WO2020248235A1 - 差分波束形成方法及模块、信号处理方法及装置、芯片 - Google Patents

差分波束形成方法及模块、信号处理方法及装置、芯片 Download PDF

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Publication number
WO2020248235A1
WO2020248235A1 PCT/CN2019/091307 CN2019091307W WO2020248235A1 WO 2020248235 A1 WO2020248235 A1 WO 2020248235A1 CN 2019091307 W CN2019091307 W CN 2019091307W WO 2020248235 A1 WO2020248235 A1 WO 2020248235A1
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Prior art keywords
signal
differential
beamforming
microphones
differential beamforming
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PCT/CN2019/091307
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English (en)
French (fr)
Inventor
李娜
李国梁
王鑫山
朱虎
郭红敬
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深圳市汇顶科技股份有限公司
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Application filed by 深圳市汇顶科技股份有限公司 filed Critical 深圳市汇顶科技股份有限公司
Priority to EP19926741.0A priority Critical patent/EP3783609A4/en
Priority to CN201980001065.9A priority patent/CN110383378B/zh
Priority to PCT/CN2019/091307 priority patent/WO2020248235A1/zh
Priority to US17/079,193 priority patent/US11381909B2/en
Publication of WO2020248235A1 publication Critical patent/WO2020248235A1/zh

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques 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
    • G10L21/0232—Processing in the frequency domain
    • 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
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/307—Frequency adjustment, e.g. tone control
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H04R2430/25—Array processing for suppression of unwanted side-lobes in directivity characteristics, e.g. a blocking matrix

Definitions

  • This application relates to the field of signal processing technology, and in particular to a differential beam forming method and module, signal processing method and device, and chip.
  • a microphone array is generally set to enhance the voice; the microphone array is formed by a set of microphones arranged in different positions in the space in a certain way, and can receive
  • the spatially distributed field signal is sampled to obtain the spatial discrete observation data of the signal source, and the spatial information in the data is used for algorithm processing to enhance the required speech and suppress useless interference and noise.
  • the signals of the two microphones can be processed through a differential algorithm to achieve voice signal enhancement.
  • the purpose of some embodiments of the present application is to provide a differential beamforming method and module, signal processing method and device, and chip, which can ensure the constant beam characteristics of the differential beamforming signal as far as possible for microphone arrays of different specifications.
  • the embodiment of the application provides a differential beamforming method, which includes: obtaining a differential beamforming signal according to the input signals obtained by two microphones in the microphone array; based on the distance between the two microphones and the signal frequency of the input signal at least for the differential beam The amplitude of the formed signal is adjusted nonlinearly to obtain the adjusted differential beam forming signal.
  • An embodiment of the application also provides a signal processing method, including: correcting the sound signals collected by two microphones in the microphone array to obtain an input signal; and performing differential beamforming processing on the input signal based on the above-mentioned differential beamforming method , And obtain the adjusted differential beamforming signal; perform post-filtering on the adjusted differential beamforming signal.
  • the embodiment of the application also provides a differential beamforming module, including: a differential beamforming submodule, which is used to obtain a differential beamforming signal according to the input signals obtained by two microphones in the microphone array; and the adjustment submodule is based on the difference between the two microphones.
  • the distance between the input signal and the signal frequency of the input signal is at least nonlinearly adjusted to the amplitude of the differential beamforming signal to obtain the adjusted differential beamforming signal.
  • the embodiment of the present application also provides a signal processing device, including: a correction module, used to correct the sound signals collected by two microphones in the microphone array, and obtain an input signal; the above-mentioned differential beam forming module is used to input The signal undergoes differential beamforming processing, and an adjusted differential beamforming signal is obtained; a post filtering module is used to perform post filtering on the adjusted differential beamforming signal.
  • a correction module used to correct the sound signals collected by two microphones in the microphone array, and obtain an input signal
  • the above-mentioned differential beam forming module is used to input
  • the signal undergoes differential beamforming processing, and an adjusted differential beamforming signal is obtained
  • a post filtering module is used to perform post filtering on the adjusted differential beamforming signal.
  • An embodiment of the present application also provides a chip including the above-mentioned signal processing device.
  • the embodiment of the present application also provides an electronic device, including a microphone array and the aforementioned chip; the microphone array includes at least two microphones, and the chip is connected to each microphone.
  • the embodiment of the present application obtains input signals from two microphones of a microphone array, and then obtains a differential beamforming signal based on the input signals obtained by the two microphones, and then based on the distance between the two microphones and the input signal The signal frequency of at least non-linearly adjusts the amplitude of the differential beamforming signal to obtain the adjusted differential beamforming signal.
  • this embodiment provides an adjustment method. For microphone arrays of different specifications, based on two The distance between the microphones and the signal frequency of the input signal are at least nonlinearly adjusted to the amplitude of the differential beamforming signal, so as to ensure the constant beam characteristics of the differential beamforming signal as much as possible.
  • At least the amplitude of the differential beamforming signal is nonlinearly adjusted to obtain the adjusted differential beamforming signal, including: based on the distance between the two microphones and the input
  • the signal frequency of the signal is respectively non-linearly adjusted to the amplitude of the differential beamforming signal and the phase of the differential beamforming signal is adjusted to obtain the adjusted differential beamforming signal.
  • This embodiment provides a specific implementation for at least nonlinearly adjusting the amplitude of the differential beamforming signal based on the distance between the two microphones and the signal frequency of the input signal to obtain the adjusted differential beamforming signal.
  • the amplitude of the differential beamforming signal is nonlinearly adjusted, and the phase of the differential beamforming signal is adjusted to obtain the adjusted differential beamforming signal, including: Based on the distance between the two microphones and the signal frequency of the input signal, the amplitude of the differential beamforming signal is nonlinearly adjusted, and the phase of the differential beamforming signal is linearly adjusted to obtain the adjusted differential beamforming signal.
  • This embodiment provides the non-linear adjustment of the amplitude of the differential beamforming signal and the adjustment of the phase of the differential beamforming signal based on the distance between the two microphones and the signal frequency of the input signal to obtain the adjusted differential beamforming signal. A specific way to achieve.
  • This embodiment provides the nonlinear adjustment of the amplitude of the differential beamforming signal and the linear adjustment of the phase of the differential beamforming signal based on the distance between the two microphones and the signal frequency of the input signal to obtain the adjusted differential beamforming. A specific way of signaling.
  • obtaining the differential beamforming signal according to the input signals obtained by the two microphones in the microphone array includes: determining the position of the sound source according to the input signal; determining the beamforming mode according to the sound source position; processing the input signal according to the determined beamforming mode, And output the differential beam forming signal.
  • This embodiment provides a specific implementation manner for obtaining a differential beamforming signal according to input signals obtained by two microphones in the microphone array.
  • determining the beamforming method according to the sound source position includes: if the sound source position belongs to the preset target sound source range, determining the beamforming method is the fixed differential beamforming method; if the sound source position belongs to the preset interference range, determining the beam
  • the forming method is an adaptive differential beam forming method. This embodiment provides a specific implementation manner for determining the beamforming manner according to the position of the sound source.
  • the differential beamforming method is applied to a differential beamforming module, and the differential beamforming module includes at least a forward differential filter for receiving an input signal, a backward differential filter for receiving an input signal, and a backward differential filter connected to it.
  • the output differential beamforming signal is a figure-eight beam.
  • the cardioid beam used in the prior art for a microphone array of a larger specification, beam distortion is likely to occur, so that the amplitude of the target sound source direction is smaller than the amplitude of the non-target sound source direction.
  • This type of beam width is narrow, which can improve the problem that the amplitude of the target sound source direction in the differential beamforming signal is smaller than the amplitude of the non-target sound source direction.
  • the distance between the first microphone and the target sound source is smaller than the distance between the second microphone and the target sound source; the vertical line of the connection between the two microphones will be two
  • the microphones are divided into two different planes; the target sound source range is the plane where the first microphone is located, and the interference range is the plane where the second microphone is located. This embodiment provides a specific way to divide the target sound source range and the interference range.
  • the distance between two microphones is greater than or equal to 2.5 cm.
  • the differential beam forming method of the present application can still maintain the constant beam of the differential beam forming signal characteristic.
