WO2018209893A1 - 麦克风阵列指向的调整方法和装置 - Google Patents

麦克风阵列指向的调整方法和装置 Download PDF

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Publication number
WO2018209893A1
WO2018209893A1 PCT/CN2017/108027 CN2017108027W WO2018209893A1 WO 2018209893 A1 WO2018209893 A1 WO 2018209893A1 CN 2017108027 W CN2017108027 W CN 2017108027W WO 2018209893 A1 WO2018209893 A1 WO 2018209893A1
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Prior art keywords
sound source
voice signals
microphone array
incident angle
speech signal
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PCT/CN2017/108027
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English (en)
French (fr)
Inventor
张晓红
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歌尔科技有限公司
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Publication of WO2018209893A1 publication Critical patent/WO2018209893A1/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
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/08—Mouthpieces; Microphones; Attachments therefor
    • 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
    • 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

Definitions

  • the present invention relates to the field of microphone technologies, and in particular, to a method and apparatus for adjusting a microphone array pointing.
  • Wire-controlled headphones have been widely used in people's daily lives. For example, people can control the playing of music, the adjustment of the volume in the mobile terminal, and the voice call with friends through the wired headset.
  • a microphone array is generally provided in the online control earphone.
  • the commonly used microphone array includes a dual microphone array, and of course, a microphone array including more microphones.
  • the wire-controlled headset In the process of using the wire-controlled headset to make a call, because the user's usage habits are different, some prefer the wire control to sag naturally, and some prefer to hold the wire control by hand, so it is easy to cause the orientation of the microphone array to change, so that during the call process
  • the middle microphone array does not keep pointing in the direction of the user's mouth, that is, the direction of the sound source, resulting in poor voice quality of the call.
  • aspects of the present invention provide a method and apparatus for adjusting a microphone array pointing by adjusting a microphone array to always point to a sound source direction to ensure call quality.
  • an embodiment of the present invention provides a method for adjusting a microphone array pointing, including:
  • the main lobe direction of the microphone array is adjusted according to the sound source incident angle.
  • the at least two voice signals comprise a first voice signal s1(t) and a second voice signal s2(t), and wherein the at least two voice signals of the multiple voice signals are correlated Sex analysis, including:
  • T 12 is a range of propagation time difference between the first speech signal s1(t) and the second speech signal s2(t)
  • T 12 [-L 12 /c, L 12 /c]
  • L 12 is The distance between the two microphones corresponding to the first speech signal s1(t) and the second speech signal s2(t)
  • c is the speed of sound.
  • the determining, according to the correlation analysis result and the propagation distance difference of the at least two voice signals, determining a sound source incident angle of the sound source to the microphone array including:
  • a 12 is a difference in propagation distance between the first speech signal s1(t) and the second speech signal s2(t);
  • Adjusting the direction of the main lobe of the microphone array according to the incident angle of the sound source includes:
  • the main lobe direction of the microphone array is adjusted according to the first sound source incident angle ⁇ 12 .
  • the at least two voice signals further include at least a third voice signal s3(t), the method Also includes:
  • the angular deviation of the main lobe direction adjusted according to the first sound source incident angle ⁇ 12 and the second sound source incident angle ⁇ 34 is greater than a preset threshold, according to the second sound source incident angle ⁇ 34 Adjusting the main lobe direction of the microphone array.
  • performing correlation analysis on at least two voice signals in the multiple voice signals including:
  • Determining, according to the correlation analysis result and the propagation distance difference of the at least two voice signals, determining a sound source incident angle of the sound source to the microphone array including:
  • Determining an incident angle of the sound source to the second sound source of the microphone array is: an average of sound source incident angles corresponding to all two voice signals.
  • an embodiment of the present invention provides an apparatus for adjusting a microphone array pointing, including:
  • a receiving module configured to receive, by using a microphone array, a plurality of voice signals, where the multiple voice signals are in one-to-one correspondence with a plurality of microphones in the microphone array;
  • a determining module configured to perform correlation analysis on at least two voice signals in the multiple voice signals; determining, according to the correlation analysis result and a propagation distance difference of the at least two voice signals, determining a sound source to the Sound source incident angle of the microphone array;
  • an adjustment module configured to adjust a main lobe direction of the microphone array according to the sound source incident angle.
  • the structure of the adjustment device pointed by the microphone array includes a processor and a memory for storing an adjustment device that supports pointing of the microphone array.
  • the adjustment means to which the microphone array is directed may also include a communication interface for the adjustment means pointed to by the microphone array to communicate with other devices or communication networks.
  • Embodiments of the present invention provide a computer storage medium for storing computer software instructions for an adjustment device pointed by a microphone array, comprising a program involved in performing the adjustment method of the microphone array pointing in the first aspect.
  • the method and device for adjusting the orientation of a microphone array provided by the present invention, when a user triggers a call through a wire-controlled headset, receives a multi-channel speech signal through a microphone array including a plurality of microphones, and then passes at least two voices in the multi-channel speech signal.
  • Correlation analysis of the signal based on the correlation analysis result and the relationship between the propagation distance difference of at least two voice signals and the analysis result, it can be determined that the sound source is incident on the sound source of the microphone array when the online control earphone is in the current state
  • the angle is such that the main lobe direction of the microphone array is adjusted based on the incident angle of the sound source such that the main lobe direction is directed to the incident angle of the sound to ensure the signal quality of the outputted call speech.
  • Embodiment 1 is a flowchart of Embodiment 1 of a method for adjusting a microphone array pointing according to an embodiment of the present invention
  • FIG. 2 is a schematic view showing a positional pattern of a microphone array in actual use
  • FIG. 3 is a schematic structural view of a dual microphone array
  • Embodiment 4 is a flowchart of Embodiment 2 of a method for adjusting a microphone array pointing according to an embodiment of the present invention
  • Figure 5 is a schematic diagram showing the principle of determining the incident angle of the sound source
  • FIG. 6 is a flowchart of Embodiment 3 of a method for adjusting a microphone array pointing according to an embodiment of the present disclosure
  • FIG. 7 is a schematic structural diagram of an apparatus for adjusting a microphone array pointing according to an embodiment of the present invention.
  • FIG. 8 is a schematic structural diagram of an electronic device corresponding to the adjustment device pointed by the microphone array provided in the embodiment shown in FIG. 7.
  • first, second, third, etc. may be used to describe XXX in embodiments of the invention, these XXX should not be limited to these terms. These terms are only used to distinguish XXX.
  • the first XXX may also be referred to as a second XXX without departing from the scope of the embodiments of the present invention.
  • the second XXX may also be referred to as a first XXX.
  • the words “if” and “if” as used herein may be interpreted as “in... “or” or “in response to the determination” or “in response to the detection.”
  • the phrase “if determined” or “if detected (conditions or events stated)” may be interpreted Becoming “when determined” or “in response to determination” or “when detecting (stated condition or event)” or “in response to detection (condition or event stated)”.
  • FIG. 1 is a flowchart of Embodiment 1 of a method for adjusting a microphone array pointing according to an embodiment of the present invention.
