoustic signal processing method embodiment one provided in an embodiment of the present invention.
Specific embodiment
To make purpose, technical scheme and the advantage of the embodiment of the present invention clearer, below in conjunction with the embodiment of the present invention
In accompanying drawing, the technical scheme in the embodiment of the present invention is clearly and completely described, it is clear that described embodiment is
The a part of embodiment of the present invention, rather than the embodiment of whole.Based on the embodiment in the present invention, those of ordinary skill in the art
The every other embodiment obtained under the premise of creative work is not made, belongs to the scope of protection of the invention.
The term for using in embodiments of the present invention is, only merely for the purpose of description specific embodiment, and to be not intended to be limiting
The present invention." one kind ", " described " and " being somebody's turn to do " of singulative used in the embodiment of the present invention and appended claims
It is also intended to include most forms, unless context clearly shows that other implications, " various " generally comprise at least two, but not
Exclude and include at least one situation.
It should be appreciated that term "and/or" used herein is only a kind of incidence relation of description affiliated partner, represent
There may be three kinds of relations, for example, A and/or B can be represented：Individualism A, while there is A and B, individualism B these three
Situation.In addition, character "/" herein, typicallys represent forward-backward correlation pair as if a kind of relation of "or".
It will be appreciated that though XXX may be described in embodiments of the present invention using term first, second, third, etc., but
These XXX should not necessarily be limited by these terms.These terms are only used for that XXX is distinguished from each other out.For example, without departing from present invention enforcement
In the case of example scope, an XXX can also be referred to as the 2nd XXX, and similarly, the 2nd XXX can also be referred to as an XXX.
Depending on linguistic context, word as used in this " if ", " if " can be construed to " ... when " or
" when ... " or " in response to determining " or " in response to detection ".Similarly, depending on linguistic context, phrase " if it is determined that " or " such as
Fruit detection (condition or event of statement) " can be construed to " when it is determined that when " or " in response to determine " or " when detection (statement
Condition or event) when " or " in response to detect (condition or event of statement) ".
Also, it should be noted that term " including ", "comprising" or its any other variant are intended to nonexcludability
Comprising, so that not only include those key elements including the commodity or system of a series of key elements, but also including without clear and definite
Other key elements listed, or also include the key element intrinsic for this commodity or system.In the feelings without more restrictions
Under condition, the key element limited by sentence "including a ...", it is not excluded that in the commodity or system including the key element also
There is other identical element.
Fig. 1 is the structural representation of audio signal processing embodiment one provided in an embodiment of the present invention, such as Fig. 1 institutes
Show, the system includes：
The first microphone array 1 for dorsad arranging and second microphone array 2, coupling assembly 3, controller 4, and noise reduction
Component 5.
Wherein, have one to ensure the audio signal that the first microphone array 1 and second microphone array 2 are gathered respectively
The setting space of fixed discrimination, the first microphone array 1 and second microphone array 2 is needed more than predeterminable range, and this is preset
Distance can set according to actual application environment.
First microphone array 1 and second microphone battle array 2 have identical array structure, it is alternatively possible to using existing
The arbitrary array structure provided in technology, such as can be linear microphone array.
In addition, in order to ensure that the two microphone arrays can realize that the omnidirectional of audio signal gathers i.e. as an entirety
360 degree of collections, while, it is ensured that single microphone array has certain sound directivity, it is to avoid uncorrelated ambient noise is to it
The excessive interference of collection audio signal, can arrange and make the first microphone array 1 and second microphone array 2 be covered each by 180
The audio collection scope of degree.
The above-mentioned audio signal processing provided in the present embodiment goes in various intelligent interaction products, this reality
In applying example, as a example by applying in intelligent mobile robot.Now, the first microphone array 1 and second microphone array 2 can be with
Be arranged on robot fuselage, apart from the mutually level position in ground, such as：First microphone array 1 and second microphone battle array
Row 2 can be arranged on the front and rear sides of robot head.
After the robot is activated voice interactive function, the first microphone array 1 and second microphone array 2 are used respectively
In collection user mutual voice.
As shown in figure 1, the first microphone array 1 and the second wheat as the audio collection device of collection external audio signal
Gram wind array 2 is connected by a coupling assembly 3 with controller 4.Specifically, the He of main microphone interface 41 is provided with controller 4
Auxiliary microphone interface 42, coupling assembly 3 causes the first microphone array 1 and second microphone array 2 and main microphone interface 41 and auxiliary
Microphone interface 42 has current input annexation.
Wherein, main microphone interface 41 and auxiliary microphone interface 42 can be considered to be and the two microphone arrays are gathered respectively
Audio signal signal attribute sign.Specifically, the audio signal being input into from main microphone interface 41 can be considered as sound
Sound source signal, i.e., containing the more signals of efficient voice composition, by the audio signal being input into from auxiliary microphone interface 42 noise is considered as
Signal.