  • Fig. 1 is a specific flowchart of a differential beamforming method according to the first embodiment of the present application
  • FIG. 2 is a schematic diagram of a differential beamforming module applied according to the differential beamforming method in the first embodiment of the present application;
  • Fig. 3 is a beam diagram of a differential beamforming signal according to the first embodiment of the present application.
  • FIG. 4 is a specific flowchart of the differential beamforming method in the second embodiment of the present application.
  • Fig. 5 is a schematic plan view of two microphones and a target sound source formed according to a second embodiment of the present application
  • Fig. 6 is a schematic diagram of a figure-eight beam according to the second embodiment of the present application.
  • FIG. 7 is a specific flowchart of the signal processing method in the third embodiment of the present application.
  • Fig. 8 is a schematic diagram of a differential beamforming module in a fourth embodiment according to the present application.
  • Fig. 9 is a schematic diagram of a differential beamforming module in a fifth embodiment according to the present application.
  • Fig. 10 is a schematic diagram of a signal processing device in a sixth embodiment according to the present application.
  • the first embodiment of the present application relates to a differential beam forming method, which is applied to an electronic device including a microphone array.
  • the electronic device may be a head-mounted device, a headset, or a hearing aid.
  • the microphone array includes one or more groups of microphones, and each group of microphones It includes two microphones.
  • a microphone array including one set of microphones is used as an example for description.
  • one of the microphones can be opened according to the needs when in use.
  • the differential beam forming method of the present application are all microphone arrays suitable for differential noise suppression, that is, generally speaking, the distance between two microphones is less than or Equal to 6 cm.
  • the microphone array in the headset is in the normal use position, and the user’s mouth is the target sound source.
  • One of the two microphones is facing the user’s mouth for receiving the user
  • the other microphone is facing away from the user's mouth, and is mainly used to receive the signal of the user's mouth in the reverse direction.
  • Step 101 Obtain a differential beamforming signal according to the input signals obtained by the two microphones in the microphone array.
  • the first microphone and the second microphone respectively acquire the input signal of the target sound source, and respectively input the input signal into the differential beamforming module applied in the differential beamforming method of the present application, to obtain the differential beamforming signal.
  • the two microphones collect the input signals of the target sound source
  • Fourier transform is performed on the input signals respectively collected by the two microphones, and the input signals of each microphone are transformed by time domain signals. It is a frequency domain signal and is used as a signal input to the differential beamforming module.
  • Step 102 At least non-linearly adjust the amplitude of the differential beamforming signal based on the distance between the two microphones and the signal frequency of the input signal to obtain an adjusted differential beamforming signal.
  • the differential beamforming signal when adjusting the differential beamforming signal, it includes the adjustment of both amplitude and phase.
  • adjusting the amplitude of the differential beamforming signal it is based on the distance between the two microphones and the signal frequency of the input signal at least The amplitude of the differential beamforming signal is adjusted nonlinearly; when the phase of the differential beamforming signal is adjusted, the phase of the differential beamforming signal is adjusted based on the distance between the two microphones and the signal frequency of the input signal; In the example, the phase of the differential beamforming signal can be linearly adjusted based on the distance between the two microphones and the signal frequency of the input signal; the differential beamforming signal is adjusted in amplitude and phase to obtain the adjusted differential beam Form a signal.
  • the differential beamforming signal is adjusted based on the preset compensation filter to obtain the adjusted differential beamforming signal;
  • the distance between two microphones in the microphone array is greater than or equal to 2.5 cm.
  • the differential beam forming method of the present application can still maintain the differential beam forming signal. Constant beam characteristics.
  • the differential beamforming module can be a module of a chip in an electronic device. Please refer to Figure 2.
  • the differential beamforming module includes a delay module and The forward differential filter 1, composed of a delay module and an adder, a forward differential filter 2, an adaptive filter 3, an adder 4, and a compensation filter 5 composed of an adder.
  • the first microphone 10 and the second microphone 20 are two microphones in the microphone array of the electronic device, and the electronic device is in the normal use state, that is, when the microphone array is in the normal use position, the first microphone 10 and the target sound source The distance between the two microphones is smaller than the distance between the second microphone 20 and the target sound source as an example.
  • the amplitude expression of the target sound source is denoted as S( ⁇ ), and the direction vector of the target sound source is
  • ⁇ is the angle between the target sound source deviating from the direction facing the first microphone 10
  • ⁇ d/c
  • d is the distance between the two microphones
  • c the sound propagation speed in the air
  • ⁇ is the input signal Signal angular frequency.
  • step 101 after the first microphone 10 and the second microphone 20 obtain the input signal of the target sound source, they are respectively input to the differential beam forming module, and the signal obtained after passing through the forward differential filter 1 is the output of the forward differential filter 1.
  • the signal is
  • the signal obtained after the backward differential filter 2, that is, the signal output by the backward differential filter 2 is
  • the signal C B ( ⁇ , ⁇ ) output by the backward differential filter 2 is input to the adaptive filter 3, and ⁇ represents the coefficient of the adaptive filter 3, then the output signal of the adaptive filter 3 can be obtained as ⁇ C B ( ⁇ , ⁇ ).
  • the signal ⁇ C B ( ⁇ , ⁇ ) output by the adaptive filter 3 and the signal C F ( ⁇ , ⁇ ) output by the forward difference filter 4 are input to the adder 4, respectively, and the output of the forward difference filter 1
  • the signal C F ( ⁇ , ⁇ ) minus the signal ⁇ C B ( ⁇ , ⁇ ) output by the adaptive filter 3 is used as the output of the adder 4, which is the differential beamforming signal
  • step 102 the differential beamforming signal Y( ⁇ , ⁇ ) is input to the compensation filter 5 to obtain the adjusted differential beamforming signal Y'
  • this embodiment obtains input signals from two microphones of the microphone array, and then obtains a differential beamforming signal based on the input signals obtained by the two microphones, and then based on the distance between the two microphones and the input signal.
  • the signal frequency is at least nonlinearly adjusted to the amplitude of the differential beamforming signal to obtain the adjusted differential beamforming signal.
  • this embodiment provides an adjustment method. For microphone arrays of different specifications, based on two microphones The distance between the distance and the signal frequency of the input signal is at least nonlinearly adjusted to the amplitude of the differential beamforming signal, so as to ensure the constant beam characteristics of the differential beamforming signal.
  • the second embodiment of the present application relates to a differential beam forming method.
  • This embodiment is a refinement based on the first embodiment.
  • the main refinement lies in that it provides a method for obtaining input signals obtained from two microphones in a microphone array.
  • Step 201 includes the following sub-steps:
  • the sound source location is determined according to the input signal.
  • ⁇ ⁇ null varies, and therefore may be controlled by controlling the size of ⁇ ⁇ null, i.e., the beam is controlled by controlling the size of the differential ⁇ is formed null position signal to control the differential signal of FIG beamforming beam ;
  • the sub-beamforming signal Y( ⁇ , ⁇ ) needs to be minimized in the mean square sense, namely
  • Get Wiener Solution among them Represents the autocorrelation value of the signal C B ( ⁇ , ⁇ ) output by the backward differential filter 2, It represents the cross-correlation value between the signal C F ( ⁇ , ⁇ ) output by the forward differential filter 4 and the signal C B ( ⁇ , ⁇ ) output by the backward differential filter 2.
  • the value of ⁇ can be obtained from the signal C F ( ⁇ , ⁇ ) output by the forward differential filter 4 and the signal C B ( ⁇ , ⁇ ) output by the backward differential filter 2.
  • the input signal of the microphone calculates C F ( ⁇ , ⁇ ) and C B ( ⁇ , ⁇ ), and then the value of ⁇ can be calculated.
  • the sound source position can be determined according to the value of ⁇ .
  • the first microphone 10, the second microphone 20, and the target sound source 30 form a plane
  • the vertical line Y of the line connecting the first microphone 10 and the second microphone 20 divides the two microphones In the two different half planes of the plane, that is, the plane is divided into two half planes: 0 ⁇ 90, which is the front half plane, and 90 ⁇ 180, which is the back half plane.
  • the microphone is located in the rear half plane, and when 90 ⁇ 180, it is considered that the target sound source deviates from the first microphone 10 to a greater extent, so it is regarded as a non-sound source direction.
