WO2022226696A1 - 一种开放式耳机 - Google Patents

一种开放式耳机 Download PDF

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Publication number
WO2022226696A1
WO2022226696A1 PCT/CN2021/089670 CN2021089670W WO2022226696A1 WO 2022226696 A1 WO2022226696 A1 WO 2022226696A1 CN 2021089670 W CN2021089670 W CN 2021089670W WO 2022226696 A1 WO2022226696 A1 WO 2022226696A1
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WO
WIPO (PCT)
Prior art keywords
noise
microphone array
user
microphone
signal
Prior art date
Application number
PCT/CN2021/089670
Other languages
English (en)
French (fr)
Inventor
肖乐
郑金波
张承乾
廖风云
齐心
Original Assignee
深圳市韶音科技有限公司
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by 深圳市韶音科技有限公司 filed Critical 深圳市韶音科技有限公司
Priority to CN202180099448.1A priority Critical patent/CN117501710A/zh
Priority to PCT/CN2021/089670 priority patent/WO2022226696A1/zh
Priority to CN202110486203.6A priority patent/CN115240697A/zh
Priority to CN202180094203.XA priority patent/CN116918350A/zh
Priority to PCT/CN2021/091652 priority patent/WO2022227056A1/zh
Priority to US17/451,659 priority patent/US11328702B1/en
Priority to EP21938133.2A priority patent/EP4131997A4/en
Priority to PCT/CN2021/131927 priority patent/WO2022227514A1/zh
Priority to KR1020227044224A priority patent/KR20230013070A/ko
Priority to JP2022580472A priority patent/JP2023532489A/ja
Priority to CN202111408328.3A priority patent/CN115243137A/zh
Priority to BR112022023372A priority patent/BR112022023372A2/pt
Priority to TW111111172A priority patent/TW202243486A/zh
Priority to US17/657,743 priority patent/US11715451B2/en
Priority to TW111114832A priority patent/TW202242856A/zh
Priority to TW111115388A priority patent/TW202242855A/zh
Priority to US18/047,639 priority patent/US20230063283A1/en
Publication of WO2022226696A1 publication Critical patent/WO2022226696A1/zh
Priority to US18/332,746 priority patent/US20230317048A1/en

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Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • G10L21/0216Noise filtering characterised by the method used for estimating noise
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1083Reduction of ambient noise
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R9/00Transducers of moving-coil, moving-strip, or moving-wire type
    • H04R9/02Details
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R9/00Transducers of moving-coil, moving-strip, or moving-wire type
    • H04R9/06Loudspeakers
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • G10L21/0216Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02166Microphone arrays; Beamforming
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/13Hearing devices using bone conduction transducers

Definitions

  • the present application relates to the field of acoustics, and in particular, to an open earphone.
  • the headset device allows users to listen to audio content and make voice calls while ensuring the privacy of user interaction content, and does not disturb surrounding people when listening.
  • Headphone devices can generally be divided into two categories: in-ear headphone devices and open-back headphone devices.
  • the in-ear headphone device will block the user's ear during use, and the user is prone to experience blockage, foreign body, pain, etc. when wearing it for a long time.
  • the open earphone device can open the user's ear, which is conducive to long-term wearing, but when the external noise is large, its noise reduction effect is not obvious, which makes the user's listening experience poor.
  • an open earphone which can open both ears of a user and improve the user's listening experience.
  • An embodiment of the present application provides an open-type earphone, including: a fixing structure configured to fix the earphone near a user's ear without blocking the user's ear canal; a housing structure configured to carry: a first microphone array , configured to pick up ambient noise; at least one speaker array; and a signal processor, configured to: estimate noise at a first spatial location based on the picked-up ambient noise, the first spatial location than the first microphone array any one of the microphones is closer to the user's ear canal; and a noise reduction signal is generated based on the noise at the first spatial position, so that the at least one speaker array outputs a noise reduction wave according to the noise reduction signal, the noise reduction wave being used for Eliminates ambient noise delivered to the user's ear canal.
  • the housing structure is configured to house a second microphone array configured to pick up ambient noise and the noise reduction wave, the second microphone array being at least partially distinct from the the first microphone array; and the signal processor is configured to update the noise reduction signal based on the sound signal picked up by the second microphone array.
  • the updating the noise reduction signal based on the sound signal picked up by the second microphone array comprises: estimating a sound field at the user's ear canal based on the sound signal picked up by the second microphone array; and The parameter information of the noise reduction signal is adjusted according to the sound field at the user's ear canal.
  • the signal processor is further configured to: obtain user input; and adjust parameter information of the noise reduction signal according to the user input.
  • the second microphone array includes a microphone closer to the user's ear canal than any microphone in the first microphone array.
  • the signal processor estimating the noise in the first space based on the picked-up environmental noise includes: performing signal separation according to the picked-up environmental noise, acquiring parameter information corresponding to the environmental noise, and based on the picked-up environmental noise The parameter information generates a noise reduction signal.
  • the signal processor estimating noise at the first spatial location based on the picked-up ambient noise comprises: determining one or more spatial noise sources associated with the picked-up ambient noise; and based on the spatial noise a noise source, estimating noise at the first spatial location.
  • the determining one or more spatial noise sources associated with the picked-up ambient noise comprises: dividing the picked-up ambient noise into a plurality of subbands, each subband corresponding to a different frequency range; and On at least one subband, a spatial noise source corresponding thereto is determined.
  • the first microphone array includes a first sub-microphone array and a second sub-microphone array
  • the first and second sub-microphone arrays are located at the user's left ear and right ear, respectively
  • the determining the spatial noise source corresponding to the at least one subband comprises: acquiring a user head function, the user head function reflecting the reflection or absorption of the sound by the user's head; and on the at least one subband, By combining the ambient noise picked up by the first sub-microphone array, the ambient noise picked up by the second sub-microphone array, and the user head function, the corresponding spatial noise source is determined.
  • the determining one or more sources of spatial noise related to the picked-up ambient noise includes locating the One or more spatial noise sources.
  • the first microphone array includes one noise microphone
  • the at least one speaker array forms at least one set of acoustic dipoles
  • the noise microphone is located at an acoustic zero of the dipole radiated sound field.
  • the at least one microphone array includes a bone conduction microphone configured to pick up a user's speech
  • the signal processor estimates noise at the first spatial location based on the picked up ambient noise
  • the method includes: removing a component associated with the signal picked up by the bone conduction microphone from the picked-up environmental noise to update the environmental noise; and estimating noise at a first spatial location according to the updated environmental noise.
  • the at least one microphone array includes a bone conduction microphone and an air conduction microphone
  • the signal processor controls switch states of the bone conduction microphone and the air conduction microphone based on the working state of the earphone.
  • the state of the earphone includes a call state and a non-call state, and if the working state of the earphone is a non-call state, the signal processor controls the bone conduction microphone to be in a standby state; and if all If the working state of the earphone is the talking state, the signal processor controls the bone conduction microphone to be in the working state.
  • the signal processor controls the bone conduction microphone to maintain the working state;
  • the signal processor controls the bone conduction microphone to switch from the working state to the standby state.
  • FIG. 1 is an exemplary frame structure diagram of an open-back earphone provided according to some embodiments of the present application
  • FIG. 2 is a flow chart of an exemplary principle of an open-back earphone provided according to some embodiments of the present application
  • FIG. 3 is an exemplary flowchart of updating a noise reduction signal according to some embodiments of the present application.
  • FIG. 4A is an exemplary distribution diagram of the arrangement and positional relationship of the first microphone array and the second microphone array provided according to some embodiments of the present specification
  • 4B is an exemplary distribution diagram of the arrangement of the first microphone array and the second microphone array provided according to other embodiments of the present specification;
  • FIG. 5 is an exemplary flowchart of estimating noise at a first spatial location provided according to some embodiments of the present specification
  • FIG. 6 is an exemplary flowchart of determining a spatial noise source according to some embodiments of the present specification
  • FIG. 7 is another exemplary flowchart of determining a spatial noise source according to some embodiments of the present specification.
  • FIG. 8A is a schematic diagram of an arrangement of a first sub-microphone array provided according to some embodiments of the present specification.
  • FIG. 8B is a schematic diagram of an arrangement of the first sub-microphone array provided according to other embodiments of the present specification.
  • 8C is a schematic diagram of an arrangement of the first sub-microphone array provided according to other embodiments of the present specification.
  • 8D is a schematic diagram of an arrangement of the first sub-microphone array provided according to other embodiments of the present specification.
  • 9A is a schematic diagram of a positional relationship between a first sub-microphone array and a second sub-microphone array provided according to some embodiments of the present specification.
  • 9B is a schematic diagram of the positional relationship between the first sub-microphone array and the second sub-microphone array provided according to other embodiments of the present specification.
  • FIG. 10 is an exemplary flowchart of estimating noise at a first spatial location provided according to some embodiments of the present specification.
  • system means for distinguishing different components, elements, parts, parts or assemblies at different levels.
  • device means for converting signals into signals.
  • unit means for converting signals into signals.
  • module means for converting signals into signals.
  • An open-back headphone is a headphone device that opens the user's ear.
  • the open-back earphone can fix the speaker near the user's ear and not block the user's ear canal through a fixing structure (eg, an ear-hook, a head-hook, etc.).
  • a fixing structure eg, an ear-hook, a head-hook, etc.
  • the external environment noise can also be heard by the user, which makes the user's listening experience poor.
  • ambient noise can interfere with the user's music listening experience.
  • the microphone not only picks up the user's own speaking voice, but also picks up ambient noise, which makes the user's call experience poor.
  • the open earphone may include a fixed structure, a housing structure, a first microphone array, at least one speaker array, and a signal processor.
  • the fixing structure is configured to fix the open earphone near the user's ear without blocking the user's ear canal.
  • the housing structure is configured to house a first microphone array, at least one speaker array, and a signal processor.
  • the first microphone array is configured to pick up ambient noise.
  • the signal processor is configured to estimate noise at a first spatial location based on ambient noise picked up by the first microphone array, and generate a noise reduction signal based on the noise at the first spatial location, where the first spatial location is greater than any one of the first microphone arrays
  • the microphone is placed closer to the user's ear canal. It can be understood here that the microphones in the first microphone array may be distributed at different positions near the user's ear canal, and the position close to the user's ear canal is estimated by collecting ambient noise signals from the microphones in the first microphone array (for example, the noise at the first spatial location).
  • the at least one speaker array is configured to output noise-reducing waves based on the noise-reducing signal, which can be used to cancel ambient noise delivered to the user's ear canal.
  • the open-back earphone may further include a second microphone array configured to pick up ambient noise and noise-reducing waves output by the at least one speaker array.
  • the signal processor may update the noise reduction signal based on the sound signal picked up by the second microphone array. For example, the signal processor can estimate the sound field at the user's ear canal based on the sound signal picked up by the second microphone array, and adjust the phase or amplitude of the noise reduction signal according to the sound field at the user's ear canal to update the noise reduction signal.
  • the noise reduction wave is used to eliminate the environmental noise at the user's ear canal through the above method, so as to realize the active noise reduction of the open-type earphone, and improve the user's hearing experience in the process of using the open-type earphone.
  • FIG. 1 is an exemplary frame structure diagram of an open-back earphone provided according to some embodiments of the present application.
  • the open earphone 100 may include a housing structure 110, a first microphone array 130, a signal processor 140 and a speaker array 150, wherein the first microphone array 130, the signal processor 140 and the speaker array 150 are located in the housing body structure 110.
  • the open earphone 100 can fix the earphone near the user's ear through the fixing structure 120 without blocking the user's ear canal.
  • the first microphone array 130 located at the housing structure 110 can pick up the ambient noise at the user's ear canal, and convert the picked-up ambient noise signal into an electrical signal and transmit it to the signal processor 140 for signal processing.
  • the signal processor 140 is coupled to the first microphone array 130 and the speaker array 150.
  • the signal processor 140 receives the ambient noise signal picked up by the first microphone array 130 and performs signal processing on it to obtain parameter information (eg, amplitude information) of the ambient noise. , phase information, etc.).
  • the signal processor 140 may estimate the noise at the first spatial location of the user based on parameter information (eg, amplitude information, phase information, etc.) of the ambient noise, and generate a noise reduction signal based on the noise at the first spatial location.
  • the parameter information of the noise reduction signal corresponds to the parameter information of the environmental noise. For example, the amplitude of the noise reduction signal is approximately equal to that of the environmental noise, and the phase of the noise reduction signal is approximately opposite to that of the environmental noise.
  • the signal processor 140 delivers the generated noise reduction signal to the speaker array 150 .
  • the speaker array 150 can output a noise reduction wave according to the noise reduction signal generated by the signal processor 140, and the noise reduction wave can cancel each other with the ambient noise at the position of the user's ear canal, so as to realize the active noise reduction of the open earphone 100 and improve the user's ability to avoid noise.
  • the listening experience while using the open-back headphones 100 .
  • the housing structure 110 may be configured to carry the first microphone array 130 , the signal processor 140 and the speaker array 150 .
  • the housing structure 110 may be a closed or semi-closed housing structure with a hollow interior, and the first microphone array 130 , the signal processor 140 and the speaker array 150 are located at the housing structure 110 .
  • the shape of the housing structure 110 may be a three-dimensional structure with a regular or irregular shape such as a rectangular parallelepiped, a cylinder, and a truncated cone.
  • the housing structure 110 may be located close to the user's ear, for example, the housing structure 110 may be located on the peripheral side (eg, front or back) of the user's pinna, or on the user's ear on but not blocking or covering the user's ear canal.
  • the open earphone 100 may be a bone conduction earphone, and at least one side of the housing structure 110 may be in contact with the skin of the user's head.
  • Acoustic drivers eg, vibrating speakers
  • bone conduction headphones convert audio signals into mechanical vibrations that can be transmitted to the user's auditory nerves through the housing structure 110 and the user's bones.
  • the open earphone 100 may be an air conduction earphone, and at least one side of the housing structure 110 may or may not be in contact with the skin of the user's head.
  • the side wall of the housing structure 110 includes at least one sound guide hole, and the speaker in the air conduction earphone converts the audio signal into air conduction sound, and the air conduction sound can be radiated toward the user's ear through the sound guide hole.
  • the first microphone array 130 may be configured to pick up ambient noise.
  • ambient noise refers to a combination of multiple external sounds in the environment in which the user is located.
  • ambient noise may include one or more of traffic noise, industrial noise, building construction noise, social noise, and the like.
  • traffic noise may include, but is not limited to, motor vehicle driving noise, whistle noise, and the like.
  • Industrial noise may include, but is not limited to, factory power machinery operating noise, and the like.
  • Building construction noise may include, but is not limited to, power machinery excavation noise, hole drilling noise, stirring noise, and the like.
  • Social living environment noise may include, but is not limited to, crowd assembly noise, entertainment and publicity noise, crowd noise, household appliance noise, and the like.
  • the first microphone array 130 may be disposed near the user's ear canal for picking up ambient noise transmitted to the user's ear canal, the first microphone array 130 may convert the picked-up ambient noise signal into an electrical signal and It is passed to the signal processor 140 for signal processing.
  • the ambient noise may also include the sound of the user speaking.
  • the open earphone 100 is not in a call state, the sound produced by the user's own speech can also be regarded as environmental noise, and the first microphone array 130 can pick up the user's own speaking sound and other environmental noises, and convert the sound produced by the user's speech. Signals and other environmental noises are converted into electrical signals and passed to the signal processor 140 for signal processing.
  • the first microphone array 130 may be distributed at the user's left or right ear. In some embodiments, the first microphone array 130 may also be located at the user's left and right ears.
  • the first microphone array 130 may include a first sub-microphone array and a second sub-microphone array, wherein the first sub-microphone array is located at the user's left ear, the second sub-microphone array is located at the user's right ear, and the first sub-microphone array is located at the user's right ear.
  • the microphone array and the second sub-microphone array may enter the working state at the same time or one of the two may enter the working state.
  • the first microphone array 130 may include air conduction microphones and/or bone conduction microphones.
  • the first microphone array 130 may include one or more air conduction microphones.
  • the air conduction microphone can simultaneously acquire the noise of the external environment and the voice of the user when speaking, convert them into electrical signals as ambient noise and transmit them to the signal processor 140 for processing.
  • the first microphone array 130 may also include one or more bone conduction microphones.
  • the bone conduction microphone can be in direct contact with the skin of the user's head, and the vibration signals generated by the facial bones or muscles of the user when the user speaks can be directly transmitted to the bone conduction microphone, and then the bone conduction microphone converts the vibration signals into electrical signals, The electrical signal is transmitted to the signal processor 140 for signal processing.
  • the bone conduction microphone may not be in direct contact with the human body.
  • the vibration signal generated by the facial bones or muscles can be transmitted to the shell structure 110 first, and then transmitted to the bone conduction microphone by the shell structure 110.
  • the conductive microphone further converts the human body vibration signal into an electrical signal containing voice information.
  • the signal processor 140 can perform noise reduction processing on the sound signal collected by the air conduction microphone as environmental noise, and the sound signal collected by the bone conduction microphone can be retained as a voice signal, so as to ensure the call quality of the user during the call .
  • the first microphone array 130 may include a dynamic microphone, a ribbon microphone, a condenser microphone, an electret microphone, an electromagnetic microphone, a carbon particle microphone, etc., according to the working principle of the microphone as a classification, or any combination thereof.
  • the array arrangement of the first microphone array 130 may be a linear array (eg, a straight line, a curve), a planar array (eg, a cross, a circle, a ring, a polygon, a mesh, etc.) and/or irregular shape) or a three-dimensional array (eg, cylindrical, spherical, hemispherical, polyhedron, etc.), for the arrangement of the first microphone array 130, reference may be made to FIG. 8 and related contents of this specification.
  • the signal processor 140 is configured to estimate the noise at the first spatial location based on the ambient noise picked up by the first microphone array 130, and to generate a noise reduction signal based on the noise at the first spatial location.
  • the first spatial position refers to a spatial position close to the user's ear canal at a specific distance, and the first spatial position is closer to the user's ear canal than any microphone in the first microphone array 130 .
  • the specific distance here may be a fixed distance, for example, 0.5 cm, 1 cm, 2 cm, 3 cm, and the like.
  • the first spatial position is related to the distribution position and quantity of each microphone in the first microphone array 130 relative to the user's ear.
  • the quantity can be adjusted for the first spatial position. For example, by increasing the number of microphones in the first microphone array 130, the first spatial position can be made closer to the user's ear canal.