  • the method for adjusting the pointing of the microphone array provided by the embodiment may be performed by an adjustment device pointed by a microphone array, and the microphone array is pointed
  • the adjustment device can be implemented as software or as a combination of software and hardware, and the adjustment device pointed to by the microphone array can be integrated into the online control headset. As shown in FIG. 1, the method includes the following steps:
  • the microphone array may be a dual microphone array, or may be a multi-microphone array composed of more than two microphones.
  • the multi-channel voice signal is in one-to-one correspondence with a plurality of microphones in the microphone array, that is, a continuous voice signal of the user received by each microphone.
  • the headset may exhibit various forms due to different personal usage habits, resulting in the microphone array may exhibit different position patterns. And if the main lobe direction of the microphone array is not adjusted, then At different positions, the direction of the main lobe of the microphone array is different, and it is likely that it will not point to the direction of the sound source, resulting in poor quality of the final synthesized speech signal based on the received multi-channel speech signal, which affects the call effect.
  • the direction of the main lobe of the microphone array is adaptively adjusted so that the main lobe direction is directed to the sound source direction.
  • the microphone array In order to adjust the main lobe direction of the microphone array to point to the sound source direction, it is first necessary to determine the sound source incident angle with respect to the positional shape maintained by the current microphone array, so as to adjust the microphone array by beamforming according to the sound source incident angle. The main lobe is pointed so that it points in the direction of the sound source.
  • correlation analysis may be performed on at least two voice signals in the plurality of voice signals, and then the sound source is determined to the microphone array according to the correlation analysis result and the propagation distance difference of the at least two voice signals. Sound source incident angle.
  • the at least two voice signals may be all or part of the microphones randomly selected from the microphone array.
  • performing correlation analysis on the at least two voice signals may be implemented by using a correlation function, or may be implemented by using a convolution operation.
  • the main purpose of the correlation analysis may be to find a propagation parameter that makes the at least two voice signals most relevant, and the propagation parameter may be a propagation time difference.
  • the time when the same voice signal from the same sound source is transmitted to the microphone 1 and the microphone 2 is different.
  • the same voice signal propagates to the microphone 1 at a certain time T1
  • T2 propagates to the microphone 2 at another time T2
  • the time difference of T2-T1 is such that the microphone 1 and the microphone 2
  • the two received voice signals have the strongest correlation time difference.
  • the correlation time analysis obtains the propagation time difference of at least two voice signals, based on the correspondence between the propagation time difference and the propagation distance difference and the correspondence between the combined propagation distance difference and the sound source incident angle, the sound source incident angle corresponding to the microphone array can be determined.
  • the main lobe direction of the microphone array is adjusted by beamforming to point to the sound source direction.
  • the multi-channel voice signal is received through the microphone array including the plurality of microphones, and then the correlation analysis is performed on at least two voice signals in the multi-channel voice signal, based on the correlation.
  • the result of the sexual analysis and the relationship between the propagation distance difference of at least two voice signals and the analysis result can determine the incident angle of the sound source to the sound source of the microphone array when the online control earphone is in the current state, so as to be based on the sound source
  • the incident angle adjusts the main lobe direction of the microphone array such that the main lobe direction is directed to the direction of the incident angle of the sound to ensure the signal quality of the output call speech.
  • FIG. 4 is a flowchart of Embodiment 2 of a method for adjusting a microphone array pointing according to an embodiment of the present invention. As shown in FIG. 4, the method may include the following steps:
  • T 12 is a range of propagation time difference between the first speech signal s1(t) and the second speech signal s2(t)
  • T 12 [-L 12 /c, L 12 /c]
  • L 12 is the first speech signal S1(t) is the distance between the two microphones corresponding to the second speech signal s2(t)
  • c is the speed of sound.
  • a 12 is a difference in propagation distance between the first speech signal s1(t) and the second speech signal s2(t).
  • the microphone 1 and the microphone 2 are included in the microphone array, the microphone 1 and the microphone 2 are respectively used to receive the first voice signal s1(t) and the second voice signal s2(t), and the processing of the above steps can be completed. Adjustment of the direction of the main lobe of the microphone array.
  • the microphone array includes not only the microphone 1 and the microphone 2 but also other microphones such as the microphone 3 and the microphone 4, the microphone 1 for receiving the first voice signal s1(t) and the second voice signal s2(t) And the microphone 2 can be any two microphones in the microphone array.
  • correlation analysis is performed on at least two voice signals of the multi-channel voice signals received by the microphone array, and the correlation function may be used for analysis.
  • the correlation function corr[s1(t), s2(t+T 12 )] of the first speech signal s1(t) and the second speech signal s2(t) may be first calculated to correspond to the maximum value.
  • the propagation time difference is T 12(0) .
  • the first sound source incident angle ⁇ 12 is obtained based on the above formula.
  • the main lobe direction of the microphone array is adjusted according to the first sound source incident angle ⁇ 12 .
  • the principle of determining the incident angle of the first sound source can be seen in FIG. 5.
  • the fourth speech signal s4(t) is the same as or different from the first speech signal s1(t) or the second speech signal s2(t).
  • the first sound source incident angle ⁇ 12 is used as the sole basis for adjusting the main lobe direction of the microphone array.
  • the microphone array when the microphone array further includes other microphones, it can continue to be combined with other microphones to adjust the incident angle ⁇ 12 based on the first sound source.
  • the direction of the main lobe of the rear microphone array is then adjusted.
  • the microphone used for performing the re-adjustment may be completely identical to the microphone 1 and the microphone 2 that have already been used. For example, if the microphone array 3 is included in the microphone array, the microphone 3 and the microphone 1 may be combined.
  • the microphone 2 adjusts the direction of the main lobe of the microphone array.
  • a processing method for performing correlation analysis on the first speech signal s1(t) and the second speech signal s2(t) can obtain a propagation time difference T corresponding to the third speech signal s3(t) and the fourth speech signal s4(t) 34(0) , thereby determining a corresponding second sound source incident angle ⁇ 34 .
  • the microphone array adjusted based on the first sound source incident angle ⁇ 12 is previously described.
  • the main lobe direction of the microphone array is adjusted again according to the second sound source incident angle ⁇ 34 , and conversely, if the main sound source is adjusted according to the first sound source incident angle ⁇ 12
  • the angular deviation of the lobes direction from the second sound source incident angle ⁇ 34 is less than or equal to a preset threshold, indicating that the main lobe direction of the microphone array adjusted based on the first sound source incident angle ⁇ 12 has been able to point well to the sound source direction. , there is no need to adjust again.
  • the preset threshold can be set at about 5 degrees.
  • the multi-channel voice signal by performing correlation analysis on at least two voice signals in the multi-channel voice signal to determine an incident angle of the sound source to the sound source of the microphone array when the online control earphone is in the current state, so as to be based on the The incident angle of the sound source adjusts the main lobe direction of the microphone array such that the main lobe direction is directed to the incident angle of the sound to ensure the signal quality of the output call speech.