  • the first microphone 10 When the microphone array is in the normal use position, the first microphone 10 is closer to the target sound source 30 than the second microphone 20, and the target sound source range is the plane where the first microphone 10 is located, that is, the target sound source range is the front half plane, 0 ⁇ 90 ,
  • the interference range is the plane where the second microphone 20 is located, that is, the interference sound source range is the second half plane, 90 ⁇ 180.
  • the beam forming mode is determined according to the position of the sound source.
  • the sound source position belongs to the target sound source range, and the input signal comes from the front half plane.
  • the received signal contains the signal of the target sound source and cannot be nulled, so the fixed difference is adopted
  • the beamforming method is used as the beamforming method.
  • the absolute value of the coefficient ⁇ is 1, so that the formed differential beamforming signal is a figure-eight beam.
  • the cardioid beam used in the prior art beam distortion is likely to occur for a microphone array of larger specifications, which makes the target sound source direction
  • the amplitude of is smaller than the amplitude of the non-target sound source direction, and the figure-eight beam is used in this application.
  • This type of beam has a narrower beam width and can improve the amplitude of the target sound source direction in the differential beamforming signal. The problem.
  • the coefficient of the adaptive filter 5 is a fixed value; that is, the fixed differential beamforming method can be understood as the input signals of the two microphones respectively passing through the forward differential filter 1 and the backward differential filter.
  • 2 Perform the difference, the signal after the backward difference filter 2 is input to the adaptive filter 3 with a fixed coefficient, and the signal output from the adaptive filter 3 and the signal output from the forward difference filter 1 are input to the addition After the device 4, the adder 4 outputs a differential beamforming signal.
  • the sound source position belongs to the preset interference range, and the input signal is from the second half plane.
  • the received signal is considered to be an interference signal and needs to be nulled, and the beamforming method is determined to be adaptive differential beamforming In this way, the calculated value of ⁇ is used as the coefficient of the adaptive filter 5, so that the interference signal can be suppressed by adaptive nulling.
  • the coefficient of the adaptive filter 5 is adaptively changed; that is, the adaptive differential beamforming method can be understood as the input signals of the two microphones respectively passing through the forward differential filter 1 and the backward differential Filter 2 performs the differential, the signal after the differential filter 2 is input to the adaptive filter 3 with adaptive coefficient change, and the signal output from the adaptive filter 3 and the signal output from the forward differential filter 1 are input After the adder 4, the adder 4 outputs a differential beamforming signal.
  • the input signal is processed according to the determined beamforming mode, and a differential beamforming signal is output.
  • the input signals obtained by the first microphone 10 and the second microphone 20 are processed according to the beamforming manner determined in the sub-step 2012, and the corresponding differential beamforming signals are output.
  • step 202 at least the amplitude of the differential beamforming signal is nonlinearly adjusted based on the distance between the two microphones and the signal frequency of the input signal to obtain an adjusted differential beamforming signal.
  • step 102 in the first embodiment is substantially the same as step 102 in the first embodiment, and will not be repeated here.
  • this embodiment provides a specific implementation manner for obtaining a differential beamforming signal according to the input signals obtained by two microphones in the microphone array.
  • the third embodiment of the present application relates to a signal processing method, which is applied to an electronic device including a microphone array.
  • the electronic device may be a head-mounted device, a headset, or a hearing aid.
  • the microphone array includes one or more groups of microphones, each group of microphones including Two microphones.
  • a microphone array including a set of microphones is used as an example for description.
  • one of the microphones can be turned on according to the needs when in use, and the same applies In the signal processing method of this application.
  • Step 301 Correct the sound signals collected by the two microphones in the microphone array, and obtain input signals.
  • the amplitude and phase of the sound signals collected by the two microphones are corrected to obtain the input signal, so that the input signal meets the requirements of the differential beam forming method in the first embodiment or the second embodiment; for example, this
  • the amplitude and phase of one of the two sound signals collected by the two microphones may be corrected, so that the amplitude and phase of the sound signal after the amplitude and phase correction are consistent with the amplitude and phase of the other sound signal.
  • Step 302 Perform differential beamforming processing on the input signal based on the differential beamforming method in the first embodiment or the second embodiment, and obtain an adjusted differential beamforming signal.
  • differential beamforming method in the first embodiment or the second embodiment uses the differential beamforming method in the first embodiment or the second embodiment to perform differential beamforming processing on the input signal obtained in step 301, and obtain the adjusted differential beamforming signal.
  • specific processing process please refer to The first embodiment and the second embodiment will not be repeated here.
  • Step 303 Perform post filtering on the adjusted differential beamforming signal.
  • the post-filtering process is based on the difference between the desired signal and the interference signal in the time domain, so that the residual interference signal in the adjusted differential beamforming signal can be suppressed more effectively.
  • the post-filtering method can be dimensional
  • the post-filtering method can accurately estimate the spectral information of the desired signal or the interference signal, and then determine the filter coefficients of the Wiener post-filtering according to different optimization criteria, such as the minimum mean square error criterion, Then, the adjusted differential beamforming signal can be post-filtered to obtain the output signal.
  • this embodiment provides a signal processing method applying the differential beamforming method of the first embodiment or the second embodiment.
  • the two microphones of the microphone array obtain input signals, and then according to the two
  • the input signal obtained by the microphone obtains the differential beamforming signal, and then based on the distance between the two microphones and the signal frequency of the input signal, at least the amplitude of the differential beamforming signal is nonlinearly adjusted to obtain the adjusted differential beamforming signal.
  • this embodiment provides an adjustment method.
  • at least the amplitude of the differential beamforming signal can be adjusted nonlinearly based on the distance between the two microphones and the signal frequency of the input signal. Try to ensure the constant beam characteristics of the differential beamforming signal.
  • the fourth embodiment of the present application relates to a differential beamforming module, which is applied to an electronic device including a microphone array.
  • the electronic device may be a head-mounted device, a headset, or a hearing aid.
  • the microphone array includes at least one set of microphones, and each set of microphones includes two Two microphones of a group of microphones in the microphone array are used as an example for description in this embodiment and subsequent embodiments.
  • the differential beamforming module 100 includes:
  • the differential beamforming submodule 101 is configured to obtain a differential beamforming signal according to the input signals obtained by two microphones in the microphone array;
  • the adjustment sub-module 102 is configured to perform nonlinear adjustment at least on the amplitude of the differential beamforming signal based on the distance between the two microphones and the signal frequency of the input signal to obtain the adjusted differential beamforming signal. Specifically, when adjusting the differential beamforming signal, it includes adjustments in both amplitude and phase. When adjusting the amplitude of the differential beamforming signal, it is based on the distance between the two microphones and the signal frequency of the input signal.
  • the amplitude of the differential beamforming signal is adjusted nonlinearly; when the phase of the differential beamforming signal is adjusted, the phase of the differential beamforming signal is adjusted based on the distance between the two microphones and the signal frequency of the input signal; in an example , The phase of the differential beamforming signal can be linearly adjusted based on the distance between the two microphones and the signal frequency of the input signal. After the differential beamforming signal is adjusted in terms of amplitude and phase, an adjusted differential beamforming signal is obtained.
  • the adjustment sub-module 102 when adjusting the amplitude and phase of the differential beamforming signal, adjusts the differential beamforming signal based on a preset compensation filter to obtain the adjusted differential beamforming signal;
  • the differential beamforming module of the present application can still maintain the constant beam characteristics of the differential beamforming signal.
  • the first embodiment corresponds to this embodiment, this embodiment can be implemented in cooperation with the first embodiment.
  • the related technical details mentioned in the first embodiment are still valid in this embodiment, and the technical effects that can be achieved in the first embodiment can also be achieved in this embodiment. In order to reduce repetition, details are not repeated here. Correspondingly, the related technical details mentioned in this embodiment can also be applied to the first embodiment.
  • this embodiment obtains input signals from two microphones of the microphone array, and then obtains a differential beam forming signal based on the input signals obtained by the two microphones, and then at least a differential beam forming signal is obtained based on the distance between the two microphones.
  • the amplitude of the formed signal is adjusted nonlinearly to obtain the adjusted differential beamforming signal.
  • this embodiment provides an adjustment method.
  • the distance between the two microphones is at least After the amplitude of the differential beamforming signal is adjusted nonlinearly, the constant beam characteristics of the differential beamforming signal can be ensured as much as possible.