  • the signal processor 140 may perform signal processing on the received environmental noise signal to estimate the noise at the first spatial position.
  • a signal processor 140 may be coupled to the first microphone array 130 and the speaker array 150, and the signal processor 140 may receive ambient noise picked up by the first microphone array 130 to estimate noise at the first spatial location.
  • the signal processor 140 may determine one or more sources of spatial noise related to the picked-up ambient noise.
  • the signal processor 140 may perform azimuth estimation, phase information estimation, amplitude information estimation, and the like for spatial noise sources related to environmental noise.
  • the signal processor 140 may generate a noise reduction signal based on the noise estimate (eg, phase information, amplitude information) of the first spatial location.
  • the noise reduction signal refers to a sound signal that is approximately equal in magnitude and approximately opposite in phase to the noise at the first spatial position.
  • the speaker array 150 is configured to output a noise reduction wave based on the noise reduction signal, the noise reduction wave being used to cancel the ambient noise delivered to the ear canal of the user.
  • the speaker array 150 may be disposed at the housing structure 110, and when the user wears the open earphone 100, the speaker array 150 may be located near the user's ear.
  • the speaker array 150 may output a noise reduction wave based on the noise reduction signal to cancel out the ambient noise at the first spatial location.
  • the signal processor 140 controls the speaker array 150 to output a sound signal with approximately the same amplitude and approximately opposite phase as the noise at the first spatial position to cancel the noise at the first spatial position.
  • the distance between the first spatial position and the user's ear canal is relatively small, and the noise at the first spatial position can be approximately regarded as the noise transmitted to the user's ear.
  • the noise reduction output of the speaker array 150 is based on the noise reduction signal.
  • the sound waves can cancel each other out with the noise at the first spatial position, and it can be approximated that the ambient noise transmitted to the user's ear canal is eliminated.
  • the speaker array 150 may include electrodynamic speakers (eg, moving coil speakers), magnetic speakers, ionic speakers, electrostatic speakers (or condenser speakers), One or more of the speakers, etc.
  • the speaker array 150 may include air-conduction speakers or bone-conduction speakers, classified according to how the sound output by the speakers propagates.
  • the speaker array 150 when the speaker array 150 only includes air conduction speakers, part of the air conduction speakers in the speaker array 150 can be used to output noise reduction waves to eliminate noise, and other air conduction speakers in the speaker array 150 can be used to convey to the user what the user needs to hear audio information (for example, device media audio, call far-end audio).
  • the loudspeakers in the loudspeaker array 150 used to convey the sound information that the user needs to hear to the user may also be used to output noise reduction waves.
  • the speaker array 150 when the speaker array 150 includes bone conduction speakers and air conduction speakers, the air conduction speakers can be used to output noise reduction waves to eliminate noise, and the bone conduction speakers can be used to transmit the sound information that the user needs to hear to the user.
  • bone conduction speakers transmit mechanical vibrations directly through the user's body (eg, bone, skin tissue, etc.) to the user's auditory nerves with less interference with the air-conduction microphones that pick up ambient noise.
  • the bone conduction speaker will cause the casing structure 110 to generate mechanical vibration.
  • the mechanical vibration generated by the casing structure 110 acts on the air to generate air conduction sound.
  • the casing structure 110 generates The air conduction sound can also act as a noise reduction wave.
  • the speaker array 150 may be an independent functional device, or may be part of a single device capable of implementing multiple functions.
  • the signal processor 140 may be integrated and/or formed integrally with the speaker array 150 .
  • the arrangement of the speaker array 150 may be a linear array (eg, a straight line, a curve), a planar array (eg, a cross, a mesh, a circle, a ring, a polygon, etc., regular and/or irregular shape) or a three-dimensional array (eg, cylindrical, spherical, hemispherical, polyhedral, etc.), which are not limited herein.
  • the open-back earphone 100 may further include a securing structure 120 configured to secure the open-back earphone 100 in a position near the user's ear without blocking the user's ear canal.
  • the fixing structure 120 may include ear hooks, head beams, or elastic bands, etc., so that the open-back earphone 100 can be better fixed near the user's ear and prevent the user from falling during use.
  • the securing structure 120 may be an earhook, which may be configured to be worn around the ear area.
  • the securing structure 120 may be a neckband configured to be worn around the neck/shoulder area.
  • the earhook can be a continuous hook and can be elastically stretched to fit on the user's ear, while the earhook can also apply pressure to the user's pinna, making the open-back earphone 100 secure fixed to a specific location on the user's ear or head.
  • the earhook may be a discontinuous band.
  • the earhook may include a rigid portion and a flexible portion, wherein the rigid portion may be made of a rigid material (eg, plastic or metal), and the rigid portion may be physically connected (eg, snap-fit, threaded connection, etc.)
  • the flexible portion may be made of an elastic material (eg, cloth, composite or/and neoprene).
  • FIGS. 1 and 2 are provided for illustrative purposes only and are not intended to limit the scope of the present application. Numerous changes and modifications will occur to those of ordinary skill in the art in light of the teachings of this disclosure. However, such changes and modifications do not depart from the scope of this application.
  • one or more elements of open-back earphone 100 eg, securing structure 120, etc.
  • one element may be replaced by other elements that perform similar functions.
  • the open-back earphone 100 may not include the fixed structure 120, and the housing structure 110 may be a housing structure having a shape adapted to the human ear, such as a circular ring, an oval, a polygon (regular or irregular) ), U-shaped, V-shaped, semi-circular, so that the housing structure 110 can hang near the user's ear.
  • an element may be split into multiple sub-elements, or multiple elements may be combined into a single element.
  • FIG. 2 is an exemplary schematic flow chart of an open-back earphone provided according to some embodiments of the present application. As shown in FIG. 2, the process 200 may include:
  • step 210 ambient noise is picked up.
  • ambient noise refers to a combination of multiple external sounds in the environment in which the user is located.
  • ambient noise may include one or more of traffic noise, industrial noise, building construction noise, social noise, and the like.
  • traffic noise may include, but is not limited to, motor vehicle driving noise, whistle noise, and the like.
  • Industrial noise may include, but is not limited to, factory power machinery operating noise, and the like.
  • Building construction noise may include, but is not limited to, power machinery excavation noise, hole drilling noise, stirring noise, and the like.
  • Social living environment noise may include, but is not limited to, crowd assembly noise, entertainment and publicity noise, crowd noise, household appliance noise, and the like.
  • the first microphone array 130 may be located near the user's ear canal for picking up ambient noise transmitted to the user's ear canal, the first microphone array 130 may convert the picked-up ambient noise signal into an electrical signal and It is passed to the signal processor 140 for signal processing.
  • the ambient noise may also include the sound of the user speaking.
  • the open earphone 100 is not in a call state (for example, when listening to audio or watching a video)
  • the sound produced by the user's own speech can also be regarded as environmental noise
  • the first microphone array 130 can pick up the user's own speech and other environmental noises. noise, and convert the sound signal and other environmental noise generated by the user's speech into electrical signals and transmit them to the signal processor 140 for signal processing.
  • step 220 the noise at the first spatial location is estimated based on the picked-up ambient noise.
  • the first spatial position refers to a spatial position close to the user's ear canal by a specific distance.
  • the specific distance here can be a fixed distance, for example, 0.5cm, 1cm, 2cm, 3cm, etc., which can be adaptively adjusted according to actual application conditions.
  • the ambient noise picked up by the first microphone array 130 may come from different azimuths and different types of spatial noise sources, so the parameter information (eg, phase information, amplitude information) corresponding to each spatial noise source is different.
  • the signal processor 140 may perform signal separation and extraction on the noise at the first spatial position according to the statistical distribution and structural features of different types of noise in different dimensions (eg, spatial domain, time domain, frequency domain, etc.), Thereby, different types of noise (eg, different frequencies, different phases, etc.) are estimated, and parameter information (eg, amplitude information, phase information, etc.) corresponding to each noise is estimated.
  • the signal processor 140 may further determine overall parameter information of the noise at the first spatial position according to parameter information corresponding to different types of noise at the first spatial position.
  • estimating noise at the first spatial location based on the picked-up ambient noise may further include determining one or more spatial noise sources associated with the picked-up ambient noise, estimating noise at the first spatial location based on the spatial noise sources.
  • the picked-up environmental noise is divided into a plurality of subbands, each subband corresponds to a different frequency range, and on at least one subband, the corresponding spatial noise source is determined.
  • the spatial noise source estimated by the subbands here is a virtual noise source corresponding to the external real noise source.
  • the open earphone 100 does not block the user's ear canal, and cannot obtain ambient noise by arranging a microphone at the ear canal.
  • the first spatial position is a spatial area constructed by the first microphone array 130 for simulating the position of the user's ear canal.
  • the first spatial location is closer to the user's ear canal than any microphone in the first microphone array 130 .
  • the first spatial position is related to the distribution position and quantity of each microphone in the first microphone array 130 relative to the user's ear, and by adjusting the distribution position or quantity of each microphone in the first microphone array 130 relative to the user's ear Adjust the first spatial position.
  • the first spatial position can be made closer to the user's ear canal.
  • the first spatial position can also be made closer to the user's ear canal by reducing the distance between the microphones in the first microphone array 130 .
  • the arrangement of the microphones in the first microphone array 130 can also be changed to make the first spatial position closer to the user's ear canal.
  • a noise reduction signal is generated based on the noise at the first spatial location.
  • this step may be performed by signal processor 140 .
  • the signal processor 140 may generate a noise reduction signal based on the parameter information (eg, amplitude information, phase information, etc.) of the noise at the first spatial location obtained in step 220 .
  • the phase of the noise reduction wave may be approximately opposite to the phase of the noise at the first spatial location.
  • the phase of the noise reduction wave may be approximately opposite to that of the noise at the first spatial position, and the magnitude of the noise reduction signal may be approximately equal to the magnitude of the noise at the first spatial position.
  • the speaker array 150 may output a noise reduction wave based on the noise reduction signal generated by the signal processor 140, and the noise reduction wave may cancel each other out with the noise at the first spatial position.
  • the noise at the first spatial position can be approximately regarded as the noise at the user's ear canal. Therefore, the noise reduction signal and the noise at the first spatial position cancel each other out, which can be approximated as the ambient noise transmitted to the user's ear canal is eliminate.
  • the open-back earphone 100 can eliminate the ambient noise at the position of the user's ear canal through the method steps described in FIG. 2 , so as to realize the active noise reduction of the open-back earphone 100 .
  • the open-back earphone 100 may also include a second microphone array 160 .
  • the second microphone array 160 may be located inside the housing structure 110 .
  • the second microphone array 160 is at least partially distinct from the first microphone array 130 .
  • the microphones in the second microphone array 160 are different from the microphones in the first microphone array 130 in one or more of the number, type, location, arrangement, and the like.
  • the arrangement of microphones in the first microphone array 130 may be linear, and the arrangement of microphones in the second microphone array 160 may be circular.
  • the microphones in the second microphone array 160 may only include air conduction microphones, and the first microphone array 130 may include air conduction microphones and bone conduction microphones.
  • the microphones in the second microphone array 160 may be any one or more microphones included in the first microphone array 130 , and the microphones in the second microphone array 160 may also be independent of the microphones in the first microphone array 130 .
  • the second microphone array 160 may be configured to pick up ambient noise and noise reduction waves output by the speaker array 150 .
  • the ambient noise and noise reduction waves picked up by the second microphone array 160 may be transferred to the signal processor 140 .
  • the signal processor 140 may update the noise reduction signal based on the sound signal picked up by the second microphone array 160 .
  • the signal processor 140 may adjust the parameter information of the noise reduction signal according to the parameter information (eg, frequency information, amplitude information, phase information, etc.) of the sound signal picked up by the second microphone array 160, so that the adjusted noise reduction signal has a
  • the amplitude can be more consistent with the amplitude of the noise at the first spatial position, or the phase of the adjusted noise reduction signal can be more consistent with the inverse phase of the phase of the noise at the first spatial position, so that the updated noise reduction wave
  • the noise at the first spatial position can be canceled more comprehensively.
  • FIG. 3 is an exemplary flowchart of updating a noise reduction signal according to some embodiments of the present application. As shown in FIG. 3, the process 300 may include:
  • step 310 the sound field at the user's ear canal is estimated based on the sound signal picked up by the second microphone array 160.
  • this step may be performed by signal processor 140 .
  • the sound signal picked up by the second microphone array 160 includes ambient noise and noise reduction waves output by the speaker array 150 .
  • the ambient noise and the noise reduction wave output by the speaker array 150 may still be some uncancelled sound signals near the user's ear canal, and these uncancelled sound signals may be the residual environment Noise and/or residual noise reduction waves, so there is still a certain amount of noise in the user's ear canal after the ambient noise and the noise reduction waves are canceled.
  • the signal processor 140 may perform signal processing according to the sound signal (eg, ambient noise, noise reduction wave) picked up by the second microphone array 160 to obtain parameter information of the sound field at the user's ear canal, such as frequency information, amplitude information and phase information, etc., so as to realize the estimation of the sound field at the user's ear canal.
  • the sound signal eg, ambient noise, noise reduction wave
  • step 320 the parameter information of the noise reduction signal is adjusted according to the sound field at the user's ear canal.
  • step 320 may be performed by signal processor 140 .
  • the signal processor 140 may adjust the parameter information (eg, frequency information, amplitude information and/or phase information) of the noise reduction signal according to the parameter information of the sound field at the user's ear canal obtained in step 310,
  • the amplitude information and frequency information of the updated noise reduction signal are more consistent with the amplitude information and frequency information of the environmental noise at the user's ear canal, and the phase information of the updated noise reduction signal is inverse to the environmental noise at the user's ear canal.
  • the phase information is more consistent, so that the updated noise reduction signal can eliminate ambient noise more accurately.
  • the microphone array that picks up the sound field at the user's ear canal is not limited to the second microphone array, but may also include other microphone arrays, such as a third microphone array, a fourth microphone array, etc.
  • the relevant parameter information of the sound field of the user can be used to estimate the sound field at the user's ear canal by means of averaging or weighting algorithms.
  • the second microphone array 160 in order to obtain the sound field at the user's ear canal more accurately, includes a microphone that is closer to the user's ear canal than any microphone in the first microphone array 130 .
  • the sound signal picked up by the first microphone array 130 is ambient noise
  • the sound signal picked up by the second microphone array 160 is ambient noise and noise reduction waves.
  • the signal processor 140 may estimate the sound field at the user's ear canal according to the sound signal picked up by the second microphone array 160 to update the noise reduction signal.
  • the second microphone array 160 needs to monitor the sound field at the user's ear canal after the noise reduction signal and the ambient noise are cancelled.
  • the second microphone array 160 includes a microphone that is closer to the user's ear canal than any microphone in the first microphone array 130, which can be more accurate. Characterizing the sound signal heard by the user, the sound field of the second microphone array 160 is estimated to update the noise reduction signal, which can further improve the noise reduction effect and the user's sense of hearing experience.
  • the arrangement of the first microphone array 130 and the second microphone array 160 may be the same.
  • the arrangement of the first microphone array 130 and the second microphone array 160 in the same manner can be understood as the arrangement shapes of the two are approximately the same.
  • FIG. 4A is an exemplary distribution diagram of the arrangement and positional relationship of the first microphone array and the second microphone array provided according to some embodiments of the present specification. As shown in FIG. 4A , the first microphone array 130 is arranged at the human ear in a semicircular arrangement, and the second microphone array 160 is also arranged at the human ear in a semicircular arrangement. The microphones in the second microphone array 160 are closer to the user's ear canal than any microphone in the first microphone array 130 .
  • the microphones in the first microphone array 130 may be arranged independently of the microphones in the second microphone array 160 .
  • the microphones in the first microphone array 130 in FIG. 4A are arranged in a semicircular arrangement
  • the microphones in the second microphone array 160 are arranged in a semicircular arrangement
  • the first microphone array 160 is arranged in a semicircular arrangement.
  • the microphones in 130 do not overlap or intersect with the microphones in the second microphone array 160 .
  • the microphones in the first microphone array 130 may partially overlap or intersect with the microphones in the second microphone array 160 .
  • the arrangement of the first microphone array 130 and the second microphone array 160 may be different.
  • FIG. 4B is an exemplary distribution diagram of the arrangement of the first microphone array and the second microphone array provided according to other embodiments of the present specification. As shown in FIG. 4B , the first microphone array 130 is arranged at the human ear in a semicircular arrangement, and the second microphone array 160 is arranged at the human ear in a linear arrangement. The microphones in the second microphone array 160 are closer to the user's ear canal than any microphone in the first microphone array 130 . In some embodiments, the first microphone array 130 and the second microphone array 160 may also be arranged in a combined manner. For example, the second microphone array 160 in FIG.
  • the arrangement of the first microphone array 130 and the second microphone array 160 is not limited to the semicircle and the line shown in FIG. 4A and FIG. 4B , and the semicircle and line here are only for the purpose of illustration , for the arrangement of the microphone array, please refer to FIG. 8 and related descriptions in this specification.
  • the noise reduction signal may also be updated based on manual user input.
  • the user's own listening experience effect is not ideal, and the user can manually adjust the parameter information of the noise reduction signal according to the user's own hearing effect (eg, frequency information, phase information, or amplitude information).
  • the hearing ability of the special user is different from that of the ordinary user, and the hearing ability generated by the open-back earphone 100 itself is different. Noise waves do not match the hearing ability of special populations, resulting in poor hearing experience for special users.
  • the special user can manually adjust the frequency information, phase information or amplitude information of the noise reduction signal according to his own hearing effect, so as to update the noise reduction signal to improve the hearing experience of the special user.
  • the way for the user to manually adjust the noise reduction signal may be manual adjustment through the keys on the open-back earphone 100 .
  • the way for the user to manually adjust the noise reduction signal may also be manual input adjustment through a terminal device.
  • the open earphone 100 or electronic products such as mobile phones, tablet computers, computers and the like communicated with the open earphone 100 may display the sound field at the user's ear canal, and feedback the frequency information of the noise reduction signal suggested to the user. The user can manually input the range, amplitude information range or phase information range according to the parameter information of the suggested noise reduction signal, and then fine-tune the parameter information according to their own listening experience.
  • estimating the noise in the first space based on the picked-up environmental noise by the signal processor 140 may include: performing signal separation according to the picked-up environmental noise, acquiring parameter information corresponding to the environmental noise, and generating a noise reduction signal based on the parameter information corresponding to the environmental noise. noise signal.