  • FIG. 6 is a flowchart of Embodiment 3 of a method for adjusting a microphone array pointing according to an embodiment of the present invention. As shown in FIG. 6, the method may include the following steps:
  • the microphone array includes a microphone 1, a microphone 2, and a microphone 3 for receiving a first voice signal, a second voice signal, and a third voice signal, respectively.
  • a corresponding propagation time difference T 12 (0) is obtained , and further, based on the propagation time difference T 12 (0)
  • the first source is incident angle ⁇ 12 .
  • the method of the above embodiment can be referred to, and the corresponding propagation time difference T 13 (0) is obtained, and then the propagation time difference T 13 (0) is determined.
  • the two sources have an incident angle ⁇ 13 .
  • the corresponding propagation time difference T 23(0) is obtained, and then the propagation time difference T 23(0) is determined.
  • determining the sound source incident angle of the sound source to the microphone array is: the mean value of the sound source incident angle corresponding to any two voice signals, that is, the mean value of ⁇ 12 , ⁇ 13 , ⁇ 23 , according to the average sound source incident.
  • the angle of the main lobe of the microphone array is adjusted such that the main lobe direction is directed to the incident angle of the sound to ensure the signal quality of the outputted call speech.
  • FIG. 7 is a schematic structural diagram of an apparatus for adjusting a microphone array pointing according to an embodiment of the present invention. As shown in FIG. 7, the apparatus includes: a receiving module 11, a determining module 12, and an adjusting module 13.
  • the receiving module 11 is configured to receive, by using a microphone array, a plurality of voice signals, where the multiple voice signals are in one-to-one correspondence with the plurality of microphones in the microphone array.
  • a determining module 12 configured to perform correlation analysis on at least two voice signals in the multiple voice signals; and according to the correlation analysis result and a propagation distance difference between the at least two voice signals, The angle of incidence of the sound source to the sound source of the microphone array is determined.
  • the adjusting module 13 is configured to adjust a main lobe direction of the microphone array according to the sound source incident angle.
  • the at least two voice signals include a first voice signal s1(t) and a second voice signal s2(t), and the determining module 12 includes: a first determining unit 121 and a second determining unit 122. .
  • the first determining unit 121 is configured to determine that the correlation function corr[s1(t), s2(t+T 12 )] of the first speech signal s1(t) and the second speech signal s2(t) is at a maximum The propagation time difference T 12(0) corresponding to the value.
  • T 12 is a range of propagation time difference between the first speech signal s1(t) and the second speech signal s2(t)
  • T 12 [-L 12 /c, L 12 /c]
  • L 12 is The distance between the two microphones corresponding to the first speech signal s1(t) and the second speech signal s2(t)
  • c is the speed of sound.
  • the second determining unit 122 is configured to determine the first sound source incident angle ⁇ 12 according to the following formula:
  • the adjustment module 13 includes: a first adjustment unit 131.
  • the first adjusting unit 131 is configured to adjust a main lobe direction of the microphone array according to the first sound source incident angle ⁇ 12 .
  • the at least two voice signals further include at least a third voice signal s3(t), and the determining module 12 further includes: a third determining unit 123.
  • a third determining unit 123 configured to determine a second sound source incident angle ⁇ 34 according to the third voice signal s3(t) and the fourth voice signal s4(t), wherein the fourth voice signal s4(t ) the same or different from the first speech signal s1(t) or the second speech signal s2(t).
  • the implementation process of the third determining unit 123 may refer to the implementation process of the first determining unit 121 and the second determining unit 122.
  • the adjustment module 13 further includes: a second adjustment unit 132.
  • the second adjusting unit 132 is configured to: if the angular deviation of the main lobe direction and the second sound source incident angle ⁇ 34 adjusted according to the first sound source incident angle ⁇ 12 is greater than a preset threshold, according to the The two source incident angles ⁇ 34 then adjust the main lobe direction of the microphone array.
  • the determining module 12 further includes: a fourth determining unit 124, a fifth determining unit 125, The sixth determining unit 126.
  • the fourth determining unit 124 is configured to determine, for any two of the multiple voice signals, a propagation time difference corresponding to a correlation function of the any two voice signals at a maximum value.
  • a fifth determining unit 125 configured to determine, according to a propagation distance difference between the two voice signals and a propagation time difference corresponding to a correlation function of the any two voice signals at a maximum value, determining that the two voice signals correspond to Sound source incident angle.
  • the sixth determining unit 126 is configured to determine an incident angle of the sound source of the sound source to the microphone array as: an average value of incident angles of sound sources corresponding to any two voice signals.
  • the apparatus shown in FIG. 7 can perform the method of the embodiment shown in FIG. 1, FIG. 4, and FIG. 6.
  • the apparatus shown in FIG. 7 can perform the method of the embodiment shown in FIG. 1, FIG. 4, and FIG. 6.
  • FIG. 7 For the parts not described in detail in this embodiment, reference may be made to the related description of the embodiment shown in FIG. 1, FIG. 4, and FIG.
  • FIG. 1 , FIG. 4 and FIG. 6 For the implementation process and technical effects of the technical solution, reference is made to the description in the embodiment shown in FIG. 1 , FIG. 4 and FIG. 6 , and details are not described herein again.
  • the structure of the adjustment device pointed by the microphone array can be implemented as an electronic device, such as an earphone, as shown in FIG.
  • the electronic device can include a processor 21 and a memory 22.
  • the memory 22 is configured to store a program for the adjustment device that supports the microphone array pointing to perform the adjustment method of the microphone array pointing provided in the foregoing method embodiments, and the processor 21 is configured to execute the memory 22 Stored program.
  • the program includes one or more computer instructions, wherein the one or more computer instructions are executed by the processor 21 to implement the following steps:
  • the main lobe direction of the microphone array is adjusted according to the sound source incident angle.
  • the processor 21 is further configured to perform all or part of the steps in the foregoing embodiment shown in FIG. Step.
  • the structure of the adjustment device pointed by the microphone array may further include a communication interface 23, and the adjustment device for the microphone array pointing communicates with other devices or communication networks.
  • an embodiment of the present invention provides a computer storage medium, which is used to store computer software instructions used by an adjustment device pointed by a microphone array, and includes a program for performing an adjustment method of a microphone array pointing in each of the foregoing method embodiments. .
  • embodiments of the present invention can be provided as a method, system, or computer program product. Accordingly, the present invention may take the form of an entirely hardware embodiment, an entirely software embodiment, or a combination of software and hardware. Moreover, the invention can take the form of a computer program product embodied on one or more computer-usable storage media (including but not limited to disk storage, CD-ROM, optical storage, etc.) including computer usable program code.
  • computer-usable storage media including but not limited to disk storage, CD-ROM, optical storage, etc.
  • the computer program instructions can also be stored in a computer readable memory that can direct a computer or other programmable data processing device to operate in a particular manner, such that the instructions stored in the computer readable memory produce an article of manufacture comprising the instruction device.
  • the apparatus implements the functions specified in one or more blocks of a flow or a flow and/or block diagram of the flowchart.
  • These computer program instructions can also be loaded onto a computer or other programmable data processing device such that a series of operational steps are performed on a computer or other programmable device to produce computer-implemented processing for execution on a computer or other programmable device.