  • the fifth embodiment of the present application relates to a differential beamforming module.
  • This embodiment is a refinement on the basis of the fourth embodiment.
  • the main details are as follows: please refer to FIG. 9, the differential beamforming submodule 101 includes:
  • the distance between the first microphone 10 and the target sound source is smaller than the distance between the second microphone 20 and the target sound source; the vertical line connecting the two microphones will be The two microphones are divided into two different planes; the target sound source range is the plane where the first microphone is located, and the interference range is the plane where the second microphone is located.
  • the first determining unit 1011 is configured to determine the position of the sound source according to the input signal.
  • the second determining unit 1012 is configured to determine the beam forming mode according to the position of the sound source.
  • the beam forming unit 1013 is configured to process the input signal according to the determined beam forming mode and output a differential beam forming signal.
  • the second determining unit 1012 is specifically configured to determine that the beamforming mode is fixed differential beamforming when the sound source position belongs to the preset target sound source range, and when the sound source position belongs to the preset interference range, determine the beamforming
  • the method is adaptive differential beam forming.
  • the coefficient of the adaptive filter 5 is a fixed value; that is, the fixed differential beamforming mode can be understood as the input signals of the two microphones passing through the forward direction respectively.
  • the differential filter 1 and the backward differential filter 2 are differentiated, and the signal after the differential differential filter 2 is input to the adaptive filter 3 with a fixed coefficient.
  • the output signal of the adaptive filter 3 is the same as the forward After the signal output by the differential filter 1 is input to the adder 4, the adder 4 outputs a differential beamforming signal.
  • the coefficients of the adaptive filter 5 are adaptively changed; that is, the adaptive differential beamforming method can be understood as the input signals of the two microphones through the forward differential filter 1 and the backward differential filter.
  • 2 Perform the difference, the signal after the backward difference filter 2 is input to the adaptive filter 3 with adaptive coefficient change, the signal output from the adaptive filter 3 and the signal output from the forward difference filter 1 are input to the addition After the device 4, the adder 4 outputs a differential beamforming signal.
  • the output differential beamforming signal is a figure-eight beam.
  • This type of beam has a narrower beam width, which can improve the amplitude of the sound source in the differential beamforming signal. The problem of the amplitude of the sound source position.
  • the second embodiment corresponds to this embodiment, this embodiment can be implemented in cooperation with the second embodiment.
  • the related technical details mentioned in the second embodiment are still valid in this embodiment, and the technical effects that can be achieved in the second embodiment can also be achieved in this embodiment. In order to reduce repetition, details are not repeated here. Correspondingly, the related technical details mentioned in this embodiment can also be applied to the second embodiment.
  • this embodiment provides a specific implementation manner for obtaining a differential beamforming signal according to the input signals obtained by two microphones in the microphone array.
  • the sixth embodiment of the present application relates to a signal processing device, which is applied to an electronic device including a microphone array.
  • the electronic device may be a head-mounted device, a headset, or a hearing aid.
  • the microphone array includes at least one set of microphones, and each set of microphones includes two Microphones. In this embodiment and the following embodiments, two microphones of a group of microphones in the microphone array are used as an example for description.
  • the signal processing device includes:
  • the correction module 200 is used to correct the sound signals collected by the two microphones in the microphone array and obtain input signals;
  • the differential beamforming module 100 of the fourth embodiment or the fifth embodiment is configured to perform differential beamforming processing on an input signal and obtain an adjusted differential beamforming signal;