Wherein, current input annexation can be certain annexation, or upper for setting in default in advance
The annexation adopted during secondary interactive voice.As an example it is assumed that current input annexation is, coupling assembly 3 is caused
First microphone array 1 is connected with main microphone interface 41, and second microphone array 2 is connected with auxiliary microphone interface 42.So now,
Controller 4 will receive the audio signal that the first microphone array 1 is gathered from main microphone interface 41, receive from auxiliary microphone interface 42
To the audio signal of the collection of second microphone array 2.
Under above-mentioned current input annexation, if now user triggers phonetic entry, the first microphone array
Row 1 and second microphone array 2 collect respectively audio signal, and based on current input annexation, the two microphones
Array is by the audio signal input controller for each collecting, now, 4 pairs of first received from main microphone interface 41 of controller
Audio signal carries out signal strength signal intensity and compares with the second audio signal received from auxiliary microphone interface 42, is passed through according to comparative result
Coupling assembly 3 keeps or changes the current input annexation.
In the example above, the first audio signal now is the audio signal that the first microphone array 1 is collected, second
Audio signal is the audio signal that second microphone array 2 is collected, and controller 4 can be by the two audio signal difference
Certain signal transacting is carried out, such as amplification, filtering etc. are processed, and ask for the signal strength signal intensity of the two audio signals, are compared.
If the comparison show that signal strength signal intensity of the signal strength signal intensity of the first audio signal more than the second audio signal, explanation
Really it is now sound source signals from the first audio signal of the input of main microphone interface 41, from the second of the input of auxiliary microphone interface 42
Audio signal is noise signal, need not currently change the annexation and of the first microphone array 1 and main microphone interface 41
The annexation of two microphone arrays 2 and auxiliary microphone interface 42.
On the contrary, if the comparison show that the signal strength signal intensity of the first audio signal is strong less than the signal of the second audio signal
Degree, illustrates that the first audio signal now from the input of main microphone interface 41 should be considered as noise signal, and from auxiliary microphone interface 42
Second audio signal of input should be sound source signals, then need to change the first microphone array 1 and main microphone interface 41
The annexation of annexation and second microphone array 2 and auxiliary microphone interface 42, change into the first microphone array 1 with it is auxiliary
Microphone interface 42 connects, and second microphone array 2 is connected with main microphone interface 41.
Wherein, the input annexation change can controller 4 by controlling coupling assembly 3 realizing, specifically
Alternatively, coupling assembly 3 can be implemented as switching switch, so as to controller 4 can be it is determined that it be received from main microphone interface 41
The first audio signal signal strength signal intensity less than its second audio signal received from auxiliary microphone interface 42 signal strength signal intensity when, to
Switching switch sends switch-over control signal, and to control to switch to switch current input annexation is changed.
In the present embodiment, alternatively, controller 4 can use various application specific integrated circuits (ASIC), data signal
Processor (DSP), digital signal processing appts (DSPD), PLD (PLD), field programmable gate array
(FPGA), micro- middle control element, microprocessor or other electronic components are realized.
After the switching control of input audio signal of above-mentioned main and auxiliary microphone interface has been performed, can be dropped based on diamylose gram
Make an uproar principle, using the two-way audio signals of 5 pairs of inputs of noise reduction components noise reduction process is carried out.
Specifically, the input of noise reduction components 5 is connected with controller 4, and controller 4 can be by it constantly from main and auxiliary Mike
Interface to audio signal be input to the input of noise reduction components 5.Specifically, noise reduction components 5 typically have main signal
Input and auxiliary signal input part, main signal input is used to be input into the audio signal that main microphone interface 41 is received, auxiliary signal
Input is used to be input into the audio signal that auxiliary microphone interface 42 is received, and is input into main signal input and auxiliary signal input part
Audio signal does additive operation, completes noise reduction process.The noise reduction process process is only simplified schematic description, at actual noise reduction
Reason process may refer to the processing procedure of prior art.
What deserves to be explained is, in above-mentioned switching control strategy, the foundation for whether switching is to be based on to have passed through main and auxiliary wheat
The signal strength signal intensity of the audio signal of gram interface input, the audio signal that this has been input into is relative to defeated during an interactive voice
It is very short for all audio frequency signal for entering, therefore ideally, the part audio signal can consider and be served only for cutting
Change judgement to be used, for follow-up noise reduction, speech recognition process do not affect, i.e., will not be input in subsequent components.