  • the instructions provide steps for implementing the functions specified in a block or blocks of a flow or a flow and/or a block diagram of a flowchart Step.
  • a computing device includes one or more processors (CPUs), input/output interfaces, network interfaces, and memory.
  • processors CPUs
  • input/output interfaces network interfaces
  • memory volatile and non-volatile memory
  • the memory may include non-persistent memory, random access memory (RAM), and/or non-volatile memory in a computer readable medium, such as read only memory (ROM) or flash memory.
  • RAM random access memory
  • ROM read only memory
  • Memory is an example of a computer readable medium.
  • Computer readable media includes both permanent and non-persistent, removable and non-removable media.
  • Information storage can be implemented by any method or technology.
  • the information can be computer readable instructions, data structures, modules of programs, or other data.
  • Examples of computer storage media include, but are not limited to, phase change memory (PRAM), static random access memory (SRAM), dynamic random access memory (DRAM), other types of random access memory (RAM), read only memory. (ROM), electrically erasable programmable read only memory (EEPROM), flash memory or other memory technology, compact disk read only memory (CD-ROM), digital versatile disk (DVD) or other optical storage, Magnetic tape cartridges, magnetic tape storage or other magnetic storage devices or any other non-transportable media can be used to store information that can be accessed by a computing device.
  • computer readable media does not include temporary storage of computer readable media, such as modulated data signals and carrier waves.
  • embodiments of the present invention can be provided as a method, system, or computer program product. Accordingly, the present invention may take the form of an entirely hardware embodiment, an entirely software embodiment, or a combination of software and hardware. Moreover, the invention can take the form of a computer program product embodied on one or more computer-usable storage media (including but not limited to disk storage, CD-ROM, optical storage, etc.) including computer usable program code.
  • computer-usable storage media including but not limited to disk storage, CD-ROM, optical storage, etc.

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Abstract

本发明实施例提供一种麦克风阵列指向的调整方法和装置，该方法包括：通过麦克风阵列接收多路语音信号,多路语音信号与麦克风阵列中的多个麦克风一一对应；对多路语音信号中的至少两路语音信号进行相关性分析；根据相关性分析结果和该至少两路语音信号的传播距离差，确定声源向麦克风阵列的声源入射角；根据声源入射角调整麦克风阵列的主瓣方向，使得主瓣方向指向该声音入射角方向，以保证输出的通话语音的信号质量。

Description

麦克风阵列指向的调整方法和装置
交叉引用
本发明引用于2017年5月19日递交的名称为“麦克风阵列指向的调整方法和装置”的第201710359446.7号中国专利申请，其通过引用被全部并入本发明。
技术领域
本发明涉及麦克风技术领域，尤其涉及一种麦克风阵列指向的调整方法和装置。
背景技术
线控耳机已经在人们的日常生活中被广泛使用。比如，人们可以通过线控耳机控制诸如手机终端中音乐的播放、音量的调整，还可以通过线控耳机与朋友进行语音通话。
为了保证通话质量，在线控耳机中一般都会设置有麦克风阵列，通常使用的麦克风阵列包含双麦克阵列，当然，也有包含更多麦克风的麦克风阵列。
在使用线控耳机进行通话的过程中，由于用户的使用习惯不同，有的喜欢线控自然下垂，有的喜欢用手拿着线控，因此容易导致麦克风阵列的指向发生改变，使得在通话过程中麦克风阵列并非保持指向用户嘴部方向，即声源方向，导致通话语音质量不佳。
发明内容
本发明的多个方面提供一种麦克风阵列指向的调整方法和装置，通过调整麦克风阵列一直指向声源方向，以保证通话质量。
第一方面，本发明实施例提供一种麦克风阵列指向的调整方法，包括：
通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应；
对所述多路语音信号中的至少两路语音信号进行相关性分析；
根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角；
根据所述声源入射角调整所述麦克风阵列的主瓣方向。
可选地，所述至少两路语音信号包括第一语音信号s1(t)和第二语音信号s2(t)，以及，所述对所述多路语音信号中的至少两路语音信号进行相关性分析，包括：