  • the post filtering module 300 is used to perform post filtering on the adjusted differential beamforming signal to obtain an output signal.
  • the third embodiment corresponds to this embodiment, this embodiment can be implemented in cooperation with the third embodiment.
  • the related technical details mentioned in the third embodiment are still valid in this embodiment, and the technical effects that can be achieved in the third embodiment can also be achieved in this embodiment. In order to reduce repetition, details are not repeated here. Correspondingly, the related technical details mentioned in this embodiment can also be applied in the third embodiment.
  • this embodiment provides a signal processing device including the differential beamforming module of the fourth embodiment or the fifth embodiment.
  • the two microphones of the microphone array obtain input signals, and then according to the two The input signal obtained by the microphone obtains the differential beamforming signal, and then at least the amplitude of the differential beamforming signal is nonlinearly adjusted based on the distance between the two microphones to obtain the adjusted differential beamforming signal.
  • this embodiment An adjustment method is provided. For microphone arrays of different specifications, after at least the amplitude of the differential beamforming signal is adjusted nonlinearly based on the distance between the two microphones, the constant beam characteristics of the differential beamforming signal can be ensured as much as possible.
  • the seventh embodiment of the present application relates to a chip, including the signal processing device of the sixth embodiment.
  • the eighth embodiment of the present application relates to an electronic device, which includes a microphone array and the chip of the seventh embodiment; the microphone array includes at least two microphones, and the chip is connected to each microphone.

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Abstract

本申请部分实施例提供了一种差分波束形成方法及模块、信号处理方法及装置、芯片。差分波束形成方法包括：根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号(101)；基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号(102)。采用上述方案，对于不同规格的麦克风阵列，都能尽量保证差分波束形成信号的恒定波束特性。

Description

差分波束形成方法及模块、信号处理方法及装置、芯片 技术领域
本申请涉及信号处理技术领域，特别涉及一种差分波束形成方法及模块、信号处理方法及装置、芯片。
背景技术
目前，在免提设备和头戴式设备中，为了更好的满足通话需求，一般设置麦克风阵列来对语音进行增强处理；麦克风阵列由一组麦克风按照一定方式布置在空间不同位置形成，能够接收空间信号，对空间分布的场信号进行采样，得到信号源的空间离散观测数据，利用该数据中的空间信息进行算法处理，增强所需要的语音，抑制无用的干扰和噪声。
对于全指向性双麦克风小型阵列，可以通过差分算法来对两个麦克风的信号进行处理，来实现语音信号的增强。
发明人发现现有技术至少存在以下问题：现有的差分算法仅适用于麦克风阵列中前后两个麦克风的距离小于2.5厘米的情况，当前后两个麦克风的距离稍大于2.5厘米时，无法保证恒定波束特性。
发明内容
本申请部分实施例的目的在于提供一种差分波束形成方法及模块、信号 处理方法及装置、芯片，对于不同规格的麦克风阵列，都能尽量保证差分波束形成信号的恒定波束特性。
本申请实施例提供了一种差分波束形成方法，包括：根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号；基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号。
本申请实施例还提供了一种信号处理方法，包括：对麦克风阵列中两个麦克风采集的声音信号进行校正，并得到输入信号；基于上述的差分波束形成方法，对输入信号进行差分波束形成处理，并得到调整后的差分波束形成信号；对调整后的差分波束形成信号进行后置滤波。
本申请实施例还提供了一种差分波束形成模块，包括：差分波束形成子模块，用于根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号；调整子模块，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号。
本申请实施例还提供了一种信号处理装置，包括：校正模块，用于对麦克风阵列中两个麦克风采集的声音信号进行校正，并得到输入信号；上述的差分波束形成模块，用于对输入信号进行差分波束形成处理，并得到调整后的差分波束形成信号；后置滤波模块，用于对调整后的差分波束形成信号进行后置滤波。
本申请实施例还提供了一种芯片，包括上述的信号处理装置。
本申请实施例还提供了一种电子设备，包括麦克风阵列以及上述的芯片；麦克风阵列包括至少两个麦克风，芯片连接于各麦克风。
本申请实施例现对于现有技术而言，由麦克风阵列的两个麦克风获取输入信号，再根据两个麦克风获取的输入信号得到差分波束形成信号，继而基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，换而言之，本实施例提供了一种调整方法，对于不同规格的麦克风阵列，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整后，都能尽量保证差分波束形成信号的恒定波束特性。
例如，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，包括：基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行调整，得到调整后的差分波束形成信号。本实施例提供了基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号的一种具体实现方式。
例如，基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行调整，得到调整后的差分波束形成信号，包括：基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行线性调整，得到调整后的差分波束形成信号。本实施例提供了基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行调整，得到调整后的差分波束形成信号的一种具体实现方式。
例如，基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行线性调整，得到调整后的差分波束形成信号，包括：基于预设的补偿滤波器对差分波束形成信号进行调整，得到调整后的差分波束形成信号；补偿滤波器的系统函数为
Figure PCTCN2019091307-appb-000001
其中，τ＝d/c，d为两个麦克风之间的距离，c为声音在空气中的传播速度，ω为输入信号的信号角频率。本实施例提供了基于两个麦克风之间的距离与输入信号的信号频率分别对差分波束形成信号的幅度进行非线性调整、对差分波束形成信号的相位进行线性调整，得到调整后的差分波束形成信号的一种具体方式。
例如，根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号，包括：根据输入信号确定声源位置；根据声源位置确定波束形成方式；根据确定的波束形成方式对输入信号进行处理，并输出差分波束形成信号。本实施例提供了根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号的一种具体实现方式。
例如，根据声源位置确定波束形成方式，包括：若声源位置属于预设的目标声源范围，确定波束形成方式为固定差分波束形成方式；若声源位置属于预设的干扰范围，确定波束形成方式为自适应差分波束形成方式。本实施例提供了根据声源位置确定波束形成方式的一种具体实现方式。