In the present embodiment, it is assumed that current coupling assembly causes the first microphone array and second microphone array and main Mike
Interface and auxiliary microphone interface have certain input annexation, and under the input annexation, controller is to from main microphone interface
The first audio signal for receiving carries out signal strength signal intensity and compares with the second audio signal received from auxiliary microphone interface, with basis
Comparative result control coupling assembly keeps or changes input annexation.That is, if it find that in the input annexation
Under, it is less than the letter of the second audio signal from auxiliary Mike input from the signal strength signal intensity of the first audio signal of main microphone interface input
Number intensity, then illustrate that the microphone array being connected with main microphone interface should not be reconnected in main microphone interface, it should which switching connects
Auxiliary microphone interface is connected to, so as to switch the annexation of two microphone arrays and main and auxiliary microphone interface, it is achieved thereby that according to
Audio signal strength adaptively switches main and auxiliary microphone interface input signal, with the current intelligent sound interaction scenarios of self adaptation,
Good noise reduction is also ensure that, the accuracy of voice identification result is favorably improved.
Fig. 2 is the structural representation of audio signal processing embodiment two provided in an embodiment of the present invention, such as Fig. 2 institutes
Show, on the basis of embodiment illustrated in fig. 1, alternatively, the system also includes：
First dust-proof wind-proof device 6 and the second dust-proof wind-proof device 7.Wherein, the first microphone array 1 is anti-installed in first
In dirt wind-proof device 6, second microphone array 2 is arranged in the second dust-proof wind-proof device 7.
In the present embodiment, in order to physically ensure adverse effect of the environmental factor to voice identification result as far as possible, in Mike
The first dust-proof wind-proof device 6 and the second dust-proof wind-proof device 7 are provided on the packaging technology of wind array, with reduce as far as possible environment because
Adverse effect of the element to voice identification result.
Wherein, the knot such as windproof cotton, Air Filter is such as included in the first dust-proof wind-proof device 6 and the second dust-proof wind-proof device 7
Structure, to reduce the impact of sound of the wind, dust to microphone array.
Alternatively, the system also includes：Speech recognition component 8 and interactive component 9.
Wherein, speech recognition component 8 is connected respectively with the output end and controller 4 of noise reduction components 5, after to noise reduction
Audio signal carries out speech recognition, and voice identification result is inputed into controller 4.
Controller 4 is additionally operable to carry out corresponding interaction feedback according to voice identification result control interactive component 9.
Audio signal processing provided in an embodiment of the present invention is applied in general in the product of intelligent sound interaction, in order to
Intelligent sound interactive function is realized, after noise reduction process has been carried out by the audio signal of 5 pairs of inputs of noise reduction components, after noise reduction
Audio signal inputs to speech recognition component, to complete the voice recognition processing of user input voice.Meanwhile, in order to realize being based on
The intelligent interaction of voice, by taking robot as an example, needs to be fed back accordingly to user based on voice identification result.The present embodiment
In, by taking robot as an example, the interactive component 9 such as can be speech player, and controller 4 can be logical based on voice identification result
Speech player is crossed to user feedback response voice；Can be for another example display screen, controller 4 can be based on voice identification result
By display screen to certain business operation interface of user feedback；Moving component is can also be for another example, and controller 4 can be based on language
Sound recognition result makes robot perform corresponding feedback action, etc. by controlling moving component.
Fig. 3 is the flow chart of acoustic signal processing method embodiment one provided in an embodiment of the present invention, and the present embodiment is provided
The acoustic signal processing method can be performed by an audio signal processing, the audio signal processing can be realized
For hardware, or it is embodied as the combination of software and hardware, the audio signal processing can such as move machine with integrally disposed
In the interactive voice equipment such as people, such as it can be the system architecture shown in Fig. 1, Fig. 2.As shown in figure 3, the method includes following step
Suddenly：
The first audio signal that step 101, reception are input into from the first input interface, and from the input of the second input interface
Second audio signal.
In the present embodiment, above-mentioned first input interface connects corresponding to the main microphone interface in previous embodiment, the second input
Mouth corresponds to auxiliary microphone interface.
Step 102, signal strength signal intensity comparison is carried out to the first audio signal and the second audio signal.
Step 103, according to the comparative result of signal strength signal intensity, keep or switch the first audio signal and the second audio signal
Input interface.
Step 104, to the audio signal that is input into from the first input interface and the audio signal from the input of the second input interface
Carry out noise reduction process.
Specifically, enter to the audio signal that is input into from the first input interface and from the audio signal of the second input interface input
Row noise reduction process, including：
If according to the comparative result of signal strength signal intensity, keeping the input interface of the first audio signal and the second audio signal, then
With the second audio signal as noise signal, noise reduction process is carried out to the first audio signal as sound source signals；
If according to the comparative result of signal strength signal intensity, switching the input interface of the first audio signal and the second audio signal, then
With the first audio signal as noise signal, noise reduction process is carried out to the second audio signal as sound source signals.