确定所述第一语音信号s1(t)与所述第二语音信号s2(t)的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)；
其中，T12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播时间差范围，T12＝[-L12/c,L12/c]，L12为所述第一语音信号s1(t)与所述第二语音信号s2(t)对应的两个麦克风之间的距离，c为声速。
可选地，所述根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角，包括：
根据如下公式确定第一声源入射角θ12：a12＝L12·cos(θ12)＝T12(0)·c；
其中，a12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播距离差；
所述根据所述声源入射角调整所述麦克风阵列的主瓣方向，包括：
根据所述第一声源入射角θ12调整所述麦克风阵列的主瓣方向。
可选地，所述至少两路语音信号还至少包括第三语音信号s3(t)，所述方法 还包括：
根据所述第三语音信号s3(t)和第四语音信号s4(t)，确定第二声源入射角θ34，其中，所述第四语音信号s4(t)与所述第一语音信号s1(t)或所述第二语音信号s2(t)相同或不同；
若根据所述第一声源入射角θ12调整后的主瓣方向与所述第二声源入射角θ34的角度偏差大于预设阈值，则根据所述第二声源入射角θ34再调整所述麦克风阵列的主瓣方向。
可选地，所述对所述多路语音信号中的至少两路语音信号进行相关性分析，包括：
对于所述多路语音信号中的任意两个语音信号，确定所述任意两个语音信号的相关函数在最大值时所对应的传播时间差；
所述根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角，包括：
根据所述任意两个语音信号的传播距离差和所述任意两个语音信号的相关函数在最大值时所对应的传播时间差，确定所述任意两个语音信号对应的声源入射角；
确定所述声源向所述麦克风阵列的第二声源入射角为：全部的任意两个语音信号对应的声源入射角的均值。
第二方面，本发明实施例提供一种麦克风阵列指向的调整装置，包括：
接收模块，用于通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应；
确定模块，用于对所述多路语音信号中的至少两路语音信号进行相关性分析；根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角；
调整模块，用于根据所述声源入射角调整所述麦克风阵列的主瓣方向。
在一个可能的设计中，上述麦克风阵列指向的调整装置的结构中包括处理器和存储器，所述存储器用于存储支持麦克风阵列指向的调整装置执行上 述第一方面中麦克风阵列指向的调整方法的程序，所述处理器被配置为用于执行所述存储器中存储的程序。所述麦克风阵列指向的调整装置还可以包括通信接口，用于麦克风阵列指向的调整装置与其他设备或通信网络通信。
本发明实施例提供了一种计算机存储介质，用于储存麦克风阵列指向的调整装置所用的计算机软件指令，其包含用于执行上述第一方面中麦克风阵列指向的调整方法所涉及的程序。
本发明提供的麦克风阵列指向的调整方法和装置，当用户触发通过线控耳机通话时，通过包含多个麦克风的麦克风阵列接收多路语音信号，进而通过对多路语音信号中的至少两路语音信号进行相关性分析，基于相关性分析结果和至少两路语音信号的传播距离差与该分析结果的关联关系，可以确定在线控耳机处于当前的状态下时，声源向麦克风阵列的声源入射角，以便于基于该声源入射角调整麦克风阵列的主瓣方向，使得主瓣方向指向该声音入射角方向，以保证输出的通话语音的信号质量。
附图说明
此处所说明的附图用来提供对本发明的进一步理解，构成本发明的一部分，本发明的示意性实施例及其说明用于解释本发明，并不构成对本发明的不当限定。在附图中：
图1为本发明实施例提供的麦克风阵列指向的调整方法实施例一的流程图；
图2为麦克风阵列在实际使用时的位置形态的示意图；
图3为一种双麦克风阵列的结构示意图；
图4为本发明实施例提供的麦克风阵列指向的调整方法实施例二的流程图；
图5为确定声源入射角的原理示意图；
图6为本发明实施例提供的麦克风阵列指向的调整方法实施例三的流程图；
图7为本发明实施例提供的麦克风阵列指向的调整装置的结构示意图；
图8为与图7所示实施例提供的麦克风阵列指向的调整装置对应的电子设备的结构示意图。
具体实施方式
为使本发明的目的、技术方案和优点更加清楚，下面将结合本发明具体实施例及相应的附图对本发明技术方案进行清楚、完整地描述。显然，所描述的实施例仅是本发明一部分实施例，而不是全部的实施例。基于本发明中的实施例，本领域普通技术人员在没有做出创造性劳动前提下所获得的所有其他实施例，都属于本发明保护的范围。
在本发明实施例中使用的术语是仅仅出于描述特定实施例的目的，而非旨在限制本发明。在本发明实施例和所附权利要求书中所使用的单数形式的“一种”、“所述”和“该”也旨在包括多数形式，除非上下文清楚地表示其他含义，“多种”一般包含至少两种。
应当理解，本文中使用的术语“和/或”仅仅是一种描述关联对象的关联关系，表示可以存在三种关系，例如，A和/或B，可以表示：单独存在A，同时存在A和B，单独存在B这三种情况。另外，本文中字符“/”，一般表示前后关联对象是一种“或”的关系。
应当理解，尽管在本发明实施例中可能采用术语第一、第二、第三等来描述XXX，但这些XXX不应限于这些术语。这些术语仅用来将XXX区分开。例如，在不脱离本发明实施例范围的情况下，第一XXX也可以被称为第二XXX，类似地，第二XXX也可以被称为第一XXX。
取决于语境，如在此所使用的词语“如果”、“若”可以被解释成为“在…… 时”或“当……时”或“响应于确定”或“响应于检测”。类似地，取决于语境，短语“如果确定”或“如果检测(陈述的条件或事件)”可以被解释成为“当确定时”或“响应于确定”或“当检测(陈述的条件或事件)时”或“响应于检测(陈述的条件或事件)”。
还需要说明的是，术语“包括”、“包含”或者其任何其他变体意在涵盖非排他性的包含，从而使得包括一系列要素的商品或者系统不仅包括那些要素，而且还包括没有明确列出的其他要素，或者是还包括为这种商品或者系统所固有的要素。在没有更多限制的情况下，由语句“包括一个……”限定的要素，并不排除在包括所述要素的商品或者系统中还存在另外的相同要素。
进一步值得说明的是，本发明各实施例中各步骤之间的顺序是可以调整的，不是必须按照以下举例的顺序执行。
图1为本发明实施例提供的麦克风阵列指向的调整方法实施例一的流程图，本实施例提供的该麦克风阵列指向的调整方法可以由一麦克风阵列指向的调整装置来执行，该麦克风阵列指向的调整装置可以实现为软件，或者实现为软件和硬件的组合，该麦克风阵列指向的调整装置可以集成设置在线控耳机中。如图1所示，该方法包括如下步骤：
101、通过麦克风阵列接收多路语音信号。
102、对多路语音信号中的至少两路语音信号进行相关性分析。
103、根据相关性分析结果和至少两路语音信号的传播距离差，确定声源向麦克风阵列的声源入射角。
104、根据声源入射角调整麦克风阵列的主瓣方向。
本发明实施例中，该麦克风阵列可以是双麦克风阵列，也可以是由超过两个麦克风组成的多麦克风阵列。上述多路语音信号与麦克风阵列中的多个麦克风一一对应，即为各麦克风接收到的用户的连续语音信号。
如图2所示，用户在使用含有麦克风阵列的耳机进行通话的过程中，由于个人使用习惯的不同，耳机可能呈现各种不同的形态，导致麦克风阵列可能呈现不同的位置形态。而如果不对麦克风阵列的主瓣方向进行调整，那么在 不同位置处，麦克风阵列的主瓣方向指向不同，很可能不会指向声源方向，导致基于接收到的多路语音信号最终合成的语音信号质量不佳，影响通话效果。
因此，本发明实施例中，为保证语音通话质量，当用户触发语音通话需求后，通过对麦克风阵列主瓣方向进行适应性调整，使得其主瓣方向指向声源方向。