  • ambient noise picked up by the microphone array eg, the first microphone array 130, the second microphone array 160
  • the audio output by the speaker array 150 may include the far end audio of the call output by the speaker array 150, device media audio, noise reduction waves, and the like.
  • the signal processor 140 may perform signal analysis on the environmental noise picked up by the microphone array, and perform signal separation of various sound signals included in the environmental noise to obtain noise, user vocals, noise reduction waves, and device media audio. , call remote audio and other single sound signals.
  • the signal processor 140 can adaptively adapt to the statistical distribution characteristics and structural characteristics of noise, user vocals, noise reduction waves, device media audio, and call far-end audio in different dimensions such as space, time domain, frequency domain, etc. Adjust the parameters of the filter bank, estimate the parameter information of each sound signal in the environmental noise (for example, noise, user voice, noise reduction wave, device media audio, call far-end audio, etc.), and complete the signal separation process according to different parameter information .
  • the microphone array may convert the picked-up noise, the user's voice, and the noise reduction wave into corresponding first signals, second signals, and third signals, respectively.
  • the signal processor 140 obtains the first signal, the second signal, and the third signal in the spatial difference (eg, where the signals are located), the time domain difference (eg, delay), the frequency domain difference (eg, amplitude, phase), and The first signal, the second signal, and the third signal are separated according to the differences in the three dimensions to obtain relatively pure first signal, second signal, and third signal.
  • the separated first signal, second signal, and third signal correspond to pure noise, user voice, and noise-reduced wave, respectively, and the signal processor 140 completes the signal separation process.
  • the signal processor 140 may update the noise reduction wave according to the parameter information of noise, noise reduction wave, device media audio, and call far-end audio obtained by signal separation, and the updated noise reduction wave is output through the speaker array 150 .
  • the structured features of noise may include noise distribution, noise intensity, global noise intensity, noise rate, etc., or any combination thereof.
  • the noise intensity may refer to the value of the noise pixel, reflecting the magnitude of noise in the noise pixel, and thus, the noise distribution may reflect the probability density of noise with different noise intensities in the image.
  • the global noise intensity can reflect the average noise intensity or weighted average noise intensity in the image.
  • the noise rate can reflect the dispersion of the noise distribution.
  • the statistical distribution characteristics of noise may include, but are not limited to, probability distribution density, power spectral density, autocorrelation function, probability density function, variance, mathematical expectation, and the like.
  • user vocals, device media audio, audio from the far end of the call, etc. obtained through signal separation can also be transmitted to the far end of the call.
  • the user's voice can be transmitted to the far end of the call.
  • FIG. 5 is an exemplary flowchart of estimating noise at a first spatial location provided according to some embodiments of the present specification. As shown in FIG. 5, the process 500 may include:
  • step 510 one or more spatial noise sources associated with the picked-up ambient noise are determined.
  • a spatial noise source related to ambient noise refers to a noise source whose sound waves can be transmitted to or near the user's ear canal (eg, a first spatial location).
  • the spatial noise sources may be noise sources in different directions (eg, front, rear, etc.) of the user's body. For example, there is crowd noise in front of the user's body, and there is vehicle whistle noise to the left of the user's body. In this case, the spatial noise sources are the crowd noise source in front of the user's body and the vehicle whistle noise source to the left of the user's body.
  • the first microphone array 130 can pick up spatial noises in all directions of the user's body, convert the spatial noises into electrical signals and transmit them to the signal processor 140 , and the signal processor 140 can convert the electrical signals corresponding to the spatial noises into electrical signals.
  • parameter information eg, azimuth information, amplitude information, phase information, etc.
  • the signal processor 140 determines the spatial noise sources in various directions according to the parameter information of the spatial noise in various directions, for example, the orientation of the spatial noise source, the phase of the spatial noise source, and the amplitude of the spatial noise source.
  • the signal processor 140 may determine the source of spatial noise through a noise localization algorithm.
  • noise localization algorithms may include one or more of beamforming, super-resolution spatial spectrum estimation, time difference of arrival, and the like.
  • the signal processor 140 may divide the picked-up ambient noise into a plurality of sub-bands according to a specific frequency bandwidth (for example, every 500 Hz as a frequency band), each sub-band may correspond to a different frequency range, and at least A spatial noise source corresponding to the subband is determined on a subband.
  • a specific location method of the spatial noise source reference may be made to other places in this specification, and details are not described here.
  • FIG. 6 For a detailed description of determining one or more spatial noise sources related to the picked-up ambient noise, reference may be made to FIG. 6 and related descriptions of this specification.
  • step 520 based on the spatial noise sources, the noise at the first spatial location is estimated.
  • this step may be performed by signal processor 140 .
  • the signal processor 140 may estimate the respective spatial noise sources based on the parameter information (eg, frequency information, amplitude information, phase information, etc.) of the spatial noise sources located in various directions of the user's body obtained in step 510 .
  • the parameter information of the noise at the first spatial location is passed to estimate the noise at the first spatial location.
  • the frequency information, phase information or amplitude information of the spatial noise source in front of the spatial location may be performed by signal processor 140 .
  • the signal processor 140 may estimate the respective spatial noise sources based on the parameter information (eg, frequency information, amplitude information, phase information, etc.) of the spatial noise sources located in various directions of the user's body obtained in step 510 .
  • the signal processor 140 estimates the frequency, phase or amplitude information of the rear spatial noise source when the rear spatial noise source is transmitted to the first spatial position according to the frequency information, phase information or amplitude information of the rear spatial noise source.
  • the signal processor 140 estimates the noise information of the first spatial position based on the frequency information, phase information or amplitude information of the front spatial noise source and the frequency information, phase information or amplitude information of the rear spatial noise source, thereby estimating the first spatial position noise.
  • the parameter information of the spatial noise source can be extracted from the frequency response curve of the spatial noise source picked up by the microphone array through a feature extraction method.
  • methods for extracting parameter information of spatial noise sources may include, but are not limited to, Principal Components Analysis (PCA), Independent Component Algorithm (ICA), Linear Discriminant Analysis (Linear Discriminant Analysis, LDA), singular value decomposition (Singular Value Decomposition, SVD) and so on.
  • PCA Principal Components Analysis
  • ICA Independent Component Algorithm
  • LDA Linear Discriminant Analysis
  • SVD singular value decomposition
  • determining one or more sources of spatial noise associated with picked-up ambient noise may include locating the one or more spatial noise sources through one or more of beamforming, super-resolution spatial spectrum estimation, or time difference of arrival source.
  • the beamforming localization method is a sound source localization method based on the controllable beamforming of the maximum output power.
  • the beamforming sound source localization method may perform weighted summation of the sound signals picked up by each microphone element in the microphone array to form a beam, and guide the beam by searching for possible positions of the spatial noise source, and modify the weights such that The output signal power of the microphone array is maximum. It should be noted that the beamforming sound source localization method can be used in both the time domain and the frequency domain.
  • the sound source localization method of super-resolution spatial spectral estimation may include autoregressive AR model, minimum variance spectral estimation (MV), and eigenvalue decomposition method (eg, Music algorithm), etc., all of which can be obtained by acquiring microphone
  • the sound signal of the array is used to calculate the correlation matrix of the spatial spectrum and effectively estimate the direction of the spatial noise source.
  • the time-of-arrival sound source localization method may first estimate the time-of-arrival sound, obtain the sound delay (TDOA) between elements in the microphone array, and then use the obtained time-of-arrival sound source in combination with known microphone arrays.
  • the spatial location of further locates the location of the spatial noise source.
  • the spatial noise source may be a far-field sound source.
  • the incident sound waves incident on the microphone array from the spatial noise source are parallel.
  • the incident angle of the incident sound wave from the spatial noise source is perpendicular to the microphone plane in the microphone array (for example, the first microphone array 130 or the second microphone array 160 )
  • the incident sound wave can reach the microphone array (for example, the first microphone array 130 or the second microphone array 160 ) at the same time.
  • each microphone in a microphone array 130 or a second microphone array 160 may have a delay, which may be determined by the angle of incidence.
  • the intensity of the noise waveform after superposition is different. For example, when the incident angle is 0°, the noise signal strength is weak, and when the incident angle is 45°, the noise signal strength is the strongest.
  • the microphone array (eg, the first microphone array 130 or the second microphone array 160 ) may be a directional array, and the directivity of the directional array may be realized by a time domain algorithm or a frequency domain phase delay algorithm, for example, Delay, overlay, etc. In some embodiments, by controlling different delays, different directions of pointing can be achieved. In some embodiments, the directivity of the directional array is controllable, which is equivalent to a spatial filter.
  • the noise localization area is divided into grids, and then each microphone is delayed in the time domain according to the delay time of each grid point, and finally the The time-domain delays are superimposed, and the sound pressure of each grid is calculated to obtain the relative sound pressure of each grid, and finally the spatial noise source localization is realized.
  • the above description about the process 500 is only for example and illustration, and does not limit the scope of application of this specification.
  • the process 500 may further include locating the spatial noise source, extracting parameter information of the spatial noise source, and the like.
  • step 510 and step 520 may be combined into one step.
  • these corrections and changes are still within the scope of this specification.
  • FIG. 6 is an exemplary flowchart of determining a spatial noise source according to some embodiments of the present specification. As shown in FIG. 6, the process 600 may include:
  • step 610 the picked-up ambient noise is divided into a plurality of subbands, each subband corresponds to a different frequency range.
  • this step may be performed by signal processor 140 .
  • the frequencies of ambient noise from different directions of the user's body may be different.
  • the signal processor 140 performs signal processing on the ambient noise signal, the ambient noise frequency band may be divided into multiple sub-bands, each sub-band corresponding to different frequency ranges.
  • the frequency range corresponding to each subband here may be a preset frequency range, for example, 80Hz-100Hz, 100Hz-300Hz, 300Hz-800Hz, and so on.
  • each subband includes parameter information of the environmental noise of the corresponding frequency band.
  • the signal processor 140 may divide the picked-up ambient noise into four sub-bands of 80Hz-100Hz, 100Hz-300Hz, 300Hz-800Hz, 800Hz-1000Hz, and the four sub-bands correspond to 80Hz-100Hz, 100Hz-300Hz, 300Hz- 800Hz, 800Hz-1000Hz environmental noise parameters.
  • step 620 on at least one subband, a spatial noise source corresponding thereto is determined.
  • this step may be performed by signal processor 140 .
  • the signal processor 140 may perform signal analysis on the subbands into which the environmental noise is divided, obtain parameter information of the environmental noise corresponding to each subband, and determine a spatial noise source corresponding to each subband according to the parameter information. For example, in the subband of 300Hz-800Hz, the signal processor 140 can obtain the parameter information (eg, frequency information, amplitude information, phase information, etc.) The acquired parameter information determines the spatial noise source corresponding to the subband 300Hz-800Hz.
  • the parameter information eg, frequency information, amplitude information, phase information, etc.
  • FIG. 7 is another exemplary flowchart of determining a spatial noise source according to some embodiments of the present specification. As shown in FIG. 7, process 700 may include:
  • step 710 a user head function is obtained, and the user head function reflects the reflection or absorption of sound by the user's head.
  • the first microphone array 130 may include a first sub-microphone array and a second sub-microphone array, which are located at the left and right ears of the user, respectively.
  • the first microphone array 130 may be a bilateral mode arrangement, ie a bilateral mode arrangement in which the first sub-microphone array and the second sub-microphone array are simultaneously enabled.
  • the first sub-microphone array when the first sub-microphone array is located at the position of the user's left ear, and the second sub-microphone array is located at the position of the user's right ear, in a bilateral pattern arrangement, during the transmission of the sound signal, the user's head will perform the sound signal on the user's head.
  • the signal processor 140 may construct a user head function based on the difference between the parameter information of the ambient noise picked up by the first sub-microphone array and the parameter information of the same ambient noise picked up by the second sub-microphone array, which uses The head function can reflect the reflection and absorption of sound by the user's head.
  • step 720 on at least one subband, combining the ambient noise picked up by the first sub-microphone array, the ambient noise picked up by the second sub-microphone array, and the user head function, determine the corresponding spatial noise source.
  • this step may be performed by signal processor 140 .
  • the amplitude information and phase information of the ambient noise signal picked up by the first sub-microphone array and the amplitude information and phase information of the ambient noise signal picked up by the second sub-microphone array are combined. There are amplitude and phase differences between them.
  • the signal processor 140 may perform the processing on at least one sub-band of the ambient noise according to the ambient noise picked up by the first sub-microphone array, the ambient noise picked up by the second sub-microphone array, and the user head function acquired by the signal processor 140 in step 710.
  • Frequency point synthesis that is, taking the head function as prior information, on at least one subband of the ambient noise, the frequency points of the ambient noise on the corresponding subband picked up by the first sub-microphone array and the corresponding subband picked up by the second sub-microphone array
  • the frequency points of the ambient noise are synthesized.
  • the parameter information contained in the subbands after the frequency point synthesis is completed corresponds to the parameter information of the reconstructed virtual noise source.
  • the signal processor 140 determines the spatial noise source based on the parameter information of the reconstructed virtual sound source, and then completes the spatial noise source localization.
  • the first microphone array 130 may also be arranged in a unilateral mode. For example, only the first sub-microphone array or the second sub-microphone array is enabled. In some embodiments, the first microphone array 130 is arranged in a unilateral mode, and when the first sub-microphone array located at the user's left ear is enabled, the signal processor 140 can use the user's head function as prior information, when the ambient noise On at least one sub-band, the frequency points of the ambient noise on the corresponding sub-band picked up by the first sub-microphone array are synthesized. The parameter information contained in the subband after frequency point synthesis is completed corresponds to the reconstructed virtual noise source parameter information. The signal processor 140 determines the spatial noise source based on the parameter information of the reconstructed virtual sound source, and then completes the spatial noise source localization.
  • the first sub-microphone array can pick up the environmental noise reaching the user's left ear, and the signal processor 140 can also estimate the environmental noise when the environmental noise reaches the user's right ear through the user head function based on the environmental noise parameter information. Parameter information. The signal processor 140 can locate the spatial noise source more accurately according to the estimated parameter information when the ambient noise reaches the user's right ear.
  • the unilateral mode of the first microphone array 130 may also be to set only one sub-microphone array, and the spatial noise source localization process in such a unilateral mode is different from enabling only the first sub-microphone array (or the second sub-microphone array). The unilateral mode spatial noise source localization process of the sub-microphone array) is similar and will not be repeated here.
  • the arrangement of the first sub-microphone array or the second sub-microphone array may be an array of regular geometric shapes.
  • FIG. 8A is a schematic diagram of an arrangement of a first sub-microphone array provided according to some embodiments of the present specification. As shown in FIG. 8A , the first sub-microphone array forms a linear array. In some embodiments, the arrangement of the first sub-microphone array or the second sub-microphone array may also be an array of other shapes.
  • FIG. 8B shows the first sub-microphone array provided according to other embodiments of this specification. Schematic diagram of the arrangement. As shown in FIG. 8B , the first sub-microphone arrays form a cross-shaped array. For another example, FIG.
  • FIG. 8C is a schematic diagram of an arrangement of the first sub-microphone array provided according to other embodiments of the present specification. As shown in Fig. 8C, the first sub-microphone array is in a circular array. It should be noted that the arrangement of the first sub-microphone array or the second sub-microphone array is not limited to the linear array, cross-shaped array, and circular array shown in FIG. 8A , FIG. 8B , and FIG. 8C , but may also be other shapes.
  • the array pattern for example, a triangular array, a spiral array, a plane array, a three-dimensional array, etc., is not limited in this specification. It should be noted that each short solid line in FIGS.
  • each short solid line is a group of microphones
  • the number of each group of microphones may be the same or different
  • the types of each group of microphones may be the same or different
  • the orientation of each group of microphones may be the same or different.
  • the type, quantity, orientation and spacing can be adaptively adjusted according to the actual application.
  • the arrangement of the first sub-microphone array or the second sub-microphone array may also be an array of irregular geometry.
  • FIG. 8D is a schematic diagram of the arrangement of the first sub-microphone array provided according to other embodiments of the present specification. As shown in FIG. 8D , the first sub-microphones are arrayed in an irregular array. It should be noted that the irregular-shaped array arrangement of the first sub-microphone array or the second sub-microphone array is not limited to the shape shown in FIG. Regular polygons, etc., are not limited in this specification.
  • the microphones in the first sub-microphone array may be uniformly distributed, and the uniform distribution here refers to the The spacing between the microphones is the same.
  • the microphones in the first sub-microphone array may also be non-uniformly distributed, and the non-uniform distribution here refers to the first sub-microphone array (or the second sub-microphone array).
  • the spacing between the microphones is different.
  • the spacing between the microphone array elements in the sub-microphone array can be adaptively adjusted according to the actual situation, which is not limited in this specification.
  • FIG. 9A is a schematic diagram of the positional relationship between the first sub-microphone array and the second sub-microphone array provided according to some embodiments of the present specification.
  • the first sub-microphone array 911 is located at the left ear of the user, and the first sub-microphone array 911 is arranged in an approximately triangular shape.
  • the second sub-microphone array 912 is located at the right ear of the user.
  • the second sub-microphone array 912 is also arranged in an approximate triangle shape.
  • the second sub-microphone array 912 is arranged in the same manner as the first sub-microphone array 911 and is symmetrically distributed with respect to the user's head. .
  • the extension line of the first sub-microphone array 911 along the array direction intersects with the extension line of the second sub-microphone array 912 along the array direction, which can form a quadrilateral structure.
  • FIG. 9B is a schematic diagram of the positional relationship between the first sub-microphone array and the second sub-microphone array provided according to other embodiments of the present specification.
  • the first sub-microphone array 921 is located at the left ear of the user, and the first sub-microphone array 921 is arranged in a line.
  • the second sub-microphone array 922 is located at the right ear of the user.
  • the second sub-microphone array 922 is arranged in an approximately triangular shape.
  • the arrangement of the second sub-microphone array 922 is different from that of the first sub-microphone array 921 and is asymmetrically distributed with respect to the user's head. .
  • the extension line of the first sub-microphone array 921 along the array direction intersects with the extension line of the second sub-microphone array 922 along the array direction, which can form a triangular structure.