The specifically applicable scene of the acoustic signal processing method that the present embodiment is provided and detailed process, may refer to aforementioned reality
The explanation in example is applied, be will not be described here.
System embodiment described above is only schematic, wherein the unit as separating component explanation
(various assemblies, device etc.) can be or may not be physically separate, and the part shown as unit can be with
It is or may not be physical location, you can with positioned at a place, or can also be distributed on multiple NEs.Can
The purpose of this embodiment scheme is realized to select some or all of module therein according to the actual needs.This area is common
Technical staff is not in the case where performing creative labour is paid, you can to understand and implement.
Through the above description of the embodiments, those skilled in the art can be understood that each embodiment can
Realize by the mode for adding required general hardware platform, naturally it is also possible to by hardware.Based on such understanding, above-mentioned skill
The part that art scheme substantially contributes in other words to prior art can be embodied in the form of product, and the computer is produced
Product can be stored in a computer-readable storage medium, such as ROM/RAM, magnetic disc, CD, including some instructions are used so that one
Platform computer installation (can be personal computer, server, either network equipment etc.) perform each embodiment or embodiment
Some parts described in method.
Finally it should be noted that：Above example only to illustrate technical scheme, rather than a limitation；Although
The present invention has been described in detail with reference to the foregoing embodiments, it will be understood by those within the art that：It still may be used
To modify to the technical scheme described in foregoing embodiments, or equivalent is carried out to which part technical characteristic；
And these modification or replace, do not make appropriate technical solution essence depart from various embodiments of the present invention technical scheme spirit and
Scope.
Claims (10)
1. a kind of audio signal processing, it is characterised in that include：
The first microphone array for dorsad arranging and second microphone array, coupling assembly, controller, and noise reduction components；Its
In,
The setting space of first microphone array and the second microphone array is more than predeterminable range；
Main microphone interface and auxiliary microphone interface are provided with the controller；
The input of the noise reduction components is connected with the controller；
The coupling assembly, for making first microphone array and the second microphone array and the main microphone interface
There is current input annexation with the auxiliary microphone interface；
The controller, for the first audio signal received from the main microphone interface with connect from the auxiliary microphone interface
The second audio signal for receiving carries out signal strength signal intensity comparison, is kept by the coupling assembly or described in changing according to comparative result
Input annexation.
2. system according to claim 1, it is characterised in that the coupling assembly includes switching switch.
3. system according to claim 2, it is characterised in that the controller specifically for：
When it is determined that the signal strength signal intensity of first audio signal is less than the signal strength signal intensity of second audio signal, cut to described
Change switch and send switch-over control signal, to control the switching switch input annexation is changed.
4. system according to claim 1, it is characterised in that first microphone array and the second microphone battle array
For linear microphone array, the audio frequency that first microphone array and the second microphone array are covered each by 180 degree is adopted
Collection scope.
5. system according to claim 1, it is characterised in that first microphone array and the second microphone battle array
Row be arranged on robot fuselage, apart from the mutually level position in ground.
6. system according to any one of claim 1 to 5, it is characterised in that also include：
First dust-proof wind-proof device and the second dust-proof wind-proof device；
First microphone array is arranged in the described first dust-proof wind-proof device, and the second microphone array is arranged on institute
State in the second dust-proof wind-proof device.
7. system according to any one of claim 1 to 5, it is characterised in that also include：
Speech recognition component and interactive component；
The speech recognition component is connected respectively with the output end and the controller of the noise reduction components, after to noise reduction
Audio signal carries out speech recognition, and voice identification result is inputed into the controller；
The controller is additionally operable to carry out corresponding interaction feedback according to institute's speech recognition result control interactive component.
8. a kind of acoustic signal processing method, it is characterised in that include：
The first audio signal from the input of the first input interface is received, and from the second audio frequency letter of the second input interface input
Number；
Signal strength signal intensity comparison is carried out to first audio signal and second audio signal；
According to the comparative result of the signal strength signal intensity, first audio signal and second audio signal are kept or switched
Input interface；
Audio signal to the audio signal being input into from first input interface and from second input interface input is carried out
Noise reduction process.
9. method according to claim 8, it is characterised in that the audio frequency letter to being input into from first input interface
Number and from second input interface input audio signal carry out noise reduction process, including：
If according to the comparative result of the signal strength signal intensity, keeping the input of first audio signal and second audio signal
Interface, then with second audio signal as noise signal, drop to first audio signal as sound source signals
Make an uproar process.
10. method according to claim 8, it is characterised in that the audio frequency to being input into from first input interface
Signal and the audio signal from second input interface input carry out noise reduction process, including：
If according to the comparative result of the signal strength signal intensity, switching the input of first audio signal and second audio signal
Interface, then with first audio signal as noise signal, drop to second audio signal as sound source signals
Make an uproar process.
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