可选地，本实施例提供的上述麦克风阵列指向的调整方法的执行时机，可以是在用户触发呼叫某用户，该用户接听后开始，也可以是在用户被呼叫，触发接听电话开始，还可以是在用户按下触发调整的某预设按钮后开始。如图3所示，图3中示意出的双麦克风阵列结构中，可以设置一按钮，当用户开始通话时，或者在通话的过程中改变耳机的位置形态时，可以按下该按钮，以触发对麦克风阵列主瓣方向的调整。
因此，当上述举例的触发条件满足时，可以通过麦克风阵列接收到多路语音信号。为了调整麦克风阵列的主瓣方向使其指向声源方向，首先需要确定出相对于当前麦克风阵列所保持的位置形态的声源入射角，以便根据该声源入射角，通过波束成形调整麦克风阵列的主瓣指向，使其指向声源方向。
而确定声源入射角，可以先对多路语音信号中的至少两路语音信号进行相关性分析，进而根据相关性分析结果和该至少两路语音信号的传播距离差，确定声源向麦克风阵列的声源入射角。
可选地，当麦克风阵列中包含了三个或三个以上的麦克风时，上述至少两路语音信号可以是从麦克风阵列中随机选定的全部或部分麦克风。
可选地，对上述至少两路语音信号进行相关性分析，可以采用相关函数的方式实现，也可以采用卷积运算的方式实现。该相关性分析的主要目的可以是：寻找使得该至少两路语音信号相关程度最强的传播参数，该传播参数可以是传播时间差。
以任意两个麦克风为例，由于麦克风1和麦克风2相距声源的距离不同，同一声源发出的同一语音信号传播到麦克风1和麦克风2时的时间也不相同， 比如：同一语音信号在某时刻T1传播到麦克风1，由麦克风1接收，在之后的另一时刻T2传播到麦克风2，由麦克风2接收，则T2-T1这一时间差是使得麦克风1和麦克风2接收到的两路语音信号相关性最强的传播时间差。
经过上述相关性分析得到至少两路语音信号的传播时间差后，基于传播时间差与传播距离差的对应关系以及结合传播距离差与声源入射角的对应关系，可以确定麦克风阵列对应的声源入射角，从而通过波束成形调整麦克风阵列的主瓣方向，以指向声源方向。
本实施例中，当用户触发通过线控耳机通话时，通过包含多个麦克风的麦克风阵列接收多路语音信号，进而通过对多路语音信号中的至少两路语音信号进行相关性分析，基于相关性分析结果和至少两路语音信号的传播距离差与该分析结果的关联关系，可以确定在线控耳机处于当前的状态下时，声源向麦克风阵列的声源入射角，以便于基于该声源入射角调整麦克风阵列的主瓣方向，使得主瓣方向指向该声音入射角方向，以保证输出的通话语音的信号质量。
图4为本发明实施例提供的麦克风阵列指向的调整方法实施例二的流程图，如图4所示，可以包括如下步骤：
201、通过麦克风阵列接收多路语音信号。
202、对于多路语音信号中的第一语音信号s1(t)和第二语音信号s2(t)，确定第一语音信号s1(t)与第二语音信号s2(t)的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)。
其中，T12为第一语音信号s1(t)与第二语音信号s2(t)的传播时间差范围，T12＝[-L12/c,L12/c]，L12为第一语音信号s1(t)与第二语音信号s2(t)对应的两个麦克风之间的距离，c为声速。
203、根据如下公式确定第一声源入射角θ12：a12＝L12·cos(θ12)＝T12(0)·c。
其中，a12为第一语音信号s1(t)与第二语音信号s2(t)的传播距离差。
204、根据第一声源入射角θ12调整麦克风阵列的主瓣方向。
在麦克风阵列中仅包含麦克风1和麦克风2时，麦克风1和麦克风2分别用于接收上述第一语音信号s1(t)与第二语音信号s2(t)，则通过上述步骤的处理即可完成对麦克风阵列主瓣方向指向的调整。而当麦克风阵列中不仅包含麦克风1和麦克风2，还包括其他麦克风比如麦克风3、麦克风4时，用于接收上述第一语音信号s1(t)与第二语音信号s2(t)的上述麦克风1和麦克风2可以是麦克风阵列中的任意两个麦克风。
本实施例中，对麦克风阵列接收到的多路语音信号中的至少两路语音信号进行相关性分析，可以采用相关函数的方式进行分析。具体来说，可以先计算求得第一语音信号s1(t)与第二语音信号s2(t)的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)。
具体地，可以预先存储有麦克风1和麦克风2之间的距离L12，以便基于该距离确定第一语音信号s1(t)与第二语音信号s2(t)的传播时间差的取值范围：T12＝[-L12/c,L12/c]。进而，可以以一定步长逐个尝试该取值范围内的各传播时间差取值，以得到在取不同传播时间差取值时相关函数的函数值，从中选出函数值最大时对应的传播时间差T12(0)。进而，基于传播距离差与传播时间差、两个麦克风相距的距离、声源入射角的对应关系，即基于上述公式得到第一声源入射角θ12。根据第一声源入射角θ12调整麦克风阵列的主瓣方向。该第一声源入射角的确定原理可以参见图5所示。
205、对于多路语音信号中的第三语音信号s3(t)和第四语音信号s4(t)，根据第三语音信号s3(t)和第四语音信号s4(t)，确定第二声源入射角θ34。
其中，第四语音信号s4(t)与第一语音信号s1(t)或第二语音信号s2(t)相同或不同。
206、若根据第一声源入射角θ12调整后的主瓣方向与第二声源入射角θ34的角度偏差大于预设阈值，则根据第二声源入射角θ34再调整麦克风阵列的主瓣方向。
本实施例中，可选地，可以不管麦克风阵列中除了麦克风1和麦克风2之外是否还包含其他麦克风，以第一声源入射角θ12作为唯一的调整麦克风阵列 主瓣方向的依据。但是，为了提高调整准确性，即提高麦克风阵列主瓣方向指向声源方向的准确性，当麦克风阵列中还包含其他麦克风时，还可以继续结合其他麦克风对基于第一声源入射角θ12调整后的麦克风阵列的主瓣方向再进行调整。此时，进行该再次调整时所采用的麦克风只要是不与已经使用的麦克风1和麦克风2完全一致即可，比如，假设麦克风阵列中还包含麦克风3，则可以再结合麦克风3与麦克风1或麦克风2对麦克风阵列的主瓣方向再进行调整。
从而，假设上述的第三语音信号s3(t)和第四语音信号s4(t)为与第一语音信号s1(t)与第二语音信号s2(t)不同的两路语音信号，则参见对第一语音信号s1(t)与第二语音信号s2(t)进行相关性分析的处理方法，可以得到第三语音信号s3(t)和第四语音信号s4(t)对应的传播时间差T34(0)，进而确定对应的第二声源入射角θ34。当根据第一声源入射角θ12调整后的主瓣方向与第二声源入射角θ34的角度偏差大于预设阈值时，说明之前基于第一声源入射角θ12调整后的麦克风阵列的主瓣方向未能够很好地指向声源方向，则根据第二声源入射角θ34再次调整麦克风阵列的主瓣方向，相反地，如果根据第一声源入射角θ12调整后的主瓣方向与第二声源入射角θ34的角度偏差小于或等于预设阈值，说明之前基于第一声源入射角θ12调整后的麦克风阵列的主瓣方向已经能够很好地指向声源方向，则无需再次调整。一般地，该预设阈值可以设置在5度左右。
本实施例中，通过对多路语音信号中的至少两路语音信号进行相关性分析，以确定在线控耳机处于当前的状态下时，声源向麦克风阵列的声源入射角，以便于基于该声源入射角调整麦克风阵列的主瓣方向，使得主瓣方向指向该声音入射角方向，以保证输出的通话语音的信号质量。
图6为本发明实施例提供的麦克风阵列指向的调整方法实施例三的流程图，如图6所示，可以包括如下步骤：
301、通过麦克风阵列接收多路语音信号。
302、对于多路语音信号中的任意两个语音信号，确定任意两个语音信号 的相关函数在最大值时所对应的传播时间差。
303、根据所述任意两个语音信号的传播距离差和所述任意两个语音信号的相关函数在最大值时所对应的传播时间差，确定所述任意两个语音信号对应的声源入射角。
304、确定声源向麦克风阵列的声源入射角为：全部的任意两个语音信号对应的声源入射角的均值。
305、根据声源入射角调整麦克风阵列的主瓣方向。