  • the first sub-microphone array 921 and the second sub-microphone array 922 can form a figure-eight shape, a circle, an ellipse, in addition to the quadrilateral shown in FIG. 9A and the triangle shown in FIG. 9B . , circular, polygonal and other regular and/or irregular shapes.
  • the first sub-microphone array and the second sub-microphone array are distributed in a specific shape or three-dimensional space, which can obtain the environmental noise of the user in all directions in an all-round way. Positioning, and then more accurately simulate the noise sound field at the user's ear canal, in order to achieve better noise reduction effect. Different arrangements of the first sub-microphone array and the second sub-microphone array have different spatial filtering performances.
  • the spatial filtering performance may include main lobe width, side lobe (also referred to as side lobe) width.
  • the main lobe width refers to the maximum radiation beam of acoustic radiation.
  • Sidelobe width refers to radiation beams other than the largest radiation beam. Among them, the narrower the main lobe width, the higher the resolution of the microphone array and the better the directivity. The lower the side lobe height, the better the anti-interference performance of the microphone array, and the higher the side lobe height, the worse the anti-jamming performance of the microphone array.
  • the width of the main lobe corresponding to the beam pattern of the cross-shaped array is narrower than that of the circular, rectangular or spiral wave pattern, that is to say, under the condition of the same number of array elements, the cross-shaped array has more High spatial resolution and better directivity.
  • the side lobe width corresponding to the beam pattern of the cross-shaped array is higher than that of the circular, rectangular or spiral wave pattern, that is to say, the anti-interference ability of the cross-shaped array is higher than Difference.
  • the arrangement of the first sub-microphone array and the second sub-microphone array may be adaptively adjusted according to the actual application situation, which is not further limited here. It should be noted that each short solid line in FIG. 9A and FIG.
  • each short solid line is a group of microphones
  • the number of each group of microphones may be the same or different
  • the types of each group of microphones may be the same or different
  • the orientation of each group of microphones may be the same or different.
  • the type, quantity, orientation and spacing can be adaptively adjusted according to the actual application.
  • a spatial super-resolution image of environmental noise can also be formed by methods such as synthetic aperture, sparse restoration, and co-element array, and the spatial super-resolution image can be used to reflect the signal reflection map of environmental noise, so as to further improve the spatial noise.
  • the positioning accuracy of the source In some embodiments, the position, spacing, on-off state, etc. of the microphones in the microphone array (eg, the first microphone array 130 and the second microphone array 160 ) can be adjusted based on the feedback of the positioning accuracy of the spatial noise source.
  • the first microphone array 130 may include a noise microphone, the noise microphone in the first microphone array 130 is used to pick up the spatial noise at the user's ear canal, and when the noise microphone picks up the spatial noise at the user's ear canal, Noise reduction waves output by the speaker array 150 are also picked up, which are not expected to be picked up by the noise microphone. Therefore, the noise microphone may be positioned at the acoustic zero point of the acoustic dipole formed in the speaker array 150 so that the noise reduction wave picked up by the noise microphone is minimized.
  • the at least one speaker array 150 forms at least one set of acoustic dipoles, and the noise microphone is located at the acoustic zero of the dipole radiated sound field.
  • the sound signals output by any two speakers in the speaker array 150 may be regarded as two point sound sources radiating sounds outward, and the radiated sounds have the same amplitude and opposite phases.
  • the two loudspeakers can form an acoustic dipole or similar acoustic dipoles, and the outwardly radiated sound has obvious directivity, forming a "8"-shaped sound radiation area. In the direction of the straight line where the two speakers are connected, the sound radiated at the speaker is the largest, the sound radiated in other directions is obviously reduced, and the sound radiated at the perpendicular line of the connection between the two speakers is the smallest.
  • the sound signal output by one speaker in the speaker array 150 may also be regarded as a dipole.
  • a set of sound signals with approximately opposite phases and approximately the same amplitude output from the front side of the speaker diaphragm and the back side of the speaker diaphragm in the speaker array 150 can be regarded as two point sound sources.
  • the ambient noise signal picked up by the microphone array (eg, the first microphone array 130 ) at the position of the acoustic zero point can also be acquired by an algorithm.
  • one or more microphones in the first microphone array 130 may be pre-set at the acoustic zero position of the acoustic dipole formed by the speaker array 150 of a specific frequency band.
  • the specific frequency band may be a frequency band that is critical to speech intelligibility, eg, 500Hz-1500Hz.
  • the signal processor 140 calculates and pre-stores compensation parameters for a specific frequency band based on the acoustic dipole position (ie, the positions of the two speakers that make up the acoustic dipole) and the acoustic transfer function.
  • the signal processor 140 may perform amplitude compensation and/or phase compensation on the ambient noise bases picked up by the remaining microphones in the first microphone array 130 (that is, the microphones not set at the acoustic zero position) according to the pre-stored compensation parameters.
  • the ambient noise signal is equivalent to the ambient noise signal picked up by the noise microphone set at the acoustic zero position.
  • the microphones in the microphone array may not be disposed at the acoustic zero point of the acoustic dipole formed by the speaker array 150 , for example, in some embodiments, the signal processor 140
  • the noise at the first spatial position picked up by the microphone array can be signal-separated and extracted according to the statistical distribution and structural features of different types of noise in different dimensions (eg, spatial domain, time domain, frequency domain, etc.), so as to obtain different types of noise (eg. noise of different frequencies, different phases, etc.), and the noise reduction waves emitted by the speaker array 150 picked up by the microphone array are eliminated by the signal processor.
  • FIG. 10 is an exemplary flowchart of estimating noise at a first spatial location provided according to some embodiments of the present specification. As shown in FIG. 10, the process 1000 may include:
  • step 1010 components associated with the signal picked up by the bone conduction microphone are removed from the picked up ambient noise in order to update the ambient noise.
  • this step may be performed by signal processor 140 .
  • the microphone array eg, the first microphone array 130, the second microphone array 160
  • the user's own speaking voice will also be picked up by the microphone array, that is, the user's own speaking voice will also be picked up by the microphone array. considered part of the ambient noise.
  • the noise reduction wave output by the speaker array 150 can cancel the user's own voice.
  • the user's own voice needs to be preserved, for example, in scenarios such as the user making a voice call or sending a voice message.
  • the bone conduction microphone can pick up the sound signal of the user's speech by picking up the vibration signal generated by the facial bones or muscles when the user speaks, and transmit it to the signal processor 140.
  • the signal processor 140 obtains parameter information from the sound signal picked up by the bone conduction microphone, and the signal processor 140 finds and removes the noise related to bone conduction from the ambient noise picked up by the microphone array (eg, the first microphone array 130, the second microphone array 160).
  • the sound signal component associated with the sound signal picked up by the microphone The signal processor 140 updates the ambient noise according to the parameter information of the ambient noise picked up by the remaining microphone arrays. The updated ambient noise no longer contains the voice signal of the user's own speech, that is, the user's own voice signal is retained when the user makes a voice call.
  • step 1020 the noise of the first spatial location is estimated according to the updated ambient noise.
  • this step may be performed by signal processor 140 .
  • the signal processor 140 may estimate the noise at the first spatial location based on the updated ambient noise. For a detailed description of estimating noise at the first spatial position according to environmental noise, reference may be made to FIG. 2 and related descriptions in this specification, and details are not repeated here.
  • the above description about the process 1000 is only for example and illustration, and does not limit the scope of application of this specification.
  • various modifications and changes can be made to the process 1000 under the guidance of this specification.
  • the components associated with the signals picked up by the bone conduction microphone can also be preprocessed, and the signals picked up by the bone conduction microphones can be transmitted to the terminal device as audio signals.
  • these corrections and changes are still within the scope of this specification.
  • the at least one microphone array may include a bone conduction microphone and an air conduction microphone
  • the signal processor 140 may control the on/off states of the bone conduction microphone and the air conduction microphone based on the working state of the open earphone 100 .
  • the working state of the open-back earphone 100 may refer to the usage state used when the user wears the open-back earphone 100 .
  • the working state of the open earphone 100 may include, but is not limited to, a music playing state, a voice calling state, a voice sending state, and the like.
  • the on/off state of the bone conduction microphone and the on/off state of the air conduction microphone in the microphone array may be determined according to the working state of the open earphone 100 .
  • the switch state of the bone conduction microphone may be the standby state, and the switch state of the air conduction microphone may be the working state.
  • the switch state of the bone conduction microphone may be the working state, and the switch state of the air conduction microphone may be the working state.
  • the signal processor 140 is coupled to the microphone array, and the signal processor 140 can control the switch state of the microphones (eg, bone conduction microphones, air conduction microphones) in the microphone array by sending control signals.
  • the working state of the open earphone 100 may include a talking state and a non-calling state.
  • the signal processor 140 may control the bone conduction microphone to be in a standby state.
  • the sound signal of the user's own speech may be regarded as environmental noise.
  • the sound signal of the user's own speech included in the environmental noise picked up by the microphone array may not be filtered out. Therefore, the voice signal of the user's own speech can also be canceled with the noise reduction wave output by the speaker array 150 .
  • the signal processor 140 may control the bone conduction microphone to be in the working state. For example, when the open earphone 100 is in a call state, the sound signal of the user speaking needs to be retained. In this case, the signal processor 140 can send a control signal to control the bone conduction microphone to be in the working state, and the bone conduction microphone can pick up the sound signal of the user speaking. , the signal processor 140 finds and removes the sound signal components associated with the sound signal picked up by the bone conduction microphone from the ambient noise picked up by the microphone array, so that the sound signal of the user's own speech is not in phase with the noise reduction wave output by the speaker array 150 offset, so as to ensure the normal call state of the user.
  • the signal processor 140 may control the bone conduction microphone to maintain the working state.
  • the sound pressure level of the ambient noise may reflect the intensity of the ambient noise.
  • the preset threshold here may be a value pre-stored in the open earphone 100, for example, any other value such as 50dB, 60dB, or 70dB.
  • the signal processor 140 can control the bone conduction microphone to maintain a working state by sending a control signal.
  • the bone conduction microphone can acquire the vibration signal of the facial muscles of the user when the user speaks without picking up external environmental noise. At this time, the vibration signal picked up by the bone conduction microphone is used. The signal is used as a voice signal during a call, so as to ensure the normal call of the user.
  • the signal processor 140 can control the bone conduction microphone to switch from the working state to the standby state.
  • the sound pressure level of the environmental noise is smaller than the preset threshold, the sound pressure level of the environmental noise is smaller than the sound pressure level of the sound signal generated by the user's speech, and the sound signal generated by the user's speech is transmitted by the speaker array 150 .
  • the signal processor 140 can control the bone conduction microphone to switch from the working state to the standby state by sending a control signal, thereby reducing the signal processing complexity and the power consumption of the open earphone 100 .
  • the open-back earphone 100 may further include an adjustment module for adjusting the sound pressure level of the noise reduction wave.
  • the adjustment module may include buttons, voice assistants, gesture sensors, and the like.
  • the user can adjust the noise reduction mode of the open earphone 100 by controlling the adjustment module. Specifically, the user can adjust (eg, amplify or reduce) the amplitude information of the noise reduction signal by controlling the adjustment module to change the sound pressure level of the noise reduction wave emitted by the speaker array, thereby achieving different noise reduction effects.
  • the noise reduction mode may include a strong noise reduction mode, a medium noise reduction mode, a weak noise reduction mode, and the like.
  • the user when the user wears the open-type earphone 100 indoors, the external environment noise is small, and the user can turn off or adjust the noise reduction mode of the open-type earphone to a weak noise reduction mode through the adjustment module.
  • the user when the user wears the open earphone 100 when walking in public places such as the street, the user needs to maintain a certain ability to perceive the surrounding environment while listening to audio signals (eg, music, voice information) to cope with emergencies , at this time, the user can select the intermediate noise reduction mode by adjusting the module (for example, a button or a voice assistant) to retain the surrounding ambient noise (such as alarm sound, impact sound, car honking, etc.).
  • the module for example, a button or a voice assistant
  • the signal processor 140 may also send prompt information to the open earphone 100 or a terminal device (eg, a mobile phone, a smart watch, etc.) communicatively connected to the open earphone 100 based on the ambient noise intensity range, so as to remind the user to adjust Noise reduction mode.
  • a terminal device eg, a mobile phone, a smart watch, etc.
  • aspects of this application may be illustrated and described in several patentable categories or situations, including any new and useful process, machine, product, or combination of matter, or combinations of them. of any new and useful improvements. Accordingly, various aspects of the present application may be performed entirely by hardware, entirely by software (including firmware, resident software, microcode, etc.), or by a combination of hardware and software.
  • the above hardware or software may be referred to as a "data block”, “module”, “engine”, “unit”, “component” or “system”.
  • aspects of the present application may be embodied as a computer product comprising computer readable program code embodied in one or more computer readable media.
  • a computer storage medium may contain a propagated data signal with the computer program code embodied therein, for example, on baseband or as part of a carrier wave.
  • the propagating signal may take a variety of manifestations, including electromagnetic, optical, etc., or a suitable combination.
  • Computer storage media can be any computer-readable media other than computer-readable storage media that can communicate, propagate, or transmit a program for use by coupling to an instruction execution system, apparatus, or device.
  • Program code on a computer storage medium may be transmitted over any suitable medium, including radio, cable, fiber optic cable, RF, or the like, or a combination of any of the foregoing.