例如，差分波束形成方法应用于差分波束形成模块，差分波束形成模块至少包括用于接收输入信号的前向差分滤波器、用于接收输入信号的后向差分滤波器、连接于后向差分滤波器的自适应滤波器、分别连接于前向差分滤波器与自适应滤波器的加法器以及连接于加法器的补偿滤波器；固定差分波束形成 方式中，自适应滤波器的系数为固定值；自适应差分波束形成方式中，自适应滤波器的系数自适应变化。
例如，波束形成方式为固定差分波束形成方式时，输出的差分波束形成信号为8字形波束。在现有技术所采用的心型波束中，对于较大规格的麦克风阵列容易产生波束畸变，使得目标声源方向的幅度小于非目标声源方向的幅度，本实施例中采用的是8字形波束，该类型波束宽度较窄，能够改善差分波束形成信号中目标声源方向的幅度小于非目标声源方向的幅度的问题。
例如，两个麦克风分别第一麦克风与第二麦克风，第一麦克风与目标声源之间的距离小于第二麦克风与目标声源之间的距离；两个麦克风的连线的中垂线将两个麦克风划分在两个不同的平面内；目标声源范围为第一麦克风所在的平面，干扰范围为第二麦克风所在的平面。本实施例提供了划分目标声源范围和干扰范围的一种具体方式。
例如，两个麦克风之间的距离大于或等于2.5厘米。本实施例中，对于两个麦克风之间的距离大于或等于2.5里面的麦克风阵列，相较于现有的差分波束形成方法，本申请的差分波束形成方法仍能够保持差分波束形成信号的恒定波束特性。
附图说明
一个或多个实施例通过与之对应的附图中的图片进行示例性说明，这些示例性说明并不构成对实施例的限定，附图中具有相同参考数字标号的元件表示为类似的元件，除非有特别申明，附图中的图不构成比例限制。
图1是根据本申请第一实施例中的差分波束形成方法的具体流程图；
图2是根据本申请第一实施例中的差分波束形成方法应用的差分波束形成模块的示意图；
图3是根据本申请第一实施例中的差分波束形成信号的波束图；
图4是根据本申请第二实施例中的差分波束形成方法的具体流程图；
图5是根据本申请第二实施例中的两个麦克风与目标声源形成的平面示意图；
图6是根据本申请第二实施例中的8字形波束的示意图；
图7是根据本申请第三实施例中的信号处理方法的具体流程图；
图8是根据本申请第四实施例中的差分波束形成模块的示意图；
图9是根据本申请第五实施例中的差分波束形成模块的示意图；
图10是根据本申请第六实施例中的信号处理装置的示意图。
具体实施例
为了使本申请的目的、技术方案及优点更加清楚明白，以下结合附图及实施例，对本申请部分实施例进行进一步详细说明。应当理解，此处所描述的具体实施例仅仅用以解释本申请，并不用于限定本申请。
本申请第一实施例涉及一种差分波束形成方法，应用于包括麦克风阵列的电子设备，电子设备可以为头戴式设备、耳机或者助听器等，麦克风阵列包括一组或多组麦克风，每组麦克风包括两个麦克风，本实施例以及之后的实施例中均以包括一组麦克风的麦克风阵列为例进行说明，对于包括多组麦克风的麦克风阵列，则可以在使用时根据需求打开其中一组麦克风，同样适用于本申请的差分波束形成方法。另外需要说明的是，本申请各实施方式的差分波束形 成方法所应用的麦克风阵列，均是适用于差分方式进行噪声抑制的麦克风阵列，即，一般而言，两个麦克风之间的距离小于或等于6厘米。
以电子设备为耳机为例，当用户佩戴耳机后，耳机中的麦克风阵列处于正常使用位置，用户的嘴巴即为目标声源，两个麦克风中的一个麦克风正对用户的嘴巴，用于接收用户嘴巴方向的信号，另一个麦克风则是背对用户的嘴巴，主要用于接收用户嘴巴反向的信号。
本实施例中差分波束形成方法的具体流程如图1所示。
步骤101，根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号。
具体而言，第一麦克风与第二麦克风分别获取目标声源的输入信号，并分别输入本申请的差分波束形成方法所应用的差分波束形成模块中，可以得到差分波束形成信号。
需要说明的是，本实施例中，在两个麦克风采集目标声源的输入信号后，会对两个麦克风分别采集的输入信号进行傅里叶变换，将各麦克风的输入信号由时域信号变换为频域信号，作为输入到差分波束形成模块中的信号。
步骤102，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号。
具体而言，在对差分波束形成信号进行调整时，包括幅度和相位两方面的调整，在对差分波束形成信号的幅度进行调整时，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整；在对差分波束形成信号的相位进行调整时，基于两个麦克风之间的距离与输入信号的信号频率对差分波束形成信号的相位进行调整；在一个例子中，可以基于 两个麦克风之间的距离与输入信号的信号频率对差分波束形成信号的相位进行线性调整；差分波束形成信号在经过幅度和相位两方面的调整后，得到调整后的差分波束形成信号。
在一个例子中，在对差分波束形成信号的幅度和相位进行调整时，基于预设的补偿滤波器对差分波束形成信号进行调整，得到调整后的差分波束形成信号；补偿滤波器的系统函数为
Figure PCTCN2019091307-appb-000002
其中，τ＝d/c，d为两个麦克风之间的距离，c为声音在空气中的传播速度，ω为输入信号的信号角频率，该角频率与频率成正比，是频率的2π倍。
在一个例子中，麦克风阵列中的两个麦克风之间的距离大于或等于2.5厘米，相较于现有的差分波束形成方法而言，本申请的差分波束形成方法仍能够保持差分波束形成信号的恒定波束特性。
下面结合本实施例的差分波束形成方法所应用的差分波束形成模块为例进行说明，差分波束形成模块可以为电子设备中芯片的模块，请参考图2，差分波束形成模块包括由延时模块与加法器组成的前向差分滤波器1、由延时模块与加法器组成的后向差分滤波器2、自适应滤波器3、加法器4以及补偿滤波器5。其中，第一麦克风10与第二麦克风20是电子设备的麦克风阵列中的两个麦克风，且以电子设备处于正常使用状态，即麦克风阵列处于正常使用位置时，第一麦克风10与目标声源之间的距离小于第二麦克风20与目标声源之间的距离为例进行说明。
本实施例中目标声源的幅度表达式记作为S(ω)，目标声源的方向向量为
Figure PCTCN2019091307-appb-000003
前向差分滤波器1的系统函数为Hf(ω)＝[1,-e -jωτ] T，后向差分滤波器2的系统函数为Hb(ω)＝[-e -jωτ,1] T补偿滤波 器的系统函数为
Figure PCTCN2019091307-appb-000004
其中，θ为目标声源偏离正对第一麦克风10方向的夹角，τ＝d/c，d为两个麦克风之间的距离，c为声音在空气中的传播速度，ω为输入信号的信号角频率。
在步骤101中，第一麦克风10与第二麦克风20获取目标声源的输入信号后，分别输入差分波束形成模块，经过前向差分滤波器1后得到的信号，即前向差分滤波器1输出的信号为
Figure PCTCN2019091307-appb-000005
Figure PCTCN2019091307-appb-000006
经过后向差分滤波器2后得到的信号，即后向差分滤波器2输出的信号为
Figure PCTCN2019091307-appb-000007
Figure PCTCN2019091307-appb-000008
后向差分滤波器2输出的信号C B(ω,θ)输入到自适应滤波器3，以β表示自适应滤波器3的系数，则可以得到自适应滤波器3输出的信号为βC B(ω,θ)。
然后，自适应滤波器3输出的信号βC B(ω,θ)与前向差分滤波器4输出的信号C F(ω,θ)分别输入到加法器4，将前向差分滤波器1输出的信号C F(ω,θ)减去自适应滤波器3输出的信号βC B(ω,θ)作为加法器4的输出，即为差分波束形成信号
Figure PCTCN2019091307-appb-000009
Figure PCTCN2019091307-appb-000010
在步骤102中，将差分波束形成信号Y(ω,θ)输入到补偿滤波器5，得到调整后的差分波束形成信号Y’
Figure PCTCN2019091307-appb-000011
Figure PCTCN2019091307-appb-000012
将差分波束形成信号Y(ω,θ)输入到补偿滤波器5后，需要使得调整后的差分波束形成信号Y’(ω,θ)较好的还原目标声源方向的信号；本实施方式中，用户的嘴巴为目标声源，第一麦克风10正对用户的嘴巴以接收用户嘴巴方向的信号，可以认为正对用户的嘴巴的方向，即θ＝0为目标声源方向。因此，为了较好地还原目标声源方向的信号，当θ＝0时，需要满足Y’(ω,θ)＝S(ω)；从而能够推导出补偿滤波器5的系统函数为
Figure PCTCN2019091307-appb-000013
如图3所示，为调整后的差分波束形成信号的波束图，可以看到不同频率的波束的幅度差距较小，具有恒定波束特性。
本实施例相对于现有技术而言，由麦克风阵列的两个麦克风获取输入信号，再根据两个麦克风获取的输入信号得到差分波束形成信号，继而基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，换而言之，本实施例提供了一种调整方法，对于不同规格的麦克风阵列，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整后，都能尽量保证差分波束形成信号的恒定波束特性。
本申请第二实施例涉及一种差分波束形成方法，本实施例是在第一实施例基础上的细化，主要细化之处在于：提供了根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号的一种具体实现方式。
本实施例的差分波束形成方法的具体流程如图4所示。
步骤201，包括以下子步骤：
子步骤2011，根据输入信号确定声源位置。
具体而言，根据第一实施例中计算出的差分波束形成信号
Figure PCTCN2019091307-appb-000014
Figure PCTCN2019091307-appb-000015
在差分波束形成信号的零陷位置，差分波束形成信号为0，以θ null表示在零陷位置时偏离正对第一麦克11的方向的角度，即θ＝θ null时，Y(ω,θ null)＝0，可以得出：
Figure PCTCN2019091307-appb-000016
求解得到
Figure PCTCN2019091307-appb-000017
可以看出β随着θ null而变化，因此也可以通过控制β的大小来控制θ null，即通过控制β的大小来控制差分波束形成信号的零陷位置，以控制差分波束形成信号的波束图；在求解β时，需要使分波束形成信号Y(ω,θ)在均方意义下最小，即
Figure PCTCN2019091307-appb-000018
Figure PCTCN2019091307-appb-000019
求得维纳解
Figure PCTCN2019091307-appb-000020
其中
Figure PCTCN2019091307-appb-000021