本实施例中，假设麦克风阵列包含麦克风1、麦克风2、麦克风3，分别用于接收第一语音信号、第二语音信号和第三语音信号。针对麦克风1和麦克风2接收到的第一语音信号和第二语音信号，可以参见上述实施例的方法，得到对应的传播时间差T12(0)，进而，基于该传播时间差T12(0)确定第一声源入射角θ12。针对麦克风1和麦克风3接收到的第一语音信号和第三语音信号，可以参见上述实施例的方法，得到对应的传播时间差T13(0)进而，基于该传播时间差T13(0)确定第二声源入射角θ13。针对麦克风2和麦克风3接收到的第二语音信号和第三语音信号，可以参见上述实施例的方法，得到对应的传播时间差T23(0)进而，基于该传播时间差T23(0)确定第三声源入射角θ23。最后，确定声源向麦克风阵列的声源入射角为：全部的任意两个语音信号对应的声源入射角的均值，即为θ12、θ13、θ23的均值，根据该平均声源入射角调整麦克风阵列的主瓣方向，使得主瓣方向指向该声音入射角方向，以保证输出的通话语音的信号质量。
图7为本发明实施例提供的麦克风阵列指向的调整装置的结构示意图，如图7所示，该装置包括：接收模块11、确定模块12、调整模块13。
接收模块11，用于通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应。
确定模块12，用于对所述多路语音信号中的至少两路语音信号进行相关性分析；根据所述相关性分析结果和所述至少两路语音信号的传播距离差， 确定声源向所述麦克风阵列的声源入射角。
调整模块13，用于根据所述声源入射角调整所述麦克风阵列的主瓣方向。
可选地，所述至少两路语音信号包括第一语音信号s1(t)和第二语音信号s2(t)，以及，所述确定模块12包括：第一确定单元121、第二确定单元122。
第一确定单元121，用于确定所述第一语音信号s1(t)与所述第二语音信号s2(t)的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)。
其中，T12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播时间差范围，T12＝[-L12/c,L12/c]，L12为所述第一语音信号s1(t)与所述第二语音信号s2(t)对应的两个麦克风之间的距离，c为声速。
第二确定单元122，用于根据如下公式确定第一声源入射角θ12：
a12＝L12·cos(θ12)＝T12(0)·c；其中，a12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播距离差。
可选地，所述调整模块13包括：第一调整单元131。
第一调整单元131，用于根据所述第一声源入射角θ12调整所述麦克风阵列的主瓣方向。
可选地，所述至少两路语音信号还至少包括第三语音信号s3(t)，所述确定模块12还包括：第三确定单元123。
第三确定单元123，用于根据所述第三语音信号s3(t)和第四语音信号s4(t)，确定第二声源入射角θ34，其中，所述第四语音信号s4(t)与所述第一语音信号s1(t)或所述第二语音信号s2(t)相同或不同。
其中，第三确定单元123的实现过程可以参见第一确定单元121和第二确定单元122的实现过程。
所述调整模块13还包括：第二调整单元132。
第二调整单元132，用于若根据所述第一声源入射角θ12调整后的主瓣方向与所述第二声源入射角θ34的角度偏差大于预设阈值，则根据所述第二声源入射角θ34再调整所述麦克风阵列的主瓣方向。
可选地，所述确定模块12还包括：第四确定单元124、第五确定单元125、 第六确定单元126。
第四确定单元124，用于对于所述多路语音信号中的任意两个语音信号，确定所述任意两个语音信号的相关函数在最大值时所对应的传播时间差。
第五确定单元125，用于根据所述任意两个语音信号的传播距离差和所述任意两个语音信号的相关函数在最大值时所对应的传播时间差，确定所述任意两个语音信号对应的声源入射角。
第六确定单元126，用于确定所述声源向所述麦克风阵列的声源入射角为：全部的任意两个语音信号对应的声源入射角的均值。
图7所示装置可以执行图1、图4、图6所示实施例的方法，本实施例未详细描述的部分，可参考对图1、图4、图6所示实施例的相关说明。该技术方案的执行过程和技术效果参见图1、图4、图6所示实施例中的描述，在此不再赘述。
以上描述了麦克风阵列指向的调整装置的内部功能和结构，在一个可能的设计中，麦克风阵列指向的调整装置的结构可实现为一电子设备，该电子设备比如为耳机，如图8所示，该电子设备可以包括：处理器21和存储器22。其中，所述存储器22用于存储支持麦克风阵列指向的调整装置执行上述各方法实施例中提供的麦克风阵列指向的调整方法的程序，所述处理器21被配置为用于执行所述存储器22中存储的程序。
所述程序包括一条或多条计算机指令，其中，所述一条或多条计算机指令被所述处理器21执行时能够实现如下步骤：
通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应；
对所述多路语音信号中的至少两路语音信号进行相关性分析；
根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角；
根据所述声源入射角调整所述麦克风阵列的主瓣方向。
可选地，所述处理器21还用于执行前述图5所示实施例中的全部或部分步 骤。
其中，所述麦克风阵列指向的调整装置的结构中还可以包括通信接口23，用于麦克风阵列指向的调整装置与其他设备或通信网络通信。
另外，本发明实施例提供了一种计算机存储介质，用于储存麦克风阵列指向的调整装置所用的计算机软件指令，其包含用于执行上述各方法实施例中麦克风阵列指向的调整方法所涉及的程序。
本领域内的技术人员应明白，本发明的实施例可提供为方法、系统、或计算机程序产品。因此，本发明可采用完全硬件实施例、完全软件实施例、或结合软件和硬件方面的实施例的形式。而且，本发明可采用在一个或多个其中包含有计算机可用程序代码的计算机可用存储介质(包括但不限于磁盘存储器、CD-ROM、光学存储器等)上实施的计算机程序产品的形式。
本发明是参照根据本发明实施例的方法、设备(系统)、和计算机程序产品的流程图和/或方框图来描述的。应理解可由计算机程序指令实现流程图和/或方框图中的每一流程和/或方框、以及流程图和/或方框图中的流程和/或方框的结合。可提供这些计算机程序指令到通用计算机、专用计算机、嵌入式处理机或其他可编程数据处理设备的处理器以产生一个机器，使得通过计算机或其他可编程数据处理设备的处理器执行的指令产生用于实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能的装置。
这些计算机程序指令也可存储在能引导计算机或其他可编程数据处理设备以特定方式工作的计算机可读存储器中，使得存储在该计算机可读存储器中的指令产生包括指令装置的制造品，该指令装置实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能。
这些计算机程序指令也可装载到计算机或其他可编程数据处理设备上，使得在计算机或其他可编程设备上执行一系列操作步骤以产生计算机实现的处理，从而在计算机或其他可编程设备上执行的指令提供用于实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能的步 骤。
在一个典型的配置中，计算设备包括一个或多个处理器(CPU)、输入/输出接口、网络接口和内存。
内存可能包括计算机可读介质中的非永久性存储器，随机存取存储器(RAM)和/或非易失性内存等形式，如只读存储器(ROM)或闪存(flash RAM)。内存是计算机可读介质的示例。
计算机可读介质包括永久性和非永久性、可移动和非可移动媒体可以由任何方法或技术来实现信息存储。信息可以是计算机可读指令、数据结构、程序的模块或其他数据。计算机的存储介质的例子包括，但不限于相变内存(PRAM)、静态随机存取存储器(SRAM)、动态随机存取存储器(DRAM)、其他类型的随机存取存储器(RAM)、只读存储器(ROM)、电可擦除可编程只读存储器(EEPROM)、快闪记忆体或其他内存技术、只读光盘只读存储器(CD-ROM)、数字多功能光盘(DVD)或其他光学存储、磁盒式磁带，磁带磁磁盘存储或其他磁性存储设备或任何其他非传输介质，可用于存储可以被计算设备访问的信息。按照本文中的界定，计算机可读介质不包括暂存电脑可读媒体(transitory media)，如调制的数据信号和载波。