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Abstract

本申请公开一种开放式耳机,该开放式耳机包括:固定结构,被配置为将所述耳机固定在用户耳朵附近且不堵塞用户耳道的位置;壳体结构,被配置为承载:第一麦克风阵列,被配置为拾取环境噪声;至少一个扬声器阵列;以及信号处理器,被配置为:基于所述拾取的环境噪声估计第一空间位置的噪声,所述第一空间位置比所述第一麦克风阵列中任一麦克风更加靠近用户耳道;以及基于所述第一空间位置的噪声生成降噪信号,使得所述至少一个扬声器阵列根据所述降噪信号输出降噪声波,所述降噪声波用于消除传递到用户耳道的环境噪声。

Description

一种开放式耳机 技术领域
本申请涉及声学领域,特别涉及一种开放式耳机。
背景技术
耳机设备允许用户在收听音频内容、进行语音通话的同时保证用户交互内容的私密性,且收听时不打扰到周围人群。耳机设备通常可以分为入耳式耳机设备和开放式耳机设备两类。入耳式耳机设备在使用过程中会堵塞用户耳部,且用户在长时间佩戴时容易产生堵塞、异物、胀痛等感受。开放式耳机设备可以开放用户耳部,有利于长期佩戴,但当外界噪声较大时,其降噪效果不明显,使得用户收听体验较差。
因此,希望提供一种开放式耳机,可以开放用户双耳以及提高用户听感体验。
发明内容
本申请实施例提供一种开放式耳机,包括:固定结构,被配置为将所述耳机固定在用户耳朵附近且不堵塞用户耳道的位置;壳体结构,被配置为承载:第一麦克风阵列,被配置为拾取环境噪声;至少一个扬声器阵列;以及信号处理器,被配置为:基于所述拾取的环境噪声估计第一空间位置的噪声,所述第一空间位置比所述第一麦克风阵列中任一麦克风更加靠近用户耳道;以及基于所述第一空间位置的噪声生成降噪信号,使得所述至少一个扬声器阵列根据所述降噪信号输出降噪声波,所述降噪声波用于消除传递到用户耳道的环境噪声。
在一些实施例中,所述壳体结构被配置为容纳第二麦克风阵列,所述第二麦克风阵列被配置为拾取环境噪声和所述降噪声波,所述第二麦克风阵列至少部分区别于所述第一麦克风阵列;以及所述信号处理器被配置为基于所述第二麦克风阵列拾取的声音信号更新所述降噪信号。
在一些实施例中,所述基于所述第二麦克风阵列拾取的声音信号更新所述降噪信号包括:基于所述第二麦克风阵列拾取的声音信号,对用户耳道处的声场进行估计;以及根据用户耳道处的声场,调整所述降噪信号的参数信息。
在一些实施例中,所述信号处理器进一步被配置为:获取用户输入;以及根据用户输入调整所述降噪信号的参数信息。
在一些实施例中,所述第二麦克风阵列包括一个比所述第一麦克风阵列中任意麦克风更加靠近用户耳道的麦克风。
在一些实施例中,所述信号处理器基于所述拾取的环境噪声估计第一空间的噪声包括:根据所述拾取的环境噪声进行信号分离,获取所述环境噪声对应的参数信息,基于所述参数信息生成降噪信号。
在一些实施例中,所述信号处理器基于所述拾取的环境噪声估计第一空间位置的噪声包括:确定一个或多个与所述拾取的环境噪声有关的空间噪声源;以及基于所述空间噪声源,估计所述第一空间位置的噪声。
在一些实施例中,所述确定一个或多个与所述拾取的环境噪声有关的空间噪声源包括:将所述拾取的环境噪声划分为多个子带,每个子带对应不同的频率范围;以及在至少一个子带上,确定与其对应的空间噪声源。
在一些实施例中,所述第一麦克风阵列包括第一子麦克阵列和第二子麦克风阵列,所述第一子麦克风阵列和所述第二子麦克风阵列分别位于用户的左耳和右耳处,所述确定与所述至少一个子带对应的空间噪声源包括:获取用户头函数,所述用户头函数反映用户头部对声音的反射或吸收情况;以及在所述至少一个子带上,结合第一子麦克风阵列拾取的环境噪声、第二子麦克风阵列拾取的环境噪声,以及所述用户头函数,确定与其对应的空间噪声源。
在一些实施例中,所述确定一个或多个与所述拾取的环境噪声有关的空间噪声源包括:通过波束形成、超分辨空间谱估计或到达时差中的一种或多种方式定位所述一个或多个空间噪声源。
在一些实施例中,所述第一麦克风阵列包括一个噪声麦克风,所述至少一个扬声器阵列形成至少一组声学偶极子,且所述噪声麦克风位于所述偶极子辐射声场的声学零点处。
在一些实施例中,所述至少一个麦克风阵列包括骨导麦克风,所述骨导麦克风被配置于拾取用户的说话声音,所述信号处理器基于所述拾取的环境噪声估计第一空间位置的噪声包括:从所述拾取的环境噪声中去除与所述骨导麦克风拾取的信号相关联的成分,以更新所述环境噪声;以及根据所述更新后的环境噪声估计第一空间位置的噪声。
在一些实施例中,所述至少一个麦克风阵列包括骨传导麦克风和气传导麦克风,所述信号处理器基于所述耳机的工作状态控制所述骨传导麦克风和所述气传导麦克风的开关状态。
在一些实施例中,所述耳机的状态包括通话状态和未通话状态,若所述耳机的工作状态为未通话状态,则所述信号处理器控制所述骨传导麦克风为待机状态;以及若 所述耳机的工作状态为通话状态,则所述信号处理器控制所述骨传导麦克风为工作状态。
在一些实施例中,所述耳机的工作状态为通话状态时,若所述环境噪声的声压级大于预设阈值时,所述信号处理器控制所述骨传导麦克风保持工作状态;若所述环境噪声的声压级小于预设阈值时,所述信号处理器控制所述骨传导麦克风由工作状态切换至待机状态。
附图说明
本申请将以示例性实施例的方式进一步说明,这些示例性实施例将通过附图进行详细描述。这些实施例并非限制性的,在这些实施例中,相同的编号表示相同的结构,其中:
图1是根据本申请的一些实施例提供的开放式耳机的示例性框架结构图;
图2是根据本申请的一些实施例提供的开放式耳机的示例性原理流程图;
图3是根据本申请一些实施例提供的更新降噪信号的示例性流程图;
图4A是根据本申请说明书一些实施例提供的第一麦克风阵列和第二麦克风阵列的排布方式和位置关系的示例性分布图;
图4B是根据本申请说明书另一些实施例提供的第一麦克风阵列和第二麦克风阵列的排布方式的示例性分布图;
图5是根据本申请说明书一些实施例提供的估计第一空间位置的噪声的示例性流程图;
图6是根据本申请说明书一些实施例提供的确定空间噪声源的示例性流程图;
图7是根据本申请说明书一些实施例提供的确定空间噪声源的另一示例性流程图;
图8A是根据本申请说明书一些实施例提供的第一子麦克风阵列的排布方式的示意图;
图8B是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图;
图8C是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图;
图8D是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图;
图9A是根据本申请说明书一些实施例提供的第一子麦克风阵列和第二子麦克风阵列的位置关系的示意图;
图9B是根据本申请说明书另一些实施例提供的第一子麦克风阵列和第二子麦克风阵列的位置关系的示意图;
图10是根据本申请说明书一些实施例提供的估计第一空间位置的噪声的示例性流程图。
具体实施方式
为了更清楚地说明本申请实施例的技术方案,下面将对实施例描述中所需要使用的附图作简单的介绍。显而易见地,下面描述中的附图仅仅是本申请的一些示例或实施例,对于本领域的普通技术人员来讲,在不付出创造性劳动的前提下,还可以根据这些附图将本申请应用于其它类似情景。除非从语言环境中显而易见或另做说明,图中相同标号代表相同结构或操作。
应当理解,本文使用的“***”、“装置”、“单元”和/或“模组”是用于区分不同级别的不同组件、元件、部件、部分或装配的一种方法。然而,如果其他词语可实现相同的目的,则可通过其他表达来替换所述词语。
如本申请和权利要求书中所示,除非上下文明确提示例外情形,“一”、“一个”、“一种”和/或“该”等词并非特指单数,也可包括复数。一般说来,术语“包括”与“包含”仅提示包括已明确标识的步骤和元素,而这些步骤和元素不构成一个排它性的罗列,方法或者设备也可能包含其它的步骤或元素。
本申请中使用了流程图用来说明根据本申请的实施例的***所执行的操作。应当理解的是,前面或后面操作不一定按照顺序来精确地执行。相反,可以按照倒序或同时处理各个步骤。同时,也可以将其他操作添加到这些过程中,或从这些过程移除某一步或数步操作。
开放式耳机是一种可以开放用户耳部的耳机设备。开放式耳机可以通过固定结构(例如,耳挂、头挂等)将扬声器固定于用户耳朵附近且不堵塞用户耳道的位置。当用户使用开放式耳机时,外界环境噪音也可以被用户听到,这就使得用户的听感体验较差。例如,在外界环境噪音较大的场所(例如,街道、景区等),用户在使用开放式耳机进行音乐播放时,外界环境的噪音会直接进入用户耳道,使得用户听到较大的环境噪音,环境噪音会干扰用户的听音乐体验。又例如,当用户佩戴开放式耳机进行通话时, 麦克风不仅会拾取用户自身的说话声音,也会拾取环境噪音,使得用户通话体验较差。
基于上述问题,本说明书实施例中提供一种开放式耳机,在一些实施例中,开放式耳机可以包括固定结构、壳体结构、第一麦克风阵列、至少一个扬声器阵列以及信号处理器。其中,固定结构被配置为将开放式耳机固定在用户耳朵附近且不堵塞用户耳道的位置。壳体结构被配置为容纳第一麦克风阵列、至少一个扬声器阵列以及信号处理器。第一麦克风阵列被配置为拾取环境噪声。信号处理器被配置为基于第一麦克风阵列拾取的环境噪声估计第一空间位置的噪声,并基于第一空间位置的噪声生成降噪信号,这里的第一空间位置比第一麦克风阵列中任一麦克风更加靠近用户耳道。这里可以理解为,第一麦克风阵列中的各麦克风可以分布于用户耳道附近的不同位置,通过对第一麦克风阵列中的各麦克风采集的环境噪声信号来估计靠近用户耳道位置处(例如,第一空间位置)的噪声。至少一个扬声器阵列被配置为基于降噪信号输出降噪声波,该降噪声波可以用于消除传递到用户耳道的环境噪声。在一些实施例中,开放式耳机还可以包括第二麦克风阵列,第二麦克风阵列被配置为拾取环境噪声和至少一个扬声器阵列输出的降噪声波。在一些实施例中,信号处理器可以基于第二麦克风阵列拾取的声音信号更新降噪信号。例如,信号处理器可以基于第二麦克风阵列拾取的声音信号对用户耳道处的声场进行估计,并根据用户耳道处的声场调整降噪信号的相位或幅值,实现降噪信号的更新。本说明书的实施例中,通过上述方法利用降噪声波消除用户耳道处的环境噪声,实现了开放式耳机的主动降噪,提高了用户在使用开放式耳机过程中的听觉体验。
图1是根据本申请的一些实施例提供的开放式耳机的示例性框架结构图。如图1所示,开放式耳机100可以包括壳体结构110、第一麦克风阵列130、信号处理器140和扬声器阵列150,其中,第一麦克风阵列130、信号处理器140和扬声器阵列150位于壳体结构110处。开放式耳机100可以通过固定结构120将耳机固定于用户耳朵附近且不堵塞用户耳道。在一些实施例中,位于壳体结构110处的第一麦克风阵列130可以拾取用户耳道处的环境噪声,并将拾取到的环境噪声信号转换为电信号传递至信号处理器140进行信号处理。信号处理器140耦接第一麦克风阵列130和扬声器阵列150,信号处理器140接收第一麦克风阵列130拾取的环境噪声信号并对其进行信号处理以获取环境噪声的参数信息(例如,幅值信息、相位信息等)。信号处理器140可以基于环境噪声的参数信息(例如,幅值信息、相位信息等)估计用户第一空间位置处的噪声,并基于第一空间位置处的噪声生成降噪信号。该降噪信号的参数信息与环境噪声的参数信息相对应,例如,降噪信号的幅值大小与环境噪声的幅值大小近似相等,降噪信号的 相位与环境噪声的相位近似相反。信号处理器140将生成的降噪信号传递至扬声器阵列150。扬声器阵列150可以根据信号处理器140生成的降噪信号输出降噪声波,该降噪声波可以与用户耳道位置处的环境噪声相互抵消,从而实现开放式耳机100的主动降噪,提高用户在使用开放式耳机100过程中的听觉体验。
壳体结构110可以被配置为承载第一麦克风阵列130、信号处理器140以及扬声器阵列150。在一些实施例中,壳体结构110可以是内部中空的封闭式或半封闭式壳体结构,且第一麦克风阵列130、信号处理器140以及扬声器阵列150位于壳体结构110处。在一些实施例中,壳体结构110的形状可以为长方体、圆柱体、圆台等规则或不规则形状的立体结构。当用户佩戴开放式耳机100时,壳体结构110可以位于靠近用户耳朵附近的位置,例如,壳体结构110可以位于用户耳廓的周侧(例如,前侧或后侧),或者位于用户耳朵上但不堵塞或覆盖用户的耳道。在一些实施例中,开放式耳机100可以为骨传导耳机,壳体结构110的至少一侧可以与用户的头部皮肤接触。骨传导耳机中声学驱动器(例如,振动扬声器)将音频信号转换为机械振动,该机械振动可以通过壳体结构110以及用户的骨骼传递至用户的听觉神经。在一些实施例中,开放式耳机100可以为气传导耳机,壳体结构110的至少一侧可以与用户的头部皮肤接触或不接触。壳体结构110的侧壁上包括至少一个导声孔,气传导耳机中的扬声器将音频信号转换为气导声音,该气导声音可以通过导声孔向用户耳朵的方向进行辐射。
第一麦克风阵列130可以被配置为拾取环境噪声。在一些实施例中,环境噪声是指用户所处环境中的多种外界声音的组合。在一些实施例中,环境噪声可以包括交通噪声、工业噪声、建筑施工噪声、社会噪声等中的一种或多种。在一些实施例中,交通噪声可以包括但不限于机动车辆的行驶噪声、鸣笛噪声等。工业噪声可以包括但不限于工厂动力机械运转噪声等。建筑施工噪声可以包括但不限于动力机械挖掘噪声、打洞噪声、搅拌噪声等。社会生活环境噪声可以包括但不限于群众集会噪声、文娱宣传噪声、人群喧闹噪声、家用电器噪声等。在一些实施例中,第一麦克风阵列130可以设置于用户耳道附近位置,用于拾取传递至用户耳道处的环境噪声,第一麦克风阵列130可以将拾取的环境噪声信号转换为电信号并传递至信号处理器140进行信号处理。在一些实施例中,环境噪声也可以包括用户讲话的声音。例如,当开放式耳机100为未通话状态时,用户自身说话产生的声音也可以视为环境噪声,第一麦克风阵列130可以拾取用户自身说话的声音以及其他环境噪声,并将用户说话产生的声音信号和其他环境噪声转化为电信号传递至信号处理器140进行信号处理。在一些实施例中,第一麦克风阵列130可以 分布于用户的左耳或右耳处。在一些实施例中,第一麦克风阵列130还可以位于用户的左耳和右耳处。例如,第一麦克风阵列130可以包括第一子麦克风阵列和第二子麦克风阵列,其中,第一子麦克风阵列位于用户的左耳处,第二子麦克风阵列位于用户的右耳处,第一子麦克风阵列和第二子麦克风阵列可以同时进入工作状态或二者中的一个进入工作状态。
在一些实施例中,第一麦克风阵列130可以包括气传导麦克风和/或骨传导麦克风。例如,在一些实施例中,第一麦克风阵列130可以包括一个或多个气传导麦克风。例如,用户在使用开放式耳机100听取音乐时,气传导麦克风可以同时获取外界环境的噪声和用户说话时的声音并将其作为环境噪声转换为电信号传输至信号处理器140中进行处理。在一些实施例中,第一麦克风阵列130还可以包括一个或多个骨传导麦克风。在一些实施例中,骨传导麦克风可以直接与用户的头部皮肤接触,用户说话时面部骨骼或肌肉产生的振动信号可以直接传递给骨传导麦克风,进而骨传导麦克风将振动信号转换为电信号,并将电信号传递至信号处理器140进行信号处理。在一些实施例中,骨传导麦克风也可以不与人体直接接触,用户说话时面部骨骼或肌肉产生的振动信号可以先传递至壳体结构110,再由壳体结构110传递至骨传导麦克风,骨传导麦克风进一步将该人体振动信号转换为包含语音信息的电信号。例如,用户在通话状态时,信号处理器140可以将气传导麦克风采集的声音信号作为环境噪声进行降噪处理,骨传导麦克风采集的声音信号作为语音信号进行保留,从而保证用户通话时的通话质量。在一些实施例中,根据麦克风的工作原理作为分类,第一麦克风阵列130可以包括动圈式麦克风、带式麦克风、电容式麦克风、驻极体式麦克风、电磁式麦克风、碳粒式麦克风等,或其任意组合。在一些实施例中,第一麦克风阵列130的阵列排布方式可以是线性阵列(例如,直线形、曲线形)、平面阵列(例如,十字形、圆形、环形、多边形、网状形等规则和/或不规则形状)或立体阵列(例如,圆柱状、球状、半球状、多面体等),关于第一麦克风阵列130的排布方式具体可以参考本说明书图8及其相关内容。
信号处理器140被配置为基于第一麦克风阵列130拾取的环境噪声估计第一空间位置的噪声,并基于第一空间位置的噪声生成降噪信号。第一空间位置是指靠近用户耳道特定距离的空间位置,该第一空间位置比第一麦克风阵列130中任一麦克风更加靠近用户耳道。这里的特定距离可以是固定的距离,例如,0.5cm、1cm、2cm、3cm等。在一些实施例中,第一空间位置与第一麦克风阵列130中各麦克风相对于用户耳朵的分布位置、数量相关,通过调整第一麦克风阵列130中各麦克风相对于用户耳朵的分布位 置和/或数量可以对第一空间位置进行调整。例如,通过增加第一麦克风阵列130中麦克风的数量可以使第一空间位置更加靠近用户耳道。
信号处理器140可以对接收到的环境噪声信号进行信号处理,估计第一空间位置的噪声。在一些实施例中,信号处理器140可以耦合到第一麦克风阵列130和扬声器阵列150,信号处理器140可以接收第一麦克风阵列130拾取的环境噪声以估计第一空间位置的噪声。例如,信号处理器140可以确定一个或多个与拾取的环境噪声有关的空间噪声源。又例如,信号处理器140可以对环境噪声有关的空间噪声源进行方位估计、相位信息估计、幅值信息估计等。信号处理器140可以根据第一空间位置的噪声估计(例如,相位信息、幅值信息)生成降噪信号。降噪信号是指与第一空间位置的噪声的幅值大小近似相等、相位近似相反的声音信号。