表示后向差分滤波器2输出的信号C B(ω,θ)自相关值，
Figure PCTCN2019091307-appb-000022
表示前向差分滤波器4输出的信号C F(ω,θ)与后向差分滤波器2输出的信号C B(ω,θ)的互相关值。
由上可知，β的值可以根据前向差分滤波器4输出的信号C F(ω,θ)与后向差分滤波器2输出的信号C B(ω,θ)求得，因此可以根据两个麦克风的输入信号计算出C F(ω,θ)与C B(ω,θ)，继而可以求出β的值。
继而可以根据β的值来确定声源位置。
在一个例子中，请参考图5，第一麦克风10、第二麦克风20以及目标声 源30形成一个平面，第一麦克风10与第二麦克风20的连线的中垂线Y将两个麦克风划分在该平面的两个不同的半平面内，即，将平面分为两个半平面：0≤θ＜90，为前半平面，90≤θ≤180，为后半平面。第一麦克风位于前半平面，θ＝0为目标声源方向，而0＜θ＜90时认为目标声源偏离正对第一麦克风10的程度较小，仍可以认为是目标声源方向；第二麦克风位于后半平面，90≤θ≤180时认为目标声源偏离正对第一麦克风10的程度较大，故认为是非声源方向。麦克风阵列处于正常使用位置时，第一麦克风10较第二麦克风20靠近目标声源30，目标声源范围为第一麦克风10所在的平面，即目标声源范围为前半平面，0≤θ＜90，干扰范围为第二麦克风20所在的平面，即干扰声源范围为后半平面，90≤θ≤180。
当|β|>1时，确定声源位置属于预设的目标声源范围；当|β|＜1，确定声源位置属于预设的干扰范围。
子步骤2012，根据声源位置确定波束形成方式。
具体而言，当|β|>1时，声源位置属于目标声源范围，输入信号来自前半平面，此时认为接收到的信号包含目标声源的信号，不能进行零陷，故采用固定差分波束形成方式作为波束形成方式，此时输出的差分波束形成信号为8字形波束，如图6所示，为8字形波束图，可知8字形波束的零陷位置为90°，根据公式
Figure PCTCN2019091307-appb-000023
可以求得8字形波束的β＝1，因此本实施例中，在β＞1时，设置β＝1，在β＜-1时，设置β＝-1，即设定自适应滤波器5的系数β的绝对值为1，从而使得形成的差分波束形成信号为8字形波束，在现有技术所采用的心型波束中，对于较大规格的麦克风阵列容易产生波束畸变，使得目标声源方向的幅度小于非目标声源方向的幅度，而本申请中采用的是8 字形波束，该类型波束宽度较窄，能够改善差分波束形成信号中目标声源方向的幅度小于非目标声源方向的幅度的问题。其中，在固定差分波束形成方式中，自适应滤波器5的系数为固定值；即固定差分波束形成方式可以理解为两个麦克风的输入信号分别经过前向差分滤波器1与后向差分滤波器2进行差分，经过后向差分滤波器2进行差分之后的信号输入到系数为固定值的自适应滤波器3，自适应滤波器3输出的信号与前向差分滤波器1输出的信号输入到加法器4之后，加法器4输出差分波束形成信号。
当|β|＜1，声源位置属于预设的干扰范围，输入信号来自后半平面，此时认为接收到的信号为干扰信号，需要进行零陷，确定波束形成方式为自适应差分波束形成方式，以计算出来的β的值作为自适应滤波器5的系数，从而能够通过自适应零陷来抑制干扰信号。其中，在自适应差分波束形成方式中，自适应滤波器5的系数自适应变化；即自适应差分波束形成方式可以理解为两个麦克风的输入信号分别经过前向差分滤波器1与后向差分滤波器2进行差分，经过后向差分滤波器2进行差分之后的信号输入到系数自适应变化的自适应滤波器3，自适应滤波器3输出的信号与前向差分滤波器1输出的信号输入到加法器4之后，加法器4输出差分波束形成信号。
子步骤2013，根据确定的波束形成方式对输入信号进行处理，并输出差分波束形成信号。
具体而言，按照子步骤2012中确定的波束形成方式对第一麦克风10与第二麦克风20获取的输入信号进行处理，输出对应的差分波束形成信号。
步骤202，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号。
具体而言，与第一实施例中的步骤102大致相同，在此不再赘述。
本实施例相对于第一实施例而言，提供了根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号的一种具体实现方式。
本申请第三实施例涉及一种信号处理方法，应用于包括麦克风阵列的电子设备，电子设备可以为头戴式设备、耳机或者助听器等，麦克风阵列包括一组或多组麦克风，每组麦克风包括两个麦克风，本实施例以及之后的实施例中均以包括一组麦克风的麦克风阵列为例进行说明，对于包括多组麦克风的麦克风，则可以在使用时根据需求打开其中一组麦克风，同样适用于本申请的信号处理方法。
本实施例的信号处理方法的具体流程如图7所示。
步骤301，对麦克风阵列中两个麦克风采集的声音信号进行校正，并得到输入信号。
具体而言，对两个麦克风采集的声音信号进行幅度和相位校正，得到输入信号，以使该输入信号满足第一实施例或第二实施例中的差分波束形成方法的使用需求；例如，本实施例中可以对两个麦克风采集的两个声音信号中的一个声音信号进行幅度和相位校正，使得经过幅度和相位校正后的声音信号与另一个声音信号的幅度和相位相一致。
步骤302，基于第一实施例或第二实施例中的差分波束形成方法，对输入信号进行差分波束形成处理，并得到调整后的差分波束形成信号。
具体而言，利用第一实施例或第二实施例中的差分波束形成方法对步骤301中得到的输入信号进行差分波束形成处理，并得到调整后的差分波束形成信号，具体的处理过程请参考第一实施例与第二实施例，在此不再赘述。
步骤303，对调整后的差分波束形成信号进行后置滤波。
具体而言，后置滤波基于期望信号与干扰信号所在时域的不同，进行滤波处理，从而能够更有效的抑制调整后的差分波束形成信号中残留的干扰信号，后置滤波的方式可以为维纳后置滤波方法，该方法能够准确的估计期望信号的谱信息，或干扰信号的谱信息，然后再根据不同的优化准则确定维纳后置滤波的滤波系数，例如，最小均方误差准则，继而就可以对调整后的差分波束形成信号进行后置滤波，以得到输出信号。
本实施例相对于现有技术而言，提供了一种应用第一实施例或第二实施例的差分波束形成方法的信号处理方法，由麦克风阵列的两个麦克风获取输入信号，再根据两个麦克风获取的输入信号得到差分波束形成信号，继而基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，换而言之，本实施例提供了一种调整方法，对于不同规格的麦克风阵列，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整后，都能尽量保证差分波束形成信号的恒定波束特性。
本申请第四实施例涉及一种差分波束形成模块，应用于包括麦克风阵列的电子设备，电子设备可以为头戴式设备、耳机或者助听器等，麦克风阵列包括至少一组麦克风，每组麦克风包括两个麦克风，本实施例以及之后的实施例中均以麦克风阵列中一组麦克风的两个麦克风为例进行说明。
如图8所示，差分波束形成模块100包括：
差分波束形成子模块101，用于根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号；
调整子模块102，用于基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号。具体的，在对差分波束形成信号进行调整时，包括幅度和相位两方面的调整，在对差分波束形成信号的幅度进行调整时，基于两个麦克风之间的距离与输入信号的信号频率至少对差分波束形成信号的幅度进行非线性调整；在对差分波束形成信号的相位进行调整时，基于两个麦克风之间的距离与输入信号的信号频率对差分波束形成信号的相位进行调整；在一个例子中，可以基于两个麦克风之间的距离与输入信号的信号频率对差分波束形成信号的相位进行线性调整。差分波束形成信号在经过幅度和相位两方面的调整后，得到调整后的差分波束形成信号。
在一个例子中，在对差分波束形成信号的幅度和相位进行调整时，调整子模块102基于预设的补偿滤波器对差分波束形成信号进行调整，得到调整后的差分波束形成信号；；补偿滤波器的系统函数为
Figure PCTCN2019091307-appb-000024
其中，τ＝d/c，d为两个麦克风之间的距离，c为声音在空气中的传播速度，ω为输入信号的信号角频率。
在一个例子中，麦克风阵列中的两个麦克风之间的距离大于或等于2.5厘米，本申请的差分波束形成模块仍能够保持差分波束形成信号的恒定波束特性。
由于第一实施例与本实施例相互对应，因此本实施例可与第一实施例互相配合实施。第一实施例中提到的相关技术细节在本实施例中依然有效，在第一实施例中所能达到的技术效果在本实施例中也同样可以实现，为了减少重复，这里不再赘述。相应地，本实施例中提到的相关技术细节也可应用在第一实施 例中。
本实施例相对于现有技术而言，由麦克风阵列的两个麦克风获取输入信号，再根据两个麦克风获取的输入信号得到差分波束形成信号，继而基于两个麦克风之间的距离至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，换而言之，本实施例提供了一种调整方法，对于不同规格的麦克风阵列，基于两个麦克风之间的距离至少对差分波束形成信号的幅度进行非线性调整后，都能尽量保证差分波束形成信号的恒定波束特性。
本申请第五实施例涉及一种差分波束形成模块，本实施例是在第四实施例基础上的细化，主要细化之处在于：请参考图9，差分波束形成子模块101包括：
本实施例中，麦克风阵列处于正常使用位置时，第一麦克风10与目标声源之间的距离小于第二麦克风20与目标声源之间的距离；两个麦克风的连线的中垂线将两个麦克风划分在两个不同的平面内；目标声源范围为第一麦克风所在的平面，干扰范围为第二麦克风所在的平面。
第一确定单元1011，用于根据输入信号确定声源位置。
第二确定单元1012，用于根据声源位置确定波束形成方式。
波束形成单元1013，用于根据确定的波束形成方式对输入信号进行处理，并输出差分波束形成信号。
其中，第二确定单元1012具体用于当声源位置属于预设的目标声源范围时，确定波束形成方式为固定差分波束形成方式，当声源位置属于预设的干扰范围时，确定波束形成方式为自适应差分波束形成方式。
请参考图2的差分波束形成模块的结构，在固定差分波束形成方式中， 自适应滤波器5的系数为固定值；即固定差分波束形成方式可以理解为两个麦克风的输入信号分别经过前向差分滤波器1与后向差分滤波器2进行差分，经过后向差分滤波器2进行差分之后的信号输入到系数为固定值的自适应滤波器3，自适应滤波器3输出的信号与前向差分滤波器1输出的信号输入到加法器4之后，加法器4输出差分波束形成信号。
在自适应差分波束形成方式中，自适应滤波器5的系数自适应变化；即自适应差分波束形成方式可以理解为两个麦克风的输入信号分别经过前向差分滤波器1与后向差分滤波器2进行差分，经过后向差分滤波器2进行差分之后的信号输入到系数自适应变化的自适应滤波器3，自适应滤波器3输出的信号与前向差分滤波器1输出的信号输入到加法器4之后，加法器4输出差分波束形成信号。