还需要说明的是，术语“包括”、“包含”或者其任何其他变体意在涵盖非排他性的包含，从而使得包括一系列要素的过程、方法、商品或者设备不仅包括那些要素，而且还包括没有明确列出的其他要素，或者是还包括为这种过程、方法、商品或者设备所固有的要素。在没有更多限制的情况下，由语句“包括一个……”限定的要素，并不排除在包括所述要素的过程、方法、商品或者设备中还存在另外的相同要素。
本领域技术人员应明白，本发明的实施例可提供为方法、系统或计算机程序产品。因此，本发明可采用完全硬件实施例、完全软件实施例或结合软件和硬件方面的实施例的形式。而且，本发明可采用在一个或多个其中包含有计算机可用程序代码的计算机可用存储介质(包括但不限于磁盘存储器、CD-ROM、光学存储器等)上实施的计算机程序产品的形式。
以上所述仅为本发明的实施例而已，并不用于限制本发明。对于本领域技术人员来说，本发明可以有各种更改和变化。凡在本发明的精神和原理之内所作的任何修改、等同替换、改进等，均应包含在本发明的权利要求范围之内。

Claims (12)

  1. 一种麦克风阵列指向的调整方法，其中，包括：
    通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应；
    对所述多路语音信号中的至少两路语音信号进行相关性分析；
    根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角；
    根据所述声源入射角调整所述麦克风阵列的主瓣方向。
  2. 根据权利要求1所述的方法，其中，所述至少两路语音信号包括第一语音信号s1(t)和第二语音信号s2(t)，以及，所述对所述多路语音信号中的至少两路语音信号进行相关性分析，包括：
    确定所述第一语音信号s1(t)与所述第二语音信号s2(t)的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)；
    其中，T12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播时间差范围，T12＝[-L12/c,L12/c]，L12为所述第一语音信号s1(t)与所述第二语音信号s2(t)对应的两个麦克风之间的距离，c为声速。
  3. 根据权利要求2所述的方法，其中，所述根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角，包括：
    根据如下公式确定第一声源入射角θ12：a12＝L12·cos(θ12)＝T12(0)·c；
    其中，a12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播距离差；
    所述根据所述声源入射角调整所述麦克风阵列的主瓣方向，包括：
    根据所述第一声源入射角θ12调整所述麦克风阵列的主瓣方向。
  4. 根据权利要求3所述的方法，其中，所述至少两路语音信号还至少包括第三语音信号s3(t)，所述方法还包括：
    根据所述第三语音信号s3(t)和第四语音信号s4(t)，确定第二声源入射角θ34，其中，所述第四语音信号s4(t)与所述第一语音信号s1(t)或所述第二语音信号s2(t)相同或不同；
    若根据所述第一声源入射角θ12调整后的主瓣方向与所述第二声源入射角θ34的角度偏差大于预设阈值，则根据所述第二声源入射角θ34再调整所述麦克风阵列的主瓣方向。
  5. 根据权利要求1所述的方法，其中，所述对所述多路语音信号中的至少两路语音信号进行相关性分析，包括：
    对于所述多路语音信号中的任意两个语音信号，确定所述任意两个语音信号的相关函数在最大值时所对应的传播时间差；
    所述根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角，包括：
    根据所述任意两个语音信号的传播距离差和所述任意两个语音信号的相关函数在最大值时所对应的传播时间差，确定所述任意两个语音信号对应的声源入射角；
    确定所述声源向所述麦克风阵列的第二声源入射角为：全部的任意两个语音信号对应的声源入射角的均值。
  6. 一种麦克风阵列指向的调整装置，其中，包括：
    接收模块，用于通过麦克风阵列接收多路语音信号,所述多路语音信号与所述麦克风阵列中的多个麦克风一一对应；
    确定模块，用于对所述多路语音信号中的至少两路语音信号进行相关性分析；根据所述相关性分析结果和所述至少两路语音信号的传播距离差，确定声源向所述麦克风阵列的声源入射角；
    调整模块，用于根据所述声源入射角调整所述麦克风阵列的主瓣方向。
  7. 根据权利要求6所述的装置，其中，所述至少两路语音信号包括第一语音信号s1(t)和第二语音信号s2(t)，以及，所述确定模块包括：
    第一确定单元，用于确定所述第一语音信号s1(t)与所述第二语音信号s2(t) 的相关函数corr[s1(t),s2(t+T12)]在最大值时所对应的传播时间差T12(0)；
    其中，T12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播时间差范围，T12＝[-L12/c,L12/c]，L12为所述第一语音信号s1(t)与所述第二语音信号s2(t)对应的两个麦克风之间的距离，c为声速。
  8. 根据权利要求7所述的装置，其中，所述确定模块包括：
    第二确定单元，用于根据如下公式确定第一声源入射角θ12：
    a12＝L12·cos(θ12)＝T12(0)·c；
    其中，a12为所述第一语音信号s1(t)与所述第二语音信号s2(t)的传播距离差；
    所述调整模块包括：
    第一调整单元，用于根据所述第一声源入射角θ12调整所述麦克风阵列的主瓣方向。
  9. 根据权利要求8所述的装置，其中，所述至少两路语音信号还至少包括第三语音信号s3(t)，所述确定模块包括：
    第三确定单元，用于根据所述第三语音信号s3(t)和第四语音信号s4(t)，确定第二声源入射角θ34，其中，所述第四语音信号s4(t)与所述第一语音信号s1(t)或所述第二语音信号s2(t)相同或不同；
    所述调整模块还包括：
    第二调整单元，用于若根据所述第一声源入射角θ12调整后的主瓣方向与所述第二声源入射角θ34的角度偏差大于预设阈值，则根据所述第二声源入射角θ34再调整所述麦克风阵列的主瓣方向。
  10. 根据权利要求6所述的装置，其中，所述确定模块还包括：
    第四确定单元，用于对于所述多路语音信号中的任意两个语音信号，确定所述任意两个语音信号的相关函数在最大值时所对应的传播时间差；
    第五确定单元，用于根据所述任意两个语音信号的传播距离差和所述任意两个语音信号的相关函数在最大值时所对应的传播时间差，确定所述任意两个语音信号对应的声源入射角；
    第六确定单元，用于确定所述声源向所述麦克风阵列的声源入射角为：全部的任意两个语音信号对应的声源入射角的均值。
  11. 一种电子设备，其中，包括存储器和处理器；其中，
    所述存储器用于存储一条或多条计算机指令，其中，所述一条或多条计算机指令被所述处理器执行时实现如权利要求1至5中任一项所述的克风阵列指向的调整方法。
  12. 一种存储有计算机程序的计算机可读存储介质，其中，所述计算机程序使计算机执行时实现如权利要求1至5中任一项所述的克风阵列指向的调整方法。
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