扬声器阵列150被配置为基于降噪信号输出降噪声波,该降噪声波用于消除传递到用户耳道的环境噪声。在一些实施例中,扬声器阵列150可以设置于壳体结构110处,当用户佩戴开放式耳机100时,扬声器阵列150可以位于用户耳部的附近位置。在一些实施例中,扬声器阵列150可以基于降噪信号输出降噪声波以与第一空间位置的环境噪声相抵消。仅作为示例性说明,例如,信号处理器140控制扬声器阵列150输出与第一空间位置的噪声的幅值大小近似相等、相位近似相反的声音信号以抵消第一空间位置的噪声。在一些实施例中,第一空间位置与用户耳道之间的间距较小,第一空间位置的噪声可以近似视为传递到用户耳朵处的噪声,扬声器阵列150基于降噪信号输出的降噪声波可以与第一空间位置的噪声相互抵消,可以近似为传递至用户耳道的环境噪声被消除。在一些实施例中,根据扬声器的工作原理进行分类,扬声器阵列150可以包括电动式扬声器(例如,动圈式扬声器)、磁式扬声器、离子扬声器、静电式扬声器(或电容式扬声器)、压电式扬声器等中的一种或多种。在一些实施例中,根据扬声器输出的声音的传播方式进行分类,扬声器阵列150可以包括气传导扬声器或骨传导扬声器。例如,扬声器阵列150只包括气传导扬声器时,扬声器阵列150中的部分气传导扬声器可以用于输出降噪声波以消除噪声,扬声器阵列150中的其他气传导扬声器可以用于向用户传递用户需要听取的声音信息(例如,设备媒体音频、通话远端音频)。在一些实施例中,扬声器阵列150中用于向用户传递用户需要听取的声音信息的扬声器也可以用于输出降噪声波。又例如,扬声器阵列150包括骨传导扬声器和气传导扬声器时,气传导扬声器可以用于输出降噪声波以消除噪声,骨传导扬声器可以用于向用户传递用户需要听取的声音信息,相比于气传导扬声器,骨传导扬声器将机械振动直接通过用户的身体 (例如,骨骼、皮肤组织等)传递至用户的听觉神经,在此过程中对于拾取环境噪声的气传导麦克风的干扰较小。骨传导扬声器在将机械振动传递至用户的过程中会造成壳体结构110产生机械振动,壳体结构110产生的机械振动作用于空气产生气传导声音,在一些实施例中,壳体结构110产生的气传导声音也可以作为降噪声波。需要注意的是,在一些实施例中,扬声器阵列150可以是独立的功能器件,也可以是能够实现多个功能的单个器件的一部分。在一些实施例中,信号处理器140可以和扬声器阵列150集成在一起和/或形成为一体。在一些实施例中,扬声器阵列150的排布方式可以是线性阵列(例如,直线形、曲线形)、平面阵列(例如,十字形、网状形、圆形、环形、多边形等规则和/或不规则形状)或立体阵列(例如,圆柱状、球状、半球状、多面体等),本说明书在此不做限定。
在一些实施例中,开放式耳机100还可以包括固定结构120,固定结构120被配置为将开放式耳机100固定在用户耳朵附近且不堵塞用户耳道的位置。在一些实施例中,固定结构120可以包括耳挂、头梁或弹性带等,使得开放式耳机100可以更好地固定在用户耳朵附近位置,防止用户在使用时发生掉落。仅作为示例性说明,例如,固定结构120可以为耳挂,耳挂可以被配置为围绕耳部区域佩戴。又例如,固定结构120可以为颈带,被配置为围绕颈/肩区域佩戴。在一些实施例中,耳挂可以是连续的钩状物,并可以被弹性地拉伸以佩戴在用户的耳部,同时耳挂还可以对用户的耳廓施加压力,使得开放式耳机100牢固地固定在用户的耳部或头部的特定位置上。在一些实施例中,耳挂可以是不连续的带状物。例如,耳挂可以包括刚性部分和柔性部分,其中,刚性部分可以由刚性材料(例如,塑料或金属)制成,刚性部分可以与声学输出装置的壳体结构通过物理连接(例如,卡接、螺纹连接等)的方式进行固定。柔性部分可以由弹性材料制成(例如,布料、复合材料或/和氯丁橡胶)。
应当注意的是,以上关于图1和图2的描述仅仅是出于说明的目的而提供的,并不旨在限制本申请的范围。对于本领域的普通技术人员来说,根据本公开的指导可以做出多种变化和修改。然而,这些变化和修改不会背离本申请的范围。例如,开放式耳机100中的一个或多个元件(例如,固定结构120等)可以省略。在一些实施例中,一个元件可以被其他能实现类似功能的元件替代。例如,在一些实施例中,开放式耳机100可以不包括固定结构120,壳体结构110可以为具有人体耳朵适配形状的壳体结构,例如圆环形、椭圆形、多边形(规则或不规则)、U型、V型、半圆形,以便壳体结构110可以挂靠在用户的耳朵附近。在一些实施例中,一个元件可以拆分成多个子元件, 或者多个元件可以合并为单个元件。
图2是根据本申请的一些实施例提供的开放式耳机的示例性原理流程图。如图2所示,流程200可以包括:
在步骤210中,拾取环境噪声。
在一些实施例中,该步骤可以由第一麦克风阵列130执行。在一些实施例中,环境噪声是指用户所处环境中的多种外界声音的组合。在一些实施例中,环境噪声可以包括交通噪声、工业噪声、建筑施工噪声、社会噪声等中的一种或多种。在一些实施例中,交通噪声可以包括但不限于机动车辆的行驶噪声、鸣笛噪声等。工业噪声可以包括但不限于工厂动力机械运转噪声等。建筑施工噪声可以包括但不限于动力机械挖掘噪声、打洞噪声、搅拌噪声等。社会生活环境噪声可以包括但不限于群众集会噪声、文娱宣传噪声、人群喧闹噪声、家用电器噪声等。在一些实施例中,第一麦克风阵列130可以位于用户耳道的附近位置,用于拾取传递至用户耳道处的环境噪声,第一麦克风阵列130可以将拾取的环境噪声信号转换为电信号并传递至信号处理器140进行信号处理。在一些实施例中,环境噪声也可以包括用户讲话的声音。例如,当开放式耳机100为未通话状态时(例如收听音频或观看视频时),用户自身说话产生的声音也可以视为环境噪声,第一麦克风阵列130可以拾取用户自身说话的声音以及其他环境噪声,并将用户说话产生的声音信号和其他环境噪声转化为电信号传递至信号处理器140进行信号处理。
在步骤220中,基于拾取的环境噪声估计第一空间位置的噪声。
在一些实施例中,该步骤可以由信号处理器140执行。第一空间位置是指靠近用户耳道特定距离的空间位置。这里的特定距离可以是固定的距离,例如,0.5cm、1cm、2cm、3cm等,可以根据实际应用情况进行适应性调整。第一麦克风阵列130拾取的环境噪声可以是来自不同方位、不同种类的空间噪声源,因而每一个空间噪声源对应的参数信息(例如,相位信息、幅值信息)是不同的。在一些实施例中,信号处理器140可以根据不同类型的噪声在不同维度(例如,空域、时域、频域等)的统计分布和结构化特征将第一空间位置的噪声进行信号分离提取,从而估计不同类型(例如不同频率、不同相位等)的噪声,并估计每种噪声所对应的参数信息(例如,幅值信息、相位信息等)。在一些实施例中,信号处理器140还可以将根据第一空间位置处不同类型噪声对应的参数信息确定第一空间位置的噪声的整体参数信息。在一些实施例中,基于拾取的环境噪声估计第一空间位置的噪声还可以包括确定一个或多个与拾取的环境噪声有关的空间噪声源,基于空间噪声源估计第一空间位置的噪声。例如,将拾取的环境噪声划 分为多个子带,每个子带对应不同的频率范围,在至少一个子带上,确定与其对应的空间噪声源。需要注意的是,这里通过子带估计的空间噪声源是与外界真实噪声源对应的虚拟噪声源。关于基于拾取的环境噪声估计第一空间位置的噪声的具体内容可以参考本申请说明书其它地方,例如,图5-图7及其相应描述。
开放式耳机100不堵塞用户耳道,无法通过在耳道处设置麦克风的方式获取环境噪声,第一空间位置是第一麦克风阵列130所构造的用于模拟用户耳道位置的空间区域,为了更加精确地估计用户耳道处传递的环境噪声,在一些实施例中,第一空间位置比第一麦克风阵列130中任一麦克风更加靠近用户耳道。在一些实施例中,第一空间位置与第一麦克风阵列130中各麦克风相对于用户耳朵的分布位置、数量相关,通过调整第一麦克风阵列130中各麦克风相对于用户耳朵的分布位置或数量可以对第一空间位置进行调整。例如,通过增加第一麦克风阵列130中麦克风的数量可以使第一空间位置更加靠近用户耳道。又例如,还可以通过减小第一麦克风阵列130中各麦克风的间距使第一空间位置更加靠近用户耳道。再例如,还可以通过改变第一麦克风阵列130中各麦克风的排列方式使第一空间位置更加靠近用户耳道。
在步骤230中,基于第一空间位置的噪声生成降噪信号。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,信号处理器140可以基于步骤220中获得的第一空间位置的噪声的参数信息(例如,幅值信息、相位信息等)生成降噪信号。例如,降噪声波的相位可以与第一空间位置的噪声的相位近似相反。又例如,降噪声波的相位可以与第一空间位置的噪声的相位近似相反,并且该降噪信号的幅值大小可以与第一空间位置的噪声的幅值大小近似相等。在一些实施例中,扬声器阵列150可以基于信号处理器140生成的降噪信号输出降噪声波,降噪声波可以与第一空间位置的噪声相互抵消。在一些实施例中,第一空间位置的噪声可以近似视为用户耳道位置的噪声,因此,降噪信号与第一空间位置的噪声相互抵消,可以近似为传递至用户耳道的环境噪声被消除。在一些实施例中,开放式耳机100可以通过图2中所描述的方法步骤消除用户耳道位置处的环境噪声,实现开放式耳机100的主动降噪。
应当注意的是,上述有关流程200的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程200进行各种修正和改变。然而,这些修正和改变仍在本说明书的范围之内。例如,还可以增加、省略或合并流程200中的步骤,比如,还可以对环境噪声进行信号处理(例如,滤 波处理等)。
继续参照图1,在一些实施例中,开放式耳机100还可以包括第二麦克风阵列160。第二麦克风阵列160可以位于壳体结构110的内部。在一些实施例中,第二麦克风阵列160至少部分区别于第一麦克风阵列130。例如,第二麦克风阵列160中的麦克风与第一麦克风阵列130中的麦克风的数量、种类、位置、排布方式等中的一种或多种不同。在一些实施例中,例如,第一麦克风阵列130中麦克风排布方式可以是线形的,第二麦克风阵列160中麦克风的排布方式可以是圆形的。又例如,第二麦克风阵列160中的麦克风可以只包括气传导麦克风,第一麦克风阵列130中可以包括气传导麦克风和骨传导麦克风。在一些实施例中,第二麦克风阵列160中的麦克风可以是第一麦克风阵列130中包括的任意一个或多个麦克风,第二麦克风阵列160中的麦克风也可以独立于第一麦克风阵列130的麦克风。在一些实施例中,第二麦克风阵列160可以被配置为拾取环境噪声和扬声器阵列150输出的降噪声波。第二麦克风阵列160拾取的环境噪声和降噪声波可以传递至信号处理器140。在一些实施例中,信号处理器140可以基于第二麦克风阵列160拾取的声音信号更新降噪信号。例如,信号处理器140可以根据第二麦克风阵列160拾取的声音信号的参数信息(例如,频率信息、幅值信息、相位信息等)调整降噪信号的参数信息,使得调整后的降噪信号的幅值能够与第一空间位置的噪声的幅值更加吻合,或调整后的降噪信号的相位能够与第一空间位置的噪声的相位的反相位更加吻合,从而使得更新后的降噪声波与第一空间位置的噪声可以更全面的抵消。关于基于第二麦克风阵列160拾取的声音信号更新降噪信号的具体内容可以参考本说明书图3及其相关描述。
图3是根据本申请一些实施例提供的更新降噪信号的示例性流程图。如图3所示,流程300可以包括:
在步骤310中,基于第二麦克风阵列160拾取的声音信号,对用户耳道处的声场进行估计。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,第二麦克风阵列160拾取的声音信号包括环境噪声和扬声器阵列150输出的降噪声波。在一些实施例中,环境噪声和扬声器阵列150输出的降噪声波相抵消后,用户耳道附近可能仍会存在一部分未相互抵消掉的声音信号,这些未抵消掉的声音信号可以是残余的环境噪声和/残余的降噪声波,因此使得环境噪声和降噪声波抵消后用户耳道处仍存在一定的噪声。信号处理器140可以根据第二麦克风阵列160拾取的声音信号(例如,环境噪 声、降噪声波)进行信号处理,得到用户耳道处的声场的参数信息,例如,频率信息、幅值信息和相位信息等,从而实现对用户耳道处的声场估计。
在步骤320中,根据用户耳道处的声场,调整降噪信号的参数信息。
在一些实施例中,步骤320可以由信号处理器140执行。在一些实施例中,信号处理器140可以根据步骤310中得到的用户耳道处的声场的参数信息,调整降噪信号的参数信息(例如,频率信息、幅值信息和/或相位信息),使得更新后降噪信号的幅值信息、频率信息与用户耳道处的环境噪声的幅值信息、频率信息更加吻合,且更新后降噪信号的相位信息与用户耳道处的环境噪声的反相位信息更加吻合,从而使得更新后降噪信号可以更加精准的消除环境噪声。
应当注意的是,上述有关流程300的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程300进行各种修正和改变。然而,这些修正和改变仍在本说明书的范围之内。例如,拾取用户耳道处的声场的麦克风阵列不限于第二麦克风阵列,还可以包括其它麦克风阵列,例如第三麦克风阵列、第四麦克风阵列等,可以将多个麦克风阵列拾取的用户耳道处的声场的相关参数信息以平均或加权算法等方式对用户耳道处的声场进行估计。
在一些实施例中,为了更加精准地获取用户耳道处的声场,第二麦克风阵列160包括一个比第一麦克风阵列130中任意麦克风更加靠近用户耳道的麦克风。在一些实施例中,第一麦克风阵列130拾取的声音信号是环境噪声,第二麦克风阵列160拾取的声音信号是环境噪声和降噪声波。在一些实施例中,信号处理器140可以根据第二麦克风阵列160拾取的声音信号对用户耳道处的声场进行估计,以更新降噪信号。第二麦克风阵列160需要对降噪信号与环境噪声抵消后用户耳道处的声场进行监测,第二麦克风阵列160包括一个比第一麦克风阵列130中任意麦克风更加靠近用户耳道的麦克风可以更加准确的表征用户听到的声音信号,通过第二麦克风阵列160的声场进行估计以更新降噪信号,可以进一步提高降噪效果和用户的听觉体验感。
在一些实施例中,第一麦克风阵列130和第二麦克风阵列160的排布方式可以相同。这里的第一麦克风阵列130和第二麦克风阵列160的排布方式相同可以理解为二者的排布形状近似相同。图4A是根据本申请说明书一些实施例提供的第一麦克风阵列和第二麦克风阵列的排布方式和位置关系的示例性分布图。如图4A所示,第一麦克风阵列130以半圆形排布的排布方式设置于人耳处,第二麦克风阵列160也是以半圆形排布的排布方式设置于人耳处,第二麦克风阵列160中的麦克风比第一麦克风阵列130中 任意麦克风更加靠近用户耳道。在一些实施例中,第一麦克风阵列130中的麦克风可以与第二麦克风阵列160中的麦克风独立设置。例如,图4A中的第一麦克风阵列130中的麦克风以半圆形的排布方式进行排布,第二麦克风阵列160中的麦克风以半圆形的排布方式进行排布,第一麦克风阵列130中的麦克风与第二麦克风阵列160中的麦克风未出现重叠或交叉。在一些实施例中,第一麦克风阵列130中的麦克风可以与第二麦克风阵列160中的麦克风可以部分重叠或交叉。
在一些实施例中,第一麦克风阵列130和第二麦克风阵列160的排布方式可以不同。图4B是根据本申请说明书另一些实施例提供的第一麦克风阵列和第二麦克风阵列的排布方式的示例性分布图。如图4B所示,第一麦克风阵列130以半圆形排布的排布方式设置于人耳处,第二麦克风阵列160以线形排布的排布方式是设置于人耳处,其中,第二麦克风阵列160中的麦克风相对于第一麦克风阵列130中任意麦克风更加靠近用户耳道。在一些实施例中,第一麦克风阵列130和第二麦克风阵列160还可以为组合排布的方式进行排列。例如,图4B中第二麦克风阵列160包括线形排布的部分和半圆形排布的部分,第二麦克风阵列160中半圆形排布的部分为第一麦克风阵列130的组成部分。需要说明的是,第一麦克风阵列130和第二麦克风阵列160的排布方式不限于图4A和图4B中所示的半圆形和线形,这里的半圆形和线形只出于说明的目的,关于麦克风阵列的排布方式可以参见本说明书图8及其相关描述。
在一些实施例中,还可以根据用户的手动输入更新降噪信号。例如,在一些实施例中,用户在比较嘈杂的外界环境中佩戴开放式耳机100进行音乐播放时,用户自身的听觉体验效果不理想,用户可以根据自身的听觉效果手动调整降噪信号的参数信息(例如,频率信息、相位信息或者幅值信息)。又例如,特殊用户(例如,听力受损用户或者年龄较大用户)在使用开放式耳机100的过程中,特殊用户的听力能力与普通用户的听力能力存在差异,开放式耳机100本身生成的降噪声波与特殊人群的听力能力不匹配,导致特殊用户的听觉体验较差。这种情况下,特殊用户可以根据自身的听觉效果手动调整降噪信号的频率信息、相位信息或者幅值信息,从而更新降噪信号以提高特殊用户的听觉体验。在一些实施例中,用户手动调整降噪信号的方式可以是通过开放式耳机100上的键位进行手动调整。在一些实施例中,用户手动调整降噪信号的方式也可以是通过终端设备进行手动输入调整。在一些实施例中,开放式耳机100或者与开放式耳机100通信连接的手机、平板电脑、电脑等电子产品上可以显示用户耳道处的声场,并反馈给用户建议的降噪信号的频率信息范围、幅值信息范围或相位信息范围,用户可以 根据建议的降噪信号的参数信息进行手动输入,然后再根据自身的听觉体验情况进行参数信息的微调。
在一些实施例中,信号处理器140基于拾取的环境噪声估计第一空间的噪声可以包括:根据拾取的环境噪声进行信号分离,获取环境噪声对应的参数信息,基于环境噪声对应的参数信息生成降噪信号。在一些实施例中,麦克风阵列(例如,第一麦克风阵列130、第二麦克风阵列160)拾取的环境噪声可以包括噪声、用户人声、扬声器阵列150输出的音频等。在一些实施例中,扬声器阵列150输出的音频可以包括扬声器阵列150输出的通话远端音频、设备媒体音频、降噪声波等。在一些实施例中,信号处理器140可以对麦克风阵列拾取的环境噪声进行信号分析,将环境噪声所包括的各种声音信号进行信号分离,得到噪声、用户人声、降噪声波、设备媒体音频、通话远端音频等多种单一的声音信号。具体地,信号处理器140可以根据噪声、用户人声、降噪声波、设备媒体音频、通话远端音频等在空间、时域、频域等不同维度的统计分布特性及结构化特征,自适应调整滤波器组参数,估计环境噪声中各个声音信号(例如,噪声、用户人声、降噪声波、设备媒体音频、通话远端音频等)的参数信息,并根据不同的参数信息完成信号分离过程。例如,在一些实施例中,麦克风阵列可以将拾取的噪声、用户人声、降噪声波分别转换成对应的第一信号、第二信号、第三信号。信号处理器140获取第一信号、第二信号、第三信号在空间差异(例如,信号所处位置)、时域差异(例如,延迟)、频域差异(例如,幅值、相位),并根据三种维度上的差异将第一信号、第二信号、第三信号进行信号分离,得到相对纯净的第一信号、第二信号、第三信号。分离后的第一信号、第二信号、第三信号分别对应纯净的噪声、用户人声、降噪声波,信号处理器140完成信号分离过程。在一些实施例中,信号处理器140可以根据信号分离得到的噪声、降噪声波、设备媒体音频、通话远端音频等的参数信息更新降噪声波,更新后的降噪声波通过扬声器阵列150输出。