在一个例子中，波束形成方式为固定差分波束形成方式时，输出的差分波束形成信号为8字形波束，该类型波束宽度较窄，能够改善差分波束形成信号中正对声源位置的幅度小于斜对声源位置的幅度的问题。
由于第二实施例与本实施例相互对应，因此本实施例可与第二实施例互相配合实施。第二实施例中提到的相关技术细节在本实施例中依然有效，在第二实施例中所能达到的技术效果在本实施例中也同样可以实现，为了减少重复，这里不再赘述。相应地，本实施例中提到的相关技术细节也可应用在第二实施例中。
本实施例相对于第四实施例而言，提供了根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号的一种具体实现方式。
本申请第六实施例涉及一种信号处理装置，应用于包括麦克风阵列的 电子设备，电子设备可以为头戴式设备、耳机或者助听器等，麦克风阵列包括至少一组麦克风，每组麦克风包括两个麦克风，本实施例以及之后的实施例中均以麦克风阵列中一组麦克风的两个麦克风为例进行说明。
如图10所示，信号处理装置包括：
校正模块200，用于对麦克风阵列中两个麦克风采集的声音信号进行校正，并得到输入信号；
第四实施例或第五实施例的差分波束形成模块100，用于对输入信号进行差分波束形成处理，并得到调整后的差分波束形成信号；
后置滤波模块300，用于对调整后的差分波束形成信号进行后置滤波，以得到输出信号。
由于第三实施例与本实施例相互对应，因此本实施例可与第三实施例互相配合实施。第三实施例中提到的相关技术细节在本实施例中依然有效，在第三实施例中所能达到的技术效果在本实施例中也同样可以实现，为了减少重复，这里不再赘述。相应地，本实施例中提到的相关技术细节也可应用在第三实施例中。
本实施例相对于现有技术而言，提供了一种包括第四实施例或第五实施例的差分波束形成模块的信号处理装置，由麦克风阵列的两个麦克风获取输入信号，再根据两个麦克风获取的输入信号得到差分波束形成信号，继而基于两个麦克风之间的距离至少对差分波束形成信号的幅度进行非线性调整，得到调整后的差分波束形成信号，换而言之，本实施例提供了一种调整方法，对于不同规格的麦克风阵列，基于两个麦克风之间的距离至少对差分波束形成信号的幅度进行非线性调整后，都能尽量保证差分波束形成信号的恒定波束特性。
本申请第七实施例涉及一种芯片，包括第六实施例的信号处理装置。
本申请第八实施例涉及一种电子设备，包括麦克风阵列以及第七实施例的芯片；麦克风阵列包括至少两个麦克风，芯片连接于各麦克风。
本领域的普通技术人员可以理解，上述各实施例是实现本申请的具体实施例，而在实际应用中，可以在形式上和细节上对其作各种改变，而不偏离本申请的精神和范围。

Claims (24)

  1. 一种差分波束形成方法，其特征在于，包括：
    根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号；
    基于两个所述麦克风之间的距离与所述输入信号的信号频率至少对所述差分波束形成信号的幅度进行非线性调整，得到调整后的所述差分波束形成信号。
  2. 如权利要求1所述的方法，其特征在于，所述基于两个所述麦克风之间的距离与所述输入信号的信号频率至少对所述差分波束形成信号的幅度进行非线性调整，得到调整后的所述差分波束形成信号，包括：
    基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行调整，得到调整后的所述差分波束形成信号。
  3. 如权利要求2所述的方法，其特征在于，所述基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行调整，得到调整后的所述差分波束形成信号，包括：
    基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行线性调整，得到调整后的所述差分波束形成信号。
  4. 如权利要求3所述的方法，其特征在于，所述基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行线性调整，得到调整后的所述差分波束形成信号，包括：
    基于预设的补偿滤波器对所述差分波束形成信号进行调整，得到调整后的所述差分波束形成信号；所述补偿滤波器的系统函数为
    Figure PCTCN2019091307-appb-100001
    Figure PCTCN2019091307-appb-100002
    其中，τ＝d/c，d为两个所述麦克风之间的距离，c为声音在空气中的传播速度，ω为所述输入信号的信号角频率。
  5. 如权利要求1所述的方法，其特征在于，所述根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号，包括：
    根据所述输入信号确定声源位置；
    根据所述声源位置确定波束形成方式；
    根据确定的所述波束形成方式对所述输入信号进行处理，并输出所述差分波束形成信号。
  6. 如权利要求5所述的方法，其特征在于，所述根据所述声源位置确定波束形成方式，包括：
    若所述声源位置属于预设的目标声源范围，确定所述波束形成方式为固定差分波束形成方式；
    若所述声源位置属于预设的干扰范围，确定所述波束形成方式为自适应差分波束形成方式。
  7. 如权利要求6所述的方法，其特征在于，应用于差分波束形成模块，所述差分波束形成模块至少包括用于接收所述输入信号的前向差分滤波器、用于接收所述输入信号的后向差分滤波器、连接于所述后向差分滤波器的自适应滤波器、分别连接于所述前向差分滤波器与所述自适应滤波器的加法器以及连接于所述加法器的补偿滤波器；
    所述固定差分波束形成方式中，所述自适应滤波器的系数为固定值；
    所述自适应差分波束形成方式中，所述自适应滤波器的系数自适应变化。
  8. 如权利要求6所述的方法，其特征在于，所述波束形成方式为所述固定差分波束形成方式时，输出的所述差分波束形成信号为8字形波束。
  9. 如权利要求6所述的方法，其特征在于，两个所述麦克风分别第一麦克风与第二麦克风，所述第一麦克风与目标声源之间的距离小于所述第二麦克风与所述目标声源之间的距离；两个所述麦克风的连线的中垂线将两个所述麦克风划分在两个不同的平面内；
    所述目标声源范围为所述第一麦克风所在的平面，所述干扰范围为所述第二麦克风所在的平面。
  10. 如权利要求1至9中任一项所述的方法，其特征在于，两个所述麦克风之间的距离大于或等于2.5厘米。
  11. 一种信号处理方法，其特征在于，包括：
    对所述麦克风阵列中两个所述麦克风采集的声音信号进行校正，并得到所述输入信号；
    基于权利要求1至10中任一项所述的差分波束形成方法，对所述输入信号进行差分波束形成处理，并得到所述调整后的所述差分波束形成信号；
    对所述调整后的所述差分波束形成信号进行后置滤波。
  12. 一种差分波束形成模块，其特征在于，包括：
    差分波束形成子模块，用于根据麦克风阵列中两个麦克风获取的输入信号得到差分波束形成信号；
    调整子模块，用于基于两个所述麦克风之间的距离与所述输入信号的信号频率至少对所述差分波束形成信号的幅度进行非线性调整，得到调整后的所述差分波束形成信号。
  13. 如权利要求12所述的模块，其特征在于，所述调整子模块具体用于基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行调整，得到调整后的所述差分波束形成信号。
  14. 如权利要求13所述的模块，其特征在于，所述调整子模块具体用于基于两个所述麦克风之间的距离与所述输入信号的信号频率分别对所述差分波束形成信号的幅度进行非线性调整、对所述差分波束形成信号的相位进行线性调整，得到调整后的所述差分波束形成信号。
  15. 如权利要求14所述的模块，其特征在于，所述调整子模块具体用于基于预设的补偿滤波器对所述差分波束形成信号进行调整，得到调整后的所述差分波束形成信号；所述补偿滤波器的系统函数为
    Figure PCTCN2019091307-appb-100003
    其中，τ＝d/c，d为两个所述麦克风之间的距离，c为声音在空气中的传播速度，ω为所述输入信号的信号角频率。
  16. 如权利要求12所述的模块，其特征在于，所述差分波束形成子模块包括：
    第一确定单元，用于根据所述输入信号确定声源位置；
    第二确定单元，用于根据所述声源位置确定波束形成方式；
    波束形成单元，用于根据确定的所述波束形成方式对所述输入信号进行处理，并输出所述差分波束形成信号。
  17. 如权利要求16所述的模块，其特征在于，所述第二确定单元具体用于当所述声源位置属于预设的目标声源范围时，确定所述波束形成方式为固定差分波束形成方式，当所述声源位置属于预设的干扰范围时，确定所述波束形成方式为自适应差分波束形成方式。
  18. 如权利要求17所述的模块，其特征在于，应用于差分波束形成模块，所述差分波束形成模块至少包括用于接收所述输入信号的前向差分滤波器、用于接收所述输入信号的后向差分滤波器、连接于所述后向差分滤波器的自适应滤波器、分别连接于所述前向差分滤波器与所述自适应滤波器的加法器以及连接于所述加法器的补偿滤波器；
    所述固定差分波束形成方式中，所述自适应滤波器的系数为固定值；
    所述自适应差分波束形成方式中，所述自适应滤波器的系数自适应变化。
  19. 如权利要求17所述的模块，其特征在于，所述波束形成方式为所述固定差分波束形成方式时，输出的所述差分波束形成信号为8字形波束。
  20. 如权利要求17所述的模块，其特征在于，两个所述麦克风分别第一麦克风与第二麦克风，所述第一麦克风与目标声源之间的距离小于所述第二麦克风与所述目标声源之间的距离；两个所述麦克风的连线的中垂线将两个所述麦克风划分在两个不同的平面内；
    所述目标声源范围为所述第一麦克风所在的平面，所述干扰范围为所述第二麦克风所在的平面。
  21. 如权利要求12至20中任一项所述的模块，其特征在于，两个所述麦克风之间的距离大于或等于2.5厘米。
  22. 一种信号处理装置，其特征在于，包括：
    校正模块，用于对所述麦克风阵列中两个所述麦克风采集的声音信号进行校正，并得到所述输入信号；
    权利要求12至21中任一项所述的差分波束形成模块，用于对所述输入信号进行差分波束形成处理，并得到所述调整后的所述差分波束形成信号；
    后置滤波模块，用于对所述调整后的所述差分波束形成信号进行后置滤波。
  23. 一种芯片，其特征在于，包括权利要求22所述的信号处理装置。
  24. 一种电子设备，其特征在于，包括麦克风阵列以及权利要求23所述的芯片；所述麦克风阵列包括至少两个麦克风，所述芯片连接于各所述麦克风。
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