在一些实施例中,噪声的结构化特征可以包括噪声分布、噪声强度、全局噪声强度、噪声率等,或其任意组合。在一些实施例中,噪声强度可以指噪声像素的值,反映噪声像素中的噪声幅度,因此,噪声分布可以反映图像中具有不同噪声强度的噪声的概率密度。全局噪声强度可以反映图像中的平均噪声强度或加权平均噪声强度。噪声率可以反映出噪声分布的分散程度。在一些实施例中,噪声的统计分布特性可以包括但不限于概率分布密度、功率谱密度、自相关函数、概率密度函数、方差、数学期望等。在一些实施例中,通过信号分离得到的用户人声、设备媒体音频、通话远端音频等也可以 传输至通话远端。例如,用户佩戴开放式耳机100进行语音通话时,用户人声可以传输至通话远端。
图5是根据本申请说明书一些实施例提供的估计第一空间位置的噪声的示例性流程图。如图5所示,流程500可以包括:
在步骤510中,确定一个或多个与拾取的环境噪声有关的空间噪声源。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,与环境噪声有关的空间噪声源是指其声波可传递至用户耳道处或靠近用户耳道处(例如,第一空间位置)的噪声源。在一些实施例中,空间噪声源可以为用户身体不同方向(例如,前方、后方等)的噪声源。例如,用户身体前方存在人群喧闹噪声,用户身体左方存在车辆鸣笛噪声,这种情况下,空间噪声源即为用户身体前方的人群喧闹噪声源和用户身体左方的车辆鸣笛噪声源。在一些实施例中,第一麦克风阵列130可以拾取用户身体各个方向的空间噪声,并将空间噪声转化为电信号传递至信号处理器140,信号处理器140可以将空间噪声对应的电信号进行信号分析,得到各个方向的空间噪声的参数信息(例如,方位信息、幅值信息、相位信息等)。信号处理器140根据各个方向的空间噪声的参数信息确定各个方向的空间噪声源,例如,空间噪声源的方位、空间噪声源的相位以及空间噪声源的幅值等。在一些实施例中,信号处理器140可以通过噪声定位算法确定空间噪声源。在一些实施例中,噪声定位算法可以包括波束形成、超分辨空间谱估计、到达时差等中的一种或多种。在一些实施例中,信号处理器140可以将拾取的环境噪声按照特定的频带宽度(例如,每500Hz作为一个频带)划分为多个子带,每个子带可以分别对应不同的频率范围,并在至少一个子带上确定与该子带对应的空间噪声源。关于空间噪声源的定位方法具体可以参考本说明书其他地方,在此不做赘述。关于确定一个或多个与拾取的环境噪声有关的空间噪声源的详细描述可以参考本说明书图6及其相关描述。
在步骤520中,基于空间噪声源,估计第一空间位置的噪声。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,信号处理器140可以基于步骤510中得到的位于用户身体各个方向的空间噪声源的参数信息(例如,频率信息、幅值信息、相位信息等),估计各个空间噪声源分别传递至第一空间位置的噪声的参数信息,从而估计出第一空间位置的噪声。例如,在一些实施例中,用户身体前方和后方分别有一个空间噪声源,信号处理器140可以根据前方空间噪声源的频率信息、相位信息或幅值信息,估计前方空间噪声源传递到第一空间位置时,前方 空间噪声源的频率信息、相位信息或幅值信息。信号处理器140根据后方空间噪声源的频率信息、相位信息或幅值信息,估计后方空间噪声源传递到第一空间位置时,后方空间噪声源的频率信息、相位信息或幅值信息。信号处理器140基于前方空间噪声源的频率信息、相位信息或幅值信息和后方空间噪声源的频率信息、相位信息或幅值信息,估计第一空间位置的噪声信息,从而估计第一空间位置的噪声。在一些实施例中,在一些实施例中,可以通过特征提取的方法从麦克风阵列拾取的空间噪声源的频率响应曲线提取空间噪声源的参数信息。在一些实施例中,提取空间噪声源的参数信息的方法可以包括但不限于主成分分析(Principal Components Analysis,PCA)、独立成分分析(Independent Component Algorithm,ICA)、线性判别分析(Linear Discriminant Analysis,LDA)、奇异值分解(Singular Value Decomposition,SVD)等。
在一些实施例中,确定一个或多个与拾取的环境噪声有关的空间噪声源可以包括通过波束形成、超分辨空间谱估计或到达时差中的一种或多种方式定位一个或多个空间噪声源。波束形成定位方式是一种基于最大输出功率的可控波束形成的声源定位方法。在一些实施例中,波束形成声源定位方法可以将麦克风阵列中的各个麦克风阵元拾取的声音信号进行加权求和形成波束,通过搜索空间噪声源的可能位置来引导该波束,修改权值使得麦克风阵列的输出信号功率最大。需要说明的是,波束形成声源定位方法既可以在时域中使用,也可以在频域中使用。波束形成在时域中的时间平移等价于在频域中的相位延迟。在一些实施例中,超分辨空间谱估计的声源定位方法可以包括自回归AR模型、最小方差谱估计(MV)和特征值分解方法(例如,Music算法)等,这些方法都可以通过获取麦克风阵列的声音信号来计算空间谱的相关矩阵,并对空间噪声源的方向进行有效估计。在一些实施例中,到达时差声源定位方法可以先进行声达时间差估计,并从中获取麦克风阵列中阵元间的声延迟(TDOA),再利用获取的声达时间差,结合已知的麦克风阵列的空间位置进一步定位出空间噪声源的位置。
为了更加清楚的说明空间噪声源的定位原理,下面以波束形成声源定位方法为例具体说明空间噪声源的定位是如何实现的。以麦克风阵列为直线形阵列作为示例,空间噪声源可以为远场声源,此时认为空间噪声源入射到麦克风阵列的入射声波是平行的。在平行的声场中,空间噪声源入射声波的入射角度与麦克风阵列(例如,第一麦克风阵列130或第二麦克风阵列160)中的麦克风平面垂直时,入射声波可以同时达到麦克风阵列(例如,第一麦克风阵列130或第二麦克风阵列160)中的各个麦克风。在一些实施例中,平行声场中的空间噪声源入射声波的入射角度与麦克风阵列(例如,第一麦克 风阵列130或第二麦克风阵列160)中的麦克风平面不垂直时,入射声波到达麦克风阵列(例如,第一麦克风阵列130或第二麦克风阵列160)中的每个麦克风会有延时,该延时可以由入射角度决定。在一些实施例中,不同的入射角度,叠加之后的噪声波形强度是不一样的。例如,入射角度为0°时,噪声信号强度较弱,入射角度为45°时,噪声信号强度最强。入射角度不同时,噪声波形叠加后的波形叠加强度不同,由此使得麦克风阵列具有极性,从而可以得到麦克风阵列的极性图。在一些实施例中,麦克风阵列(例如,第一麦克风阵列130或第二麦克风阵列160)可以是一个方向阵,该方向阵的指向性可以通过时域算法或频域相位延迟算法实现,例如,延迟、叠加等。在一些实施例中,通过控制不同的延迟,可以实现不同方向的指向。在一些实施例中,方向阵指向可控相当于一个空间滤波器,先把噪声定位区域进行网格划分,再通过各个网格点的延迟时间对各个麦克风进行时域延迟,最终将各个麦克风的时域延迟叠加起来,计算得到每个网格的声压,从而得到每个网格的相对声压,最终实现空间噪声源定位。
应当注意的是,上述有关流程500的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程500进行各种修正和改变。例如,流程500还可以包括对空间噪声源进行定位,提取空间噪声源的参数信息等。又例如,步骤510和步骤520可以合并为一个步骤。然而,这些修正和改变仍在本说明书的范围之内。
图6是根据本申请说明书一些实施例提供的确定空间噪声源的示例性流程图。如图6所示,流程600可以包括:
在步骤610中,将拾取的环境噪声划分为多个子带,每个子带对应不同的频率范围。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,来自用户身体不同方向的环境噪声的频率可以是不同的,信号处理器140在对环境噪声信号进行信号处理时,可以将环境噪声频带划分为多个子带,每个子带对应不同的频率范围。这里每个子带对应的频率范围可以是预先设定好的频率范围,例如,80Hz-100Hz、100Hz-300Hz、300Hz-800Hz等。在一些实施例中,每个子带中都包含了对应频段的环境噪声的参数信息。例如,信号处理器140可以将拾取的环境噪声划分为80Hz-100Hz、100Hz-300Hz、300Hz-800Hz、800Hz-1000Hz四个子带,这四个子带中分别对应80Hz-100Hz、100Hz-300Hz、300Hz-800Hz、800Hz-1000Hz的环境噪声的参数。
在步骤620中,在至少一个子带上,确定与其对应的空间噪声源。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,信号处理器140可以对环境噪声划分的子带进行信号分析,得到每个子带对应的环境噪声的参数信息,并根据参数信息确定与每个子带对应的空间噪声源。例如,在300Hz-800Hz这个子带上,信号处理器140可以获取该子带中包含的对应的环境噪声的参数信息(例如,频率信息、幅值信息、相位信息等),信号处理器140根据获取的参数信息确定与300Hz-800Hz这个子带对应的空间噪声源。
应当注意的是,上述有关流程600的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程600进行各种修正和改变。例如,将步骤610和步骤620进行合并。又例如,在流程600中增加其他步骤。然而,这些修正和改变仍在本说明书的范围之内。
图7是根据本申请说明书一些实施例提供的确定空间噪声源的另一示例性流程图。如图7所示,流程700可以包括:
在步骤710中,获取用户头函数,用户头函数反映用户头部对声音的反射或吸收情况。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,第一麦克风阵列130可以包括第一子麦克风阵列和第二子麦克风阵列,第一子麦克风阵列和第二子麦克风阵列分别位于用户的左耳和右耳处。在一些实施例中,第一麦克风阵列130可以是双边模式排布,即第一子麦克风阵列和第二子麦克风阵列同时启用的双边模式排布。在一些实施例中,第一子麦克风阵列位于用户左耳位置,第二子麦克风阵列位于用户右耳位置的双边模式排布时,在声音信号的传递过程中,用户头部会对声音信号进行反射或吸收,致使第一子麦克风阵列和第二子麦克风阵列对同一环境噪声的拾取有所差异。在一些实施例中,信号处理器140可以基于第一子麦克风阵列拾取的环境噪声的参数信息与第二子麦克风阵列拾取的同一环境噪声的参数信息之间的差异,构建用户头函数,该用头函数可以反映用户头部对声音的反射和吸收情况。
在步骤720中,在至少一个子带上,结合第一子麦克风阵列拾取的环境噪声,第二子麦克风阵列拾取的环境噪声,以及用户头函数,确定与其对应的空间噪声源。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,第一麦克风阵列130在双边模式下,第一子麦克风阵列拾取的环境噪声信号的幅度信息和相位信息与第二子麦克风阵列拾取的环境噪声信号的幅度信息和相位信息之间存在幅值差和相位差。信号处理器140可以根据第一子麦克风阵列拾取的环境噪声、第二子麦克 风阵列拾取的环境噪声,以及步骤710中信号处理器140获取的用户头函数,在环境噪声的至少一个子带上进行频点合成,即将头函数作为先验信息,在环境噪声的至少一个子带上将第一子麦克风阵列拾取的对应子带上的环境噪声的频点与第二子麦克风阵列拾取的对应子带上的环境噪声的频点进行合成。完成频点合成后的子带中包含的参数信息对应于重构的虚拟噪声源的参数信息。信号处理器140基于重构的虚拟声源的参数信息确定空间噪声源,进而完成空间噪声源定位。
在一些实施例中,第一麦克风阵列130也可以是单边模式排布。例如,只启用第一子麦克风阵列或者第二子麦克风阵列。在一些实施例中,第一麦克风阵列130为单边模式排布,启用位于用户左耳处的第一子麦克风阵列时,信号处理器140可以将用户头函数作为先验信息,在环境噪声的至少一个子带上将第一子麦克风阵列拾取的对应子带上的环境噪声的频点进行合成。完成频点合成后的子带中包含的参数信息对应重构的虚拟噪声源参数信息。信号处理器140基于重构的虚拟声源的参数信息确定空间噪声源,进而完成空间噪声源定位。
在一些实施例中,第一子麦克风阵列可以拾取到达用户左耳处的环境噪声,信号处理器140还可以基于该环境噪声参数信息通过用户头函数估计出该环境噪声到达用户右耳处时的参数信息。信号处理器140根据估计的环境噪声到达用户右耳处时的参数信息,由此可以更加精准的完成空间噪声源定位。在一些实施例中,第一麦克风阵列130的单边模式也可以是只设置一个子麦克风阵列,关于此类单边模式下进行空间噪声源定位过程与只启用第一子麦克风阵列(或第二子麦克风阵列)的单边模式空间噪声源定位过程类似,在此不做赘述。
应当注意的是,上述有关流程700的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程300进行各种修正和改变。然而,这些修正和改变仍在本说明书的范围之内。
在一些实施例中,第一子麦克风阵列或者第二子麦克风阵列的排布方式可以是规则几何形状的阵列。图8A是根据本申请说明书一些实施例提供的第一子麦克风阵列的排布方式的示意图。如图8A所示,第一子麦克风阵列成线形阵列。在一些实施例中,第一子麦克风阵列或者第二子麦克风阵列的排布方式也可以是其他形状的阵列,例如,图8B是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图。如图8B所示,第一子麦克风阵列成十字形阵列。又例如,图8C是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图。如图8C所示,第一子麦 克风阵列成圆形阵列。需要说明的是,第一子麦克风阵列或者第二子麦克风阵列的排布方式不限于图8A、图8B、图8C所示的线形阵列、十字形阵列、圆形阵列,也可以是其他形状的阵列图形,例如,三角形阵列、螺旋形阵列、平面阵列、立体阵列等,本说明书对此不做限定。需要说明的是,图8A-8D中的每一条短实线可以视为一个麦克风或一组麦克风。在一些实施例中,每一条短实线为一组麦克风时,每组麦克风的数量可以相同或不同,每组麦克风的种类可以相同或不同,每组麦克风的朝向可以相同或不同,关于麦克风中种类、数量、朝向以及间距可以根据实际应用情况进行适应性调整。
在一些实施例中,第一子麦克风阵列或者第二子麦克风阵列的排布方式也可以是不规则几何形状的阵列。例如,图8D是根据本申请说明书另一些实施例提供的第一子麦克风阵列的排布方式的示意图。如图8D所示,第一子麦克风阵列成不规则阵列。需要说明的是,第一子麦克风阵列或者第二子麦克风阵列的不规则形状阵列排布不限于图8D中所示的形状,还可以是其他形状的不规则图形等阵列排布,例如,不规则多边形等,本说明书对此不做限定。
在一些实施例中,第一子麦克风阵列(或第二子麦克风阵列)中的麦克风之间可以是均匀分布,这里的均匀分布是指第一子麦克风阵列(或第二子麦克风阵列)中的麦克风之间的间距相同。在一些实施例中,第一子麦克风阵列(或第二子麦克风阵列)中的麦克风也可以是非均匀分布,这里的非均匀分布是指第一子麦克风阵列(或第二子麦克风阵列)中的麦克风之间的间距不同。关于子麦克风阵列中的麦克风阵元之间的间距可以根据实际情况做适应性调整,本说明书对此不做限定。
图9A是根据本申请说明书一些实施例提供的第一子麦克风阵列和第二子麦克风阵列的位置关系的示意图。如图9A所示,第一子麦克风阵列911位于用户左耳处,第一子麦克风阵列911呈近似三角形排布。第二子麦克风阵列912位于用户右耳处,第二子麦克风阵列912也呈近似三角形排布,第二子麦克风阵列912与第一子麦克风阵列911的排布方式相同且关于用户头部对称分布。参照图9A,第一子麦克风阵列911沿阵列方向的延长线与第二子麦克风阵列912沿阵列方向的延长线相交,可以构成四边形结构。
图9B是根据本申请说明书另一些实施例提供的第一子麦克风阵列和第二子麦克风阵列的位置关系的示意图。如图9B所示,第一子麦克风阵列921位于用户左耳处,第一子麦克风阵列921呈线形排布。第二子麦克风阵列922位于用户右耳处,第二子麦克风阵列922呈近似三角形排布,第二子麦克风阵列922与第一子麦克风阵列921的排 布方式不同且关于用户头部非对称分布。参照图9B,第一子麦克风阵列921沿阵列方向的延长线与第二子麦克风阵列922沿阵列方向的延长线相交,可以构成三角形结构。
在一些实施例中,第一子麦克风阵列921和第二子麦克风阵列922除了可以构成图9A中所示的四边形、图9B中所示的三角形外,还可以构成八字形、圆形、椭圆形、环形、多边形等规则和/或不规则形状。第一子麦克风阵列和第二子麦克风阵列以特定的形状或立体空间分布,可以全方位的获取用户各个方向的环境噪声,通过各麦克风获取的环境噪声的参数信息可以对空间噪声源进行更加精准的定位,进而更加精准地模拟处用户耳道处的噪声声场,以达到更好的降噪效果。第一子麦克风阵列和第二子麦克风阵列不同的排布方式具有不同的空域滤波性能。仅作为示例性说明,空域滤波性能可以包括主瓣宽度、旁瓣(也被称为副瓣)宽度。主瓣宽度是指声波辐射的最大辐射波束。旁瓣宽度是指除最大辐射波束之外的辐射波束。其中,主瓣宽度越窄,麦克风阵列分辨率越高、指向性越好。旁瓣高度越低,麦克风阵列抗干扰性能越好,旁瓣高度越高,麦克风阵列抗干扰性能越差。例如,十字形阵列的波束图对应的主瓣宽度比圆形、矩形或螺旋形的波阵图对应的主瓣宽度要窄,也就是说同样阵元个数的条件下,十字形阵列具有较高的空间分辨率和更好指向性。而从旁瓣高度来看,十字形阵列的波束图对应的旁瓣宽度比圆形、矩形或螺旋形的波阵图对应的旁瓣宽度高,也就是说,十字形阵列的抗干扰能力较差。关于第一子麦克风阵列和第二子麦克风阵列的排布方式可以根据实际应用情况进行适应性调整,在此不做进一步限定。需要说明的是,图9A和图9B中的每一条短实线可以视为一个麦克风或一组麦克风。在一些实施例中,每一条短实线为一组麦克风时,每组麦克风的数量可以相同或不同,每组麦克风的种类可以相同或不同,每组麦克风的朝向可以相同或不同,关于麦克风中种类、数量、朝向以及间距可以根据实际应用情况进行适应性调整。在一些实施例中,还可以通过合成孔径、稀疏恢复、互素阵列等方法形成环境噪声的空间超分辨图像,该空间超分辨图像可以用于反映环境噪声的信号反射图,以进一步提高空间噪声源的定位精度。在一些实施例中,基于空间噪声源的定位精度的反馈情况可以调整麦克风阵列(例如,第一麦克风阵列130、第二麦克风阵列160)的中麦克风的位置、间距、启闭状态等。
在一些实施例中,第一麦克风阵列130可以包括一个噪声麦克风,第一麦克风阵列130中的噪声麦克风用于拾取用户耳道处的空间噪声,而噪声麦克风拾取用户耳道处的空间噪声时,也会拾取扬声器阵列150输出的降噪声波,该降噪声波是不期望被噪声麦克风拾取的。因此,可以将噪声麦克风设置于扬声器阵列150中形成的声学偶极子 的声学零点处,以使得噪声麦克风拾取的降噪声波最小。在一些实施例中,至少一个扬声器阵列150形成至少一组声学偶极子,且噪声麦克风位于偶极子辐射声场的声学零点处。在一些实施例中,扬声器阵列150中的任意两个扬声器输出的声音信号可以看作是两个向外辐射声音的点声源,其辐射声音的幅值相同,相位相反。所述两个扬声器可以构成声学偶极子或类似声学偶极子,向外辐射声音具有明显的指向性,形成一个“8”字形声音辐射区域。在所述两个扬声器连线所在的直线方向,扬声器处辐射的声音最大,其余方向辐射声音明显减小,两个扬声器连线的中垂线处辐射的声音最小。在一些实施例中,扬声器阵列150中的一个扬声器输出的声音信号也可以视为一个偶极子。例如,扬声器阵列150中的一个扬声器振膜正面和振膜背面输出的一组相位近似相反、幅值近似相同的声音信号可以视为两个点声源。
在一些实施例中,还可以通过算法获取麦克风阵列(例如,第一麦克风阵列130)在声学零点位置处拾取的环境噪声信号。例如,在一些实施例中,可以预先将第一麦克风阵列130中的一个或多个麦克风设置于特定频带的扬声器阵列150构成的声学偶极子的声学零点位置。特定频带可以是对语音可懂度起关键作用的频带,例如,500Hz-1500Hz。在一些实施例中,信号处理器140根据声学偶极子位置(即,构成声学偶极子的两个扬声器的位置)和声传递函数,计算特定频带的补偿参数并预先存储。信号处理器140可以根据预先存储的补偿参数对第一麦克风阵列130中的其余麦克风(即,未设置在声学零点位置的麦克风)拾取的环境噪声基进行幅值补偿和/或相位补偿,补偿后的环境噪声信号等效于设置在声学零点位置噪声麦克风拾取的环境噪声信号。需要注意的是,麦克风阵列(例如,第一麦克风阵列130)中的麦克风还可以不设置于扬声器阵列150构成的声学偶极子的声学零点处,例如,在一些实施例中,信号处理器140可以根据不同类型的噪声在不同维度(例如,空域、时域、频域等)的统计分布和结构化特征将麦克风阵列拾取的第一空间位置的噪声进行信号分离提取,从而获取不同类型(例如不同频率、不同相位等)的噪声,并通过信号处理器将麦克风阵列拾取的扬声器阵列150发出的降噪声波进行消除。
图10是根据本申请说明书一些实施例提供的估计第一空间位置的噪声的示例性流程图。如图10所示,流程1000可以包括:
在步骤1010中,从拾取的环境噪声中去除与骨传导麦克风拾取的信号相关联的成分,以便更新环境噪声。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,麦克 风阵列(例如,第一麦克风阵列130、第二麦克风阵列160)在拾取环境噪声时,用户自身的说话声音也会被麦克风阵列拾取,即,用户自身说话的声音也被视为环境噪声的一部分。这种情况下,扬声器阵列150输出的降噪声波会将用户自身说话的声音抵消。在一些实施例中,特定场景下,用户自身说话的声音需要被保留,例如,用户进行语音通话、发送语音消息等场景中。在一些实施例中,用户佩戴开放式耳机100进行语音通话或录制语音信息时,骨传导麦克风可以通过拾取用户说话时面部骨骼或肌肉产生的振动信号来拾取用户说话的声音信号,并传递至信号处理器140。信号处理器140获取来自骨传导麦克风拾取的声音信号的参数信息,信号处理器140从麦克风阵列(例如,第一麦克风阵列130、第二麦克风阵列160)拾取的环境噪声中找到并去除与骨传导麦克风拾取的声音信号相关联的声音信号成分。信号处理器140根据剩余的麦克风阵列拾取的环境噪声的参数信息更新环境噪声。更新后的环境噪声中不再包含用户自身说话的声音信号,即在用户进行语音通话时保留了用户自身说话的声音信号。
在步骤1020中,根据更新后的环境噪声估计第一空间位置的噪声。
在一些实施例中,该步骤可以由信号处理器140执行。在一些实施例中,信号处理器140可以根据更新后的环境噪声估计第一空间位置的噪声。关于根据环境噪声估计第一空间位置的噪声的详细描述可以参考本说明书图2及其相关描述,在此不做赘述。
应当注意的是,上述有关流程1000的描述仅仅是为了示例和说明,而不限定本说明书的适用范围。对于本领域技术人员来说,在本说明书的指导下可以对流程1000进行各种修正和改变。例如,还可以对骨传导麦克风拾取的信号相关联的成分进行预处理,并将骨传导麦克风拾取的信号作为音频信号传输至终端设备。然而,这些修正和改变仍在本说明书的范围之内。
在一些实施例中,至少一个麦克风阵列可以包括骨传导麦克风和气传导麦克风,信号处理器140可以基于开放式耳机100的工作状态控制骨传导麦克风和气传导麦克风的开关状态。在一些实施例中,开放式耳机100的工作状态可以是指用户佩戴开放式耳机100时所使用的用途状态。在一些实施例中,开放式耳机100的工作状态可以包括但不限于音乐播放状态、语音通话状态、语音发送状态等。在一些实施例中,麦克风阵列拾取环境噪声时,麦克风阵列中的骨传导麦克风的开关状态和气传导麦克风的开关状态可以根据开放式耳机100的工作状态决定。例如,用户佩戴开放式耳机100进行音乐播放时,骨传导麦克风的开关状态可以为待机状态,气传导麦克风的开关状态可以为工 作状态。又例如,用户佩戴开放式耳机100进行语音发送时,骨传导麦克风的开关状态可以为工作状态,气传导麦克风的开关状态可以为工作状态。在一些实施例中,信号处理器140耦接麦克风阵列,信号处理器140可以通过发送控制信号控制麦克风阵列中的麦克风(例如,骨传导麦克风、气传导麦克风)的开关状态。
在一些实施例中,开放式耳机100的工作状态可以包括通话状态和未通话状态。在一些实施例中,开放式耳机100的工作状态为未通话状态时,信号处理器140可以控制骨传导麦克风为待机状态。例如,开放式耳机100在未通话状态下,用户自身说话的声音信号可以视为环境噪声,这种情况下,麦克风阵列拾取的环境噪声中包括的用户自身说话的声音信号可以不被滤除,从而使得用户自身说话的声音信号也可以与扬声器阵列150输出的降噪声波相抵消。
在一些实施例中,开放式耳机100的工作状态为通话状态时,信号处理器140可以控制骨传导麦克风为工作状态。例如,开放式耳机100在通话状态下,用户自身说话的声音信号需要保留,这种情况下,信号处理器140可以发送控制信号控制骨传导麦克风为工作状态,骨传导麦克风拾取用户说话的声音信号,信号处理器140从麦克风阵列拾取的环境噪声中找到并去除与骨传导麦克风拾取的声音信号相关联的声音信号成分,以使用户自身说话的声音信号不与扬声器阵列150输出的降噪声波相抵消,从而保证用户正常的通话状态。
在一些实施例中,开放式耳机100的工作状态为通话状态时,若环境噪声的声压级大于预设阈值时,信号处理器140可以控制骨传导麦克风保持工作状态。在一些实施例中,环境噪声的声压级可以反映环境噪声的强度。这里的预设阈值可以是预先存储在开放式耳机100中的数值,例如,50dB、60dB或70dB等其它任意数值。在一些实施例中,当环境噪声的声压级大于预设阈值时,环境噪声会影响用户的通话质量。信号处理器140可以通过发送控制信号控制骨传导麦克风保持工作状态,骨传导麦克风可以获取用户讲话时的面部肌肉的振动信号,而基本不会拾取外部环境噪声,此时将骨传导麦克风拾取的振动信号作为通话时的语音信号,从而保证用户的正常通话。
在一些实施例中,开放式耳机100的工作状态为通话状态时,若环境噪声的声压级小于预设阈值时,信号处理器140可以控制骨传导麦克风由工作状态切换至待机状态。在一些实施例中,当环境噪声的声压级小于预设阈值时,环境噪声的声压级相对于用户说话产生的声音信号的声压级较小,用户说话产生的声音信号被扬声器阵列150输出的降噪声波抵消一部分后,剩余的用户说话产生的声音信号仍然可以达到通话标准, 足以保证用户的正常通话。这种情况下,信号处理器140可以通过发送控制信号控制骨传导麦克风由工作状态切换至待机状态,进而降低信号处理复杂度以及开放式耳机100的功率损耗。
在一些实施例中,开放式耳机100还可以包括用于调整降噪声波声压级的调节模块。在一些实施例中,调节模块可以包括按钮、语音助手、手势传感器等。用户通过控制调节模块可以调整开放式耳机100的降噪模式。具体地,用户通过控制调节模块可以调整(例如,放大或缩小)降噪信号的幅值信息,以改变扬声器阵列发出的降噪声波的声压级,进而达到不同的降噪效果。仅作为示例性说明,在一些实施例中,降噪模式可以包括强降噪模式、中级降噪模式、弱降噪模式等。例如,用户在室内佩戴开放式耳机100时,外界环境噪声较小,用户可以通过调节模块将开放式耳机的降噪模式关闭或调整为弱降噪模式。又例如,当用户在街边等公共场合行走时佩戴开放式耳机100,用户需要在收听音频信号(例如,音乐、语音信息)的同时,保持对周围环境的一定感知能力,以应对突发状况,此时用户可以通过调节模块(例如,按钮或语音助手)选择中级降噪模式,以保留周围环境噪声(如警报声、撞击声、汽车鸣笛声等)。再例如,用户在乘坐地铁或飞机等交通工具时,用户可以通过调节模块选择强降噪模式,以进一步降低周围环境噪声。在一些实施例中,信号处理器140还可以基于环境噪声强度范围向开放式耳机100或与开放式耳机100通信连接的终端设备(例如,手机、智能手表等)发出提示信息,以提醒用户调整降噪模式。
上文已对基本概念做了描述,显然,对于本领域技术人员来说,上述详细披露仅仅作为示例,而并不构成对本申请的限定。虽然此处并没有明确说明,本领域技术人员可能会对本申请进行各种修改、改进和修正。该类修改、改进和修正在本申请中被建议,所以该类修改、改进、修正仍属于本申请示范实施例的精神和范围。
同时,本申请使用了特定词语来描述本申请的实施例。如“一个实施例”、“一实施例”、和/或“一些实施例”意指与本申请至少一个实施例相关的某一特征、结构或特点。因此,应强调并注意的是,本说明书中在不同位置两次或多次提及的“一实施例”或“一个实施例”或“一个替代性实施例”并不一定是指同一实施例。此外,本申请的一个或多个实施例中的某些特征、结构或特点可以进行适当的组合。
此外,本领域技术人员可以理解,本申请的各方面可以通过若干具有可专利性的种类或情况进行说明和描述,包括任何新的和有用的工序、机器、产品或物质的组合,或对他们的任何新的和有用的改进。相应地,本申请的各个方面可以完全由硬件执行、 可以完全由软件(包括固件、常驻软件、微码等)执行、也可以由硬件和软件组合执行。以上硬件或软件均可被称为“数据块”、“模块”、“引擎”、“单元”、“组件”或“***”。此外,本申请的各方面可能表现为位于一个或多个计算机可读介质中的计算机产品,该产品包括计算机可读程序编码。
计算机存储介质可能包含一个内含有计算机程序编码的传播数据信号,例如在基带上或作为载波的一部分。该传播信号可能有多种表现形式,包括电磁形式、光形式等,或合适的组合形式。计算机存储介质可以是除计算机可读存储介质之外的任何计算机可读介质,该介质可以通过连接至一个指令执行***、装置或设备以实现通讯、传播或传输供使用的程序。位于计算机存储介质上的程序编码可以通过任何合适的介质进行传播,包括无线电、电缆、光纤电缆、RF、或类似介质,或任何上述介质的组合。
此外,除非权利要求中明确说明,本申请所述处理元素和序列的顺序、数字字母的使用、或其他名称的使用,并非用于限定本申请流程和方法的顺序。尽管上述披露中通过各种示例讨论了一些目前认为有用的发明实施例,但应当理解的是,该类细节仅起到说明的目的,附加的权利要求并不仅限于披露的实施例,相反,权利要求旨在覆盖所有符合本申请实施例实质和范围的修正和等价组合。例如,虽然以上所描述的***组件可以通过硬件设备实现,但是也可以只通过软件的解决方案得以实现,如在现有的服务器或移动设备上安装所描述的***。
同理,应当注意的是,为了简化本申请披露的表述,从而帮助对一个或多个发明实施例的理解,前文对本申请实施例的描述中,有时会将多种特征归并至一个实施例、附图或对其的描述中。但是,这种披露方法并不意味着本申请对象所需要的特征比权利要求中提及的特征多。实际上,实施例的特征要少于上述披露的单个实施例的全部特征。
一些实施例中使用了描述成分、属性数量的数字,应当理解的是,此类用于实施例描述的数字,在一些示例中使用了修饰词“大约”、“近似”或“大体上”来修饰。除非另外说明,“大约”、“近似”或“大体上”表明所述数字允许有±20%的变化。相应地,在一些实施例中,说明书和权利要求中使用的数值参数均为近似值,该近似值根据个别实施例所需特点可以发生改变。在一些实施例中,数值参数应考虑规定的有效数位并采用一般位数保留的方法。尽管本申请一些实施例中用于确认其范围广度的数值域和参数为近似值,在具体实施例中,此类数值的设定在可行范围内尽可能精确。
针对本申请引用的每个专利、专利申请、专利申请公开物和其他材料,如文章、书籍、说明书、出版物、文档等,特此将其全部内容并入本申请作为参考。与本申请内 容不一致或产生冲突的申请历史文件除外,对本申请权利要求最广范围有限制的文件(当前或之后附加于本申请中的)也除外。需要说明的是,如果本申请附属材料中的描述、定义、和/或术语的使用与本申请所述内容有不一致或冲突的地方,以本申请的描述、定义和/或术语的使用为准。
最后,应当理解的是,本申请中所述实施例仅用以说明本申请实施例的原则。其他的变形也可能属于本申请的范围。因此,作为示例而非限制,本申请实施例的替代配置可视为与本申请的教导一致。相应地,本申请的实施例不仅限于本申请明确介绍和描述的实施例。

Claims (15)

  1. 一种开放式耳机,其特征在于,包括:
    固定结构,被配置为将所述耳机固定在用户耳朵附近且不堵塞用户耳道的位置;
    壳体结构,被配置为承载:
    第一麦克风阵列,被配置为拾取环境噪声;
    至少一个扬声器阵列;以及
    信号处理器,被配置为:基于所述拾取的环境噪声估计第一空间位置的噪声,所述第一空间位置比所述第一麦克风阵列中任一麦克风更加靠近用户耳道;以及
    基于所述第一空间位置的噪声生成降噪信号,使得所述至少一个扬声器阵列根据所述降噪信号输出降噪声波,所述降噪声波用于消除传递到用户耳道的环境噪声。
  2. 根据权利要求1所述的开放式耳机,其特征在于,
    所述壳体结构被配置为容纳第二麦克风阵列,所述第二麦克风阵列被配置为拾取环境噪声和所述降噪声波,所述第二麦克风阵列至少部分区别于所述第一麦克风阵列;以及
    所述信号处理器被配置为基于所述第二麦克风阵列拾取的声音信号更新所述降噪信号。
  3. 根据权利要求2所述的开放式耳机,其特征在于,所述基于所述第二麦克风阵列拾取的声音信号更新所述降噪信号包括:
    基于所述第二麦克风阵列拾取的声音信号,对用户耳道处的声场进行估计;以及
    根据用户耳道处的声场,调整所述降噪信号的参数信息。
  4. 根据权利要求2所述的开放式耳机,其特征在于,所述信号处理器进一步被配置为:
    获取用户输入;以及
    根据用户输入调整所述降噪信号的参数信息。
  5. 根据权利要求2所述的开放式耳机,其特征在于,所述第二麦克风阵列包括一个比所述第一麦克风阵列中任意麦克风更加靠近用户耳道的麦克风。
  6. 根据权利要求1所述的开放式耳机,其特征在于,所述信号处理器基于所述拾取的环境噪声估计第一空间的噪声包括:
    根据所述拾取的环境噪声进行信号分离,获取所述环境噪声对应的参数信息,基于所述参数信息生成降噪信号。
  7. 根据权利要求1所述的开放式耳机,其特征在于,所述信号处理器基于所述拾取的环境噪声估计第一空间位置的噪声包括:
    确定一个或多个与所述拾取的环境噪声有关的空间噪声源;以及
    基于所述空间噪声源,估计所述第一空间位置的噪声。
  8. 根据权利要求7所述的开放式耳机,其特征在于,所述确定一个或多个与所述拾取的环境噪声有关的空间噪声源包括:
    将所述拾取的环境噪声划分为多个子带,每个子带对应不同的频率范围;以及
    在至少一个子带上,确定与其对应的空间噪声源。
  9. 根据权利要求8所述的开放式耳机,其特征在于,所述第一麦克风阵列包括第一子麦克阵列和第二子麦克风阵列,所述第一子麦克风阵列和所述第二子麦克风阵列分别位于用户的左耳和右耳处,所述确定与所述至少一个子带对应的空间噪声源包括:
    获取用户头函数,所述用户头函数反映用户头部对声音的反射或吸收情况;以及
    在所述至少一个子带上,结合第一子麦克风阵列拾取的环境噪声、第二子麦克风阵 列拾取的环境噪声,以及所述用户头函数,确定与其对应的空间噪声源。
  10. 根据权利要求7所述的开放式耳机,其特征在于,所述确定一个或多个与所述拾取的环境噪声有关的空间噪声源包括:
    通过波束形成、超分辨空间谱估计或到达时差中的一种或多种方式定位所述一个或多个空间噪声源。
  11. 根据权利要求1所述的开放式耳机,其特征在于,所述第一麦克风阵列包括一个噪声麦克风,所述至少一个扬声器阵列形成至少一组声学偶极子,且所述噪声麦克风位于所述偶极子辐射声场的声学零点处。
  12. 根据权利要求1所述的开放式耳机,其特征在于,所述至少一个麦克风阵列包括骨导麦克风,所述骨导麦克风被配置于拾取用户的说话声音,所述信号处理器基于所述拾取的环境噪声估计第一空间位置的噪声包括:
    从所述拾取的环境噪声中去除与所述骨导麦克风拾取的信号相关联的成分,以更新所述环境噪声;以及
    根据所述更新后的环境噪声估计第一空间位置的噪声。
  13. 根据权利要求1所述的开放式耳机,其特征在于,所述至少一个麦克风阵列包括骨传导麦克风和气传导麦克风,所述信号处理器基于所述耳机的工作状态控制所述骨传导麦克风和所述气传导麦克风的开关状态。
  14. 根据权利要求13所述的开放式耳机,其特征在于,所述耳机的状态包括通话状态和未通话状态,
    若所述耳机的工作状态为未通话状态,则所述信号处理器控制所述骨传导麦克风为待机状态;以及
    若所述耳机的工作状态为通话状态,则所述信号处理器控制所述骨传导麦克风为工作状态。
  15. 根据权利要求14所述的开放式耳机,其特征在于,所述耳机的工作状态为通话状态时,若所述环境噪声的声压级大于预设阈值时,所述信号处理器控制所述骨传导麦克风保持工作状态;
    若所述环境噪声的声压级小于预设阈值时,所述信号处理器控制所述骨传导麦克风由工作状态切换至待机状态